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[Serusers] SER and RTPproxy?!
by Caxton 24 Aug '06

24 Aug '06

Hi all: The document metion that "RTPproxy will not be installed on the SER server, but on a remote machine." If I install SER on "192.168.1.3", and install RTPproxy on "192.168.1.5". How do they work? Now, I install they on a same machine, and I run SER first, then usr command "rtpproxy -l xxx.xxx.xxx.xxx" to force rtpproxy. The SIP client can register behind NAT, but there are some problem with RTP stream. I found that RTP strem is able to relay to this machine, but this machine do not transmit those data to destination!! PC(A) Server(SER+RTPproxy) PC(B) RTP Stream(A) RTP Stream(B) 1. ----------------------------> <---------------------------- 2. Receive RTP stream A and B 3. Server should transmit RTP stream(A) to PC(B) and RTP stream(B) to PC(A), but this step do not work. (I am not sure that server does not forward the RTP packets or Clients do not receive RTP packet.) What is the problem? How should I operate if RTPproxy install on another machine? Because the command "rtpproxy" will not be used on the machine which have installed SER. Please help me, thank you. Regards, Caxton

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[Serusers] Alternate port for rewritehost
by Stephen Helfen 24 Aug '06

24 Aug '06

All,I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions?Thanks,Steve _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://get.live.com/toolbar/overview

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[Users] Please help for guiding serweb installation in openser server
by Ferianto siregar 24 Aug '06

24 Aug '06

Dear all, Thank you very much to all of you for help me to answer my question about serweb before. So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server. But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet. These are some questions that I need to ask: 1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like? Because when we untar it, there will be a serweb folder in the location where we untar it (for example, when we untar the packet in root there will be a serweb directory in root) . This serweb folder also contain html directory and phplib directory. As I know, There are also html directory in /root/var/www. So, What directory should I use? Should I delete the html directory and phplib directory in /root/var/www/? 2. Is there any relationship between the html directory in /root/var/www and html directory from /root/serweb? Which one should I use? 3. In the installation note I got this message : "install it, unpack the distibution on your webserver so that the directory html is in document root and phplib is not accessible via web" Can anybody help me to explain what this message means? and give me an example? 4. If I can make a wish for all of you, I do hope anybody can share the serweb configuration that she/he has configured to me. So, it can become the guidelines for me. That `s all the question that I got when I tried to install the serweb. I am confused. I do hope anybody can help me because I am new in this system. Thank you very much for your help. Thanks Regards, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] NAT (again)
by Mark Kent 24 Aug '06

24 Aug '06

I'm testing my rtpproxy config with a linksys phone and a grandstream phone, either not NAT'ed, or behind a cisco (full cone), or behind a Netgear (symmetric). Things generally work, but I have a problem with the grandstream behind the symmetric nat. This is the register: U NAT-IP:3284 -> PROXY-IP:5060 REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK0103d0ff79cc9c02 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=743335557afcade4 To: <sip:frick@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Authorization: Digest username="frick", realm="sip.example.com", algorithm=MD5, uri="sip:sip.example.com", nonce="foo", response="bar" Call-ID: 11f6defb0857db17(a)192.168.0.100 CSeq: 116 REGISTER Expires: 60 User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 This is the resulting contact: ...Record(0x2864b680).. domain: 'location' aor : 'frick(a)sip.example.com' ~~~Contact(0x2864d4e0)~~~ domain : 'location' aor : 'frick(a)sip.example.com' Contact : 'sip:frick@NAT-IP:3280;user=phone' Expires : 34 q : Call-ID : '11f6defb0857db17(a)192.168.0.100' CSeq : 119 User-Agent: 'Grandstream BT100 1.0.6.7' received : 'sip:NAT-IP:3284' Path : '' State : CS_SYNC Flags : 1 Sock : PROXY-IP:5060 (0x8122f50) Methods : 5695 next : 0x0 prev : 0x0 ~~~/Contact~~~~ .../Record.. Now, what I don't understand is the two ports used above: 3280 and 3284. It looks like the phone thinks it is being natted to port 3280, but it's really getting port 3284 When I make a call I get U NAT-IP:3284 -> PROXY-IP:5060 INVITE sip:16509415674@sip.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=2dc2fc3c94e93951 To: <sip:16509415674@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Supported: replaces Call-ID: 1f04f9c6ec474f70(a)192.168.0.100 CSeq: 26381 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 419 v=0 o=frick 8000 8002 IN IP4 NAT-IP s=SIP Call c=IN IP4 NAT-IP t=0 0 m=audio 3295 RTP/AVP 0 8 4 18 2 15 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 and my side sends back the Proxy Authentication Required, but it sends it to port 3280. Now, is that based on the Via: header or the Contact: header, or (unlikely) the stored contact info from the REGISTER? The phone acts like it never got it (and I'm guessing it didn't) and sends the INVITE again without any auth data. The thing that bugs me is that this is an outbound call, from the phone out through the NAT box, I figured that would be doable. The Linksys942 works, so I'm hoping this is a Grandstream bug, or something I can fix in my openser config. Thanks, -mark

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[Users] Limiting Call Volume and CPS
by Brian Jones 24 Aug '06

24 Aug '06

Hi, I was wondering if anyone had any suggestions on how I could use OpenSER to limit call volume and call per second on a peer basis. Thanks! Brian.

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[Users] Weird forwarding with Route header
by Jeremie Le Hen 24 Aug '06

24 Aug '06

Hi list, my proxy (yy.yy.yy.yy) receives a message like this: BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> My ruleset is something like this (with the REGISTER logic pulled out) : %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; force_rport(); if (!method == "REGISTER") { record_route(); }; if (loose_route()) { route(1); }; loopkup("location"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% I would expect the packet to leave my router, back to xx.xx.xx.xx : BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:yy.yy.yy.yy;lr;ftag=2873864> Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=zu83fdhDB8d.67235676.0 Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> But the outgoing packet is : BYE sip:xx.xx.xx.xx;ftag=1056999653;lr=on SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=z9hG4bK07cb.02005244.0 Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Do you have any idea about what is going on ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] dropping a 200
by Simon Morvan 24 Aug '06

24 Aug '06

Just to be sure : there's no way to call drop() in an onreply_route block when it's not a provisionnal reply ? -- Simon.

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[Users] how to correctly handle NAT on inbound call from another SIP-proxy?
by Gerd Feiner 24 Aug '06

24 Aug '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi list, we have some trouble handling NAT correctly when receiving a call from another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP when it finds another proxy's record-route header, though it invokes mediaproxy and sends uncorrected SDP to the other SIP-server resulting in one-way-audio as the provided c=-field includes the RFC1918-address of the client. when receiving calls through our Cisco AS5350 everything is fine ('cause it does not add the record-route). the addition of the header cannot be circumvented bacause the other provider wants to bill calls (obviously) and therefor needs to be in the SIP-dialog. is there a neat solution to this problem or do we have to get a session-border-controller or some B2BUA-config for handling the received calls? and by the way, how do i distinguish external INVITEs destined for one of our clients (uri==myself) from the INVITEs being sent out to the callee upon an external INVITE? if there was a way we could invoke mediaproxy from there? how is the recommended setup? brgds, g.feiner -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE7aVOqvDCGdeEb8oRArDjAJ9ssHEHBAIGmSB34oHQxuuT0ke15wCgmKvB chyomyJgqU9NVMep7QV1nFI= =r5Xb -----END PGP SIGNATURE-----

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[Users] Q:ABOUT:MODULE:auth_radius
by Alejandro Sanchez 24 Aug '06

24 Aug '06

Hello. openser 1.1 radiusclient-ng 5 freeradius 1.1.1 I need to enable the modules *_radius for my openser instalation a follow the guide for enable radius then when i try to start openser i get the following error. 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute Anyone knows what is the cause for this error? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

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[Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

Hi, Is there a way to coexist multiple domains in Ser, distinguishing users even if they have the same username in subscriber table but different domains? I'll explain better, I have subscriber table with: | username | domain | e-mail | ------------------------------------------------ | user_A | domain_A | user_A@domain_A | ------------------------------------------------ | user_A | domain_B | user_A@domain_B | and in domain table there are both domains domain_A and domain_B. But when someone calls user_A, (with or without writing their domain), both phones ring (if both users have their phones registered), when I want only one to ring because they are different users!! Is there a way in Ser to do this right? Regards, Ricardo.

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Re: [Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

It's true Asterisk can use different contexts, but that doesn't solve my problem because I use Ser + Asterisk and Ser is used for routing SIP messages between SIP UAs, not Asterisk. In my architecture, Asterisk is only used for advanced services like voicemail or call conferencing and for gateway, not to route messages between UAs. Ricardo. openser(a)comcast.net wrote: > The way to do that is intergrate Ser with Asterisk and in Asterisk you > can build context and keep the same numbers for different context like > a pbx. > > > -------------- Original message -------------- > From: Ricardo Carvalho <rcarvalho(a)iric.up.pt> > > > Hi, > > > > Is there a way to coexist multiple domains in Ser, > distinguishing users > > even if they have the same username in subscriber table but > different > > domains? > > > > I'll explain better, I have subscriber table with: > > > > | username | domain | e-mail | > > ------------------------------------------------ > > | user_A | domain_A | user_A@domain_A | > > ------------------------------------------------ > > | user_A | domain_B | user_A@domain_B | > > > > and in domain table there are both domains domain_A and domain_B. > > > > But when someone calls user_A, (with or without writing their > domain), both > > phones ring (if both users have their phones > > registered), when I w! ant onl y one to ring because they are > different users!! > > Is there a way in Ser to do this right? > > > > Regards, > > > > Ricardo. > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Voice Mail problem
by ram 24 Aug '06

24 Aug '06

Hi I have existing SER Running i have local users ( extensions) and DID ( all other countries) iam able to make calls in and out , there is no problem but when the user not available, it has to send to Voice mail, but its only sending the users who are locally start with 9*, how can i send the user start with 9 or any DID not availble send to voicemail here is my config if (!lookup("location") && ( uri=~"^sip:9.*@domain.com" ) ) { # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ setflag(2); if(t_newtran()){ t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log(1,"**************** vm start - begin ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log(1,"**************** vm start - end ******************\n"); break; }; if(method=="BYE"){ log(1,"**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log(1,"**************** vm end - end ******************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; break; }; break; }; any help will be great Ram

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[Serusers] Broken ser.cfg
by Jean-Michel Hiver 23 Aug '06

23 Aug '06

Can somebody please take a look at the routing logic below... I am trying to use exec_dset along with another program to do some LCR with SER but it doesn't work properly (here is a capture of what it produces: http://pastebin.ca/146771 Any ideas of what's doing wrong? Code: # -------------------------------------- route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { sl_send_reply("501", "Not Implemented"); break; }; if (method=="OPTIONS") { sl_send_reply("501", "Not Implemented"); break; }; record_route(); setflag(1); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="INVITE") { route(2); break(); }; route(1); break(); } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { # replaces the INVITE URI with the right host exec_dset ("/usr/local/Asterisk-LCR/bin/ser-lcr"); if (uri=~"^sip:STOP@127") { sl_send_reply("503", "Service Unavailable"); break; }; t_on_failure("2"); if(isflagset(1)) { append_branch(); }; setflag(1); t_relay(); } # /usr/local/Asterisk-LCR/bin/ser-lcr will give us the next uri to try # in case of failure, or send sip:STOP@127.0.0.1 if no more routes failure_route[2] { route(2); } # --------------------------------------

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[Serusers] Need config help for PSTN call routing blocks to service provider.
by Chahn John Kim 23 Aug '06

23 Aug '06

Hi, I am new to SER and need experts' help in PSTN call routing block to call termination service provider in a scenario below: We have our own sip realm/server that will handle all VoIP calls among our own subscribers; sip.ours.com. This will forward all PSTN calls, authenticated and authorized by it, to sip.theirs.com, which will authorize usning our IP address and forward to pstngateway.theirs.com. What I have been given by SP is: sip.theirs.com, IP, myuserID and password. What they are given by me is our IP and hostaname. sip.ours.com and IP. My first try was simply; rewritehostport(sip.theirs.com:5060); t_relay(); ,which fails. Can someone tell me proper approach in ser commands based on the scenario above? Thank you so much in advance. John K

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Re: [Users] options for rtpproxy.
by Bogdan-Andrei Iancu 23 Aug '06

23 Aug '06

Hi Bill, Bill Zhang wrote: >Anyone can explain following options: > >1. -2fv > > v - print the version f - stay in foreground, don't become a daemon 2 - duplicate mode >2. -s path > > command socket path >3. -L nfiles > > maximum file descriptor number that can be opened regards, bogdan >The rest of them seems to be straight forward. This rtproxy documentation >is HARD to find:(. > >usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t >tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] > >Thanks in advance. > >Best Regards, >Bill > >

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Re: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:). Re: Users Digest, Vol 15, Issue 61
by Mark Kent 23 Aug '06

23 Aug '06

In the RTPproxy source code, if you look at rtpp_log.h you really get the impression that syslog() functionality can be dropped in with ease. I still believe that, and believe it can be elegant, despite the fact that my solution is fairly crappy. But, it's done... so here it is (it relies on dprint.[ch] from the openser1.1 distribution): After ./configure, do this: In Makefile, change CFLAGS line like this: CFLAGS = -g -O2 -DUSESYSLOG -I../openser-1.1.0-tls (point to wherever you build openser) In rtpp_log.h, bracket the existing macros with this: #ifdef USESYSLOG #include "dprint.h" #define rtpp_log_open(app, call_id, flag) (0) #define rtpp_log_write(level, handle, format, args...) LOG(level, format, ##args) #define rtpp_log_ewrite(level, handle, format, args...) \ do { \ LOG(level, format, ##args); \ LOG(level, ": %s\n", strerror(errno)); \ } while(0); #define rtpp_log_close(handle) while (0) {} #else /* USESYSLOG */ {existing macros starting with rtpp_log_open and including rtpp_log_close} #endif /* USESYSLOG */ and stuff these lines up near the top of main.c, around line 64: #ifdef USESYSLOG int debug = 1; int log_stderr = 0; int log_facility = LOG_LOCAL7; char* log_name = "foobar"; int pt = 0, process_no = 0; #endif The link will fail when you do make, but just copy-and-paste the failed link command back into the shell and add ../openser-1.1.0-tls/dprint.o to the end of the line and it'll link, and run, and log. Also, some OS (Solaris) will require you to open and close the syslog channel. So, if you need to do that, then you need to define the rtpp_log_open and rtpp_log_close macros too. I'm on freebsd, and it seems I don't need to do that. -mark P.S. If someone has an elegant solution to this then please post it.

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[Serusers] SER / Missing call records
by Darren Sessions 23 Aug '06

23 Aug '06

All, We're experiencing a problem with missing call records. There is nothing in the log files that indicate a problem either with SER or our database server and likewise there is nothing in the database server's logs to indicate a problem. System loads are within reason on both servers, network connectivity has been tested and verified, and a through hardware shakedown on both boxes has been completed without any errors. We first noticed this problem when our underlying vendor's bill (based on minutes of course) did not correspond with our own. Upon doing a CDR comparison (based on what was in both the acc and calls tables), we were able to verify that the calls in dispute were indeed ours (based on the sip call-id's IP addresses). We're processing roughly 1.4 million calls per day on this SER server in question. Out of those calls, 508 calls (give or take a couple hundred) with a total duration of over 5,300 minutes - can not be accounted for in any of the SER database tables. We're running 0.9.5-pre1 with Suse Enterprise Server 9.0 (2.6.11 Kernel). Server is an HP DL380 with dual 3.2GHz processors with 6GB physical ram (only 1.4GB in use system wide); disk space is not an issue as well (only 10% in use). We've scoured the mailing lists and change logs for any hint of an issue similar to ours without any results. The closest we've been able to come that's anywhere near along the lines of our issue in any way that made any sense were these posts: http://lists.iptel.org/pipermail/serdev/2003-August/000602.html http://lists.iptel.org/pipermail/serusers/2005-February/015102.html At this point, we're not sure if this issue has something to do with the FIFO and SER making bulk dump uploads with call data to the database, or some other issue along those lines - or if this is a new issue all together. Any help would be greatly appreciated. Thanks, - Darren Fifo settings: fifo="/tmp/ser_fifo" fifo_db_url="mysql://xxx:xxx@xxx/serdb" Here are our module specific parameters: # -- timers -- modparam("tm", "wt_timer", 10) modparam("tm", "retr_timer1p1", 2) # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- database URL -- modparam("usrloc|acc|auth_db|group","db_url","mysql://xxx:xxx@xxx/serdb" ) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth","secret","xxx") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- Nathelper -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) # -- acc params -- modparam("acc", "db_flag", 3)

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[Serusers] Numbering Plan
by Daniel Mirrha Santos 23 Aug '06

23 Aug '06

Does anybody knows if I can create a numbering plan using a database? How is the easiest mode to do a numbering plan on ser? Thanks, Daniel

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Re: [Users] 403 Forbidden-----onsip configs testing. user not able to register
by Ferianto siregar 23 Aug '06

23 Aug '06

Hi, Ram its Not looking up in aliases table , it look ups only on location table. the each request and responses . So, You have to add your domain in the domain table by using : insert into domain values('urs_domain', ' date) ; in openser databases tables. I do hope it can help you. Thanks. On 8/22/06, ram wrote: > > Hi all > > iam trying to read the getting started document from onsip > when iam configuring pstn-gw.cfg > with the below config > > when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of > X-lite > > what is wrong , even i have changed AVP after reading module configs in > openser. > > any one suggest me what is wrong iam doing here > > Ram > > ----------------- > openser.cfg > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=9 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > #fork=no > #log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > listen=myip > port=5060 > children=4 > fifo="/tmp/openser_fifo" > fifo_db_url="mysql://openser:openserrw@localhost /openser" > > > # ------------------ module loading > ---------------------------------- > > > loadmodule "/usr/local/lib/openser/modules/mysql.so" > loadmodule "/usr/local/lib/openser/modules/sl.so" > loadmodule "/usr/local/lib/openser/modules/tm.so" > loadmodule "/usr/local/lib/openser/modules/rr.so" > loadmodule "/usr/local/lib/openser/modules/maxfwd.so" > loadmodule "/usr/local/lib/openser/modules/usrloc.so" > loadmodule "/usr/local/lib/openser/modules/registrar.so" > loadmodule "/usr/local/lib/openser/modules/textops.so" > loadmodule "/usr/local/lib/openser/modules/auth.so" > loadmodule "/usr/local/lib/openser/modules/auth_db.so" > loadmodule "/usr/local/lib/openser/modules/uri.so" > loadmodule "/usr/local/lib/openser/modules/uri_db.so" > loadmodule "/usr/local/lib/openser/modules/domain.so" > loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/openser/modules/nathelper.so" > loadmodule "/usr/local/lib/openser/modules/avpops.so" > loadmodule "/usr/local/lib/openser/modules/permissions.so" > > modparam("auth_db|permissions|uri_db|usrloc","db_url", > "mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "password_column", "password") > > modparam("nathelper", "rtpproxy_disable", 1) > modparam("nathelper", "natping_interval", 0) > > modparam("mediaproxy","natping_interval", 30) > modparam("mediaproxy","mediaproxy_socket", > "/var/run/mediaproxy.sock") > modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") > > modparam("usrloc", "db_mode", 2) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > > modparam("registrar", "nat_flag", 6) > > modparam("rr", "enable_full_lr", 1) > > modparam("tm", "fr_timer_avp", "i:25") > modparam("tm", "fr_inv_timer_avp", "i:25") > > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > route { > > # > ----------------------------------------------------------------- > # Sanity Check Section > # > ----------------------------------------------------------------- > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483", "Too Many Hops"); > return; > }; > > if (msg:len > max_len) { > sl_send_reply("513", "Message Overflow"); > return; > }; > > # > ----------------------------------------------------------------- > # Record Route Section > # > ----------------------------------------------------------------- > if (method=="INVITE" && client_nat_test("3")) { > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("myproxyip:5060;nat=yes"); > } else if (method!="REGISTER") { > record_route(); > }; > > # > ----------------------------------------------------------------- > # Call Tear Down Section > # > ----------------------------------------------------------------- > if (method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > # > ----------------------------------------------------------------- > # Loose Route Section > # > ----------------------------------------------------------------- > if (loose_route()) { > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > if > (client_nat_test("3")||search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > return; > }; > > # > ----------------------------------------------------------------- > # Call Type Processing Section > # > ----------------------------------------------------------------- > if (!is_uri_host_local()) { > if (is_from_local() || allow_trusted()) { > route(4); > route(1); > } else { > sl_send_reply("403", "Forbidden"); > }; > return; > }; > > if (method=="CANCEL") { > route(1); > return; > } else if (method=="INVITE") { > route(3); > return; > } else if (method=="REGISTER") { > route(2); > return; > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; > }; > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > return; > }; > > route(1); > } > > route[1] { > > # > ----------------------------------------------------------------- > # Default Message Handler > # > ----------------------------------------------------------------- > > t_on_reply("1"); > > if (!t_relay()) { > > if (method=="INVITE" || method=="ACK") { > end_media_session(); > }; > > sl_reply_error(); > }; > } > > route[2] { > > # > ----------------------------------------------------------------- > # REGISTER Message Handler > # > ----------------------------------------------------------------- > > sl_send_reply("100", "Trying"); > > if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > > if (!www_authorize("","subscriber")) { > www_challenge("","0"); > return; > }; > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > return; > }; > > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > } > > route[3] { > > # > ----------------------------------------------------------------- > # INVITE Message Handler > # > ----------------------------------------------------------------- > > if (!allow_trusted()) { > > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > return; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > return; > }; > > consume_credentials(); > }; > > if (client_nat_test("3")) { > setflag(7); > force_rport(); > fix_nated_contact(); > }; > > if (uri=~"^sip:1[0-9]{10}@") { > strip(1); > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; === message truncated === --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

2 1

[Users] Problem with forking
by Claudio Miceli 23 Aug '06

23 Aug '06

I am in a big trouble. I am using openser 1.0.1 and I am still having some problems with redirect. I get some numbers in myredirect and I try to pass it with t_relay(), but at this moment everything gets confused. The calls do not happen sequentially or in any logical order. Is there another way of doing the forking ? Am i doing something terribly wrong ? And sometimes when I make a call my openser sends messages to just one client and not to the other, or sends to both but with a great time between them. What could it be ? Do someone know it ? Thanks for helping. Claudio Miceli de Farias

2 1

RE: [Users] registered("location")
by Helge Waastad 23 Aug '06

23 Aug '06

Hi again, I have: modparam("group|auth_db|usrloc|uri_db|speeddial|registrar|avpops|alias_d b", "use_domain", 1) In location table: | username | domain | | | | | 123456 | smartnet.no | | 654321 | smartnet.no | If I set usrloc|registrar use_domain=0, the only thing that changes is that the domain column is empty. Br hw -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: 22. august 2006 18:06 To: Helge Waastad Cc: Users(a)openser.org Subject: Re: [Users] registered("location") strange.... do you have the use_domain param enabled or disabled? could you check if the location table has for "user" column the correct format (all entries): username@domain (if use_domain is on) username (if use_domain is off) regards, bogdan Helge Waastad wrote: >Hi, >Ive just stopped my openser server, >deleted all entries in the location table and restarted the OpenSER >server. > >Still i get the same: > >BUG:usrloc:db_load_urecord: AOR has no @ while use_domain is enabled > >message. > >br hw > >tir, 22,.08.2006 kl. 16.36 +0300, skrev Bogdan-Andrei Iancu: > > >>Hi, >> >>my guess is you had usrloc with use_domain and with contacts into the >>location table you switched use_domain on. >>BUG:usrloc:db_load_urecord: AOR has no @ while use_domain is enabled >>please remove the entire contact of the location table and try again. >> >>regards, >>bogdan >> >>Helge Waastad wrote: >> >> >> >>>Hi, >>> >>>I receive a >>>BUG:usrloc:db_load_urecord: AOR has no @ while use_domain is enabled >>> >>>in my syslog. Could this be the reason??? >>> >>>br hw >>> >>> >>> >>>tir, 22,.08.2006 kl. 15.28 +0300, skrev Bogdan-Andrei Iancu: >>> >>> >>> >>> >>>>Hi Helge, >>>> >>>>it should work. Can you check if any db query is performed by the function? >>>> >>>>regards, >>>>bogdan >>>> >>>>Helge Waastad wrote: >>>> >>>> >>>> >>>> >>>> >>>>>Hi, >>>>>I'm just wondering if the 'registered("location")' works when using >>>>>db only mode for usrloc? >>>>> >>>>>br hw >>>>> >>>>> >>>>>_______________________________________________ >>>>>Users mailing list >>>>>Users(a)openser.org >>>>>http://openser.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>

3 5

[Users] Please help me for SMS module (SIP to GSM)
by ade darmawan 23 Aug '06

23 Aug '06

Please help me mr... i have read this module...but i don't understand with 4 question : 1. I found that in the SMS Module (SIP to GSM) don't explained the software which used by the client/user agent or where we should to do testing in client? 2. I wanna know the other facilities should be used, beside openser-1.0.0.tar.gz,mysql,DNS? 3. For the final result, is this module could send SMS by 2 directions? 4. Could you please the step to do the testing? Thank's for all --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

2 1

[Users] Byes not accounted
by Jayesh Nambiar 23 Aug '06

23 Aug '06

Hi all, I have got a strange problem with my accounting. This is my scenario: I have a registered user dialling a pstn number. The call is connected and then the registered UA switches off the IP device and there is no BYE from the registered UA. Now the pstn gateway sends a BYE after certain period and Openser tries to send that BYE to the registered UA, but the registered UA is powered off and it never replies. So Openser retransmits the BYE messages and finally generates a 408 response towards the gateway. But Openser does not account that BYE in the acc table. According to me, the Openser should acount that BYE for proper accouning, Please correct me if I am wrong. Also suggest some method so that I can make the unacknowledged BYEs account in the acc table. In my config I have: if(method=="BYE") { setflag(1); }; if(method!="REGISTER") { record_route();}; setflag(1); if(loose_route()) { ..... #the rest logic} an so on. I have flag 1 as db_flag. thanks in advance, w/regards, jayesh. --------------------------------- Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW

3 2

[Users] libmysqlclient15-dev Download Fail
by openser＠comcast.net 23 Aug '06

23 Aug '06

I am downloading the prerequisite for Openser and one of the items keeps failing me. This the error below. Any help will be appreciated. [root@localhost openser]# yum install libmysqlclient15-dev Setting up Install Process Setting up repositories updates-released 100% |=========================| 951 B 00:00 extras 100% |=========================| 1.1 kB 00:00 base 100% |=========================| 1.1 kB 00:00 Reading repository metadata in from local files Parsing package install arguments No Match for argument: libmysqlclient15-dev Nothing to do

2 1

Re: [Users] rtpproxy or RTPProxy, where to download or checkout
by Klaus Darilion 23 Aug '06

23 Aug '06

Bill Zhang wrote: > This could be a simple question but I went through the whole user archive > and yet to find out where to download it or cvs checkout, I do see > mediaproxy as part of OPENSER module, wonder why rtpproxy or RTPProxy is > not:(. Hi Bill! There are 2 RTP proxies available. 1. Mediaproxy 2. rtpproxy To communicate with Mediaproxy, openser uses the mediaproxy module. To communicate with rtpproxy, openser uses the nathelper module. The modules are part of openser, but not the RTP proxies itself. You can download Mediaproxy from ag-projects: http://mediaproxy.ag-projects.com/ You can download rtpproxy from berlios cvs (which is insides ser's CVS repository) http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/rtpproxy/ cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser login press enter when prompted for a password cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy then do the usual: ./configure; make; make install regards klaus > > Thanks in advance for help. > > Bill > > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf > Of Atle Samuelsen > Sent: Monday, August 14, 2006 12:56 PM > To: Ever Zalazar > Cc: users(a)openser.org > Subject: Re: [Users] mediaproxy vs. rtpproxy > > > Hi ever, > > I would say this is not a proper question. Both work like a charm with > (Open)SER. > > The question you should ask yourself is what do you want to accomplish > with mediaproxy/rtpproxy. If itÂ´s only calls between nated useragensts, > you can use both without any problem. > > - Atle > * Ever Zalazar <zalazar(a)technologymaker.com> [060814 21:41]: >> Hi people! Just to know, which is much better for openser mediaproxy or > rtpproxy? >> >> Best Regards >> >> >> Ever > >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] Redirect
by Andrew And 23 Aug '06

23 Aug '06

I am using redirect with ENUM, and am I wanting to know if my code is correct, then I write the code below. Could someone help me please? . . . } else if (method=="INVITE") { # If Invite is to ENUM. # Block ENUM prefix("+"); enum_query("e164.arpa."); #Redirect sl_send_reply("300","Redirect"); return; };

2 1

Re: [Users] 403 Forbidden-----onsip configs testing. user not able to register
by Ferianto siregar 23 Aug '06

23 Aug '06

Hi, Ram its Not looking up in aliases table , it look ups only on location table. the each request and responses . So, You have to add your domain in the domain table by using : insert into domain values('urs_domain', ' date) ; in openser databases tables. I do hope it can help you. Thanks. On 8/22/06, ram wrote: > > Hi all > > iam trying to read the getting started document from onsip > when iam configuring pstn-gw.cfg > with the below config > > when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of > X-lite > > what is wrong , even i have changed AVP after reading module configs in > openser. > > any one suggest me what is wrong iam doing here > > Ram > > ----------------- > openser.cfg > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=9 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > #fork=no > #log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > listen=myip > port=5060 > children=4 > fifo="/tmp/openser_fifo" > fifo_db_url="mysql://openser:openserrw@localhost /openser" > > > # ------------------ module loading > ---------------------------------- > > > loadmodule "/usr/local/lib/openser/modules/mysql.so" > loadmodule "/usr/local/lib/openser/modules/sl.so" > loadmodule "/usr/local/lib/openser/modules/tm.so" > loadmodule "/usr/local/lib/openser/modules/rr.so" > loadmodule "/usr/local/lib/openser/modules/maxfwd.so" > loadmodule "/usr/local/lib/openser/modules/usrloc.so" > loadmodule "/usr/local/lib/openser/modules/registrar.so" > loadmodule "/usr/local/lib/openser/modules/textops.so" > loadmodule "/usr/local/lib/openser/modules/auth.so" > loadmodule "/usr/local/lib/openser/modules/auth_db.so" > loadmodule "/usr/local/lib/openser/modules/uri.so" > loadmodule "/usr/local/lib/openser/modules/uri_db.so" > loadmodule "/usr/local/lib/openser/modules/domain.so" > loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/openser/modules/nathelper.so" > loadmodule "/usr/local/lib/openser/modules/avpops.so" > loadmodule "/usr/local/lib/openser/modules/permissions.so" > > modparam("auth_db|permissions|uri_db|usrloc","db_url", > "mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "calculate_ha1", 1) > modparam("auth_db", "password_column", "password") > > modparam("nathelper", "rtpproxy_disable", 1) > modparam("nathelper", "natping_interval", 0) > > modparam("mediaproxy","natping_interval", 30) > modparam("mediaproxy","mediaproxy_socket", > "/var/run/mediaproxy.sock") > modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") > > modparam("usrloc", "db_mode", 2) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > > modparam("registrar", "nat_flag", 6) > > modparam("rr", "enable_full_lr", 1) > > modparam("tm", "fr_timer_avp", "i:25") > modparam("tm", "fr_inv_timer_avp", "i:25") > > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > > route { > > # > ----------------------------------------------------------------- > # Sanity Check Section > # > ----------------------------------------------------------------- > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483", "Too Many Hops"); > return; > }; > > if (msg:len > max_len) { > sl_send_reply("513", "Message Overflow"); > return; > }; > > # > ----------------------------------------------------------------- > # Record Route Section > # > ----------------------------------------------------------------- > if (method=="INVITE" && client_nat_test("3")) { > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("myproxyip:5060;nat=yes"); > } else if (method!="REGISTER") { > record_route(); > }; > > # > ----------------------------------------------------------------- > # Call Tear Down Section > # > ----------------------------------------------------------------- > if (method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > # > ----------------------------------------------------------------- > # Loose Route Section > # > ----------------------------------------------------------------- > if (loose_route()) { > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > if > (client_nat_test("3")||search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > return; > }; > > # > ----------------------------------------------------------------- > # Call Type Processing Section > # > ----------------------------------------------------------------- > if (!is_uri_host_local()) { > if (is_from_local() || allow_trusted()) { > route(4); > route(1); > } else { > sl_send_reply("403", "Forbidden"); > }; > return; > }; > > if (method=="CANCEL") { > route(1); > return; > } else if (method=="INVITE") { > route(3); > return; > } else if (method=="REGISTER") { > route(2); > return; > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; > }; > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > return; > }; > > route(1); > } > > route[1] { > > # > ----------------------------------------------------------------- > # Default Message Handler > # > ----------------------------------------------------------------- > > t_on_reply("1"); > > if (!t_relay()) { > > if (method=="INVITE" || method=="ACK") { > end_media_session(); > }; > > sl_reply_error(); > }; > } > > route[2] { > > # > ----------------------------------------------------------------- > # REGISTER Message Handler > # > ----------------------------------------------------------------- > > sl_send_reply("100", "Trying"); > > if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > > if (!www_authorize("","subscriber")) { > www_challenge("","0"); > return; > }; > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > return; > }; > > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > } > > route[3] { > > # > ----------------------------------------------------------------- > # INVITE Message Handler > # > ----------------------------------------------------------------- > > if (!allow_trusted()) { > > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > return; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > return; > }; > > consume_credentials(); > }; > > if (client_nat_test("3")) { > setflag(7); > force_rport(); > fix_nated_contact(); > }; > > if (uri=~"^sip:1[0-9]{10}@") { > strip(1); > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; === message truncated === --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

1 0

[Users] Create 302 message and modify contact header
by Christophe Irles 23 Aug '06

23 Aug '06

Hi, I checked archive without any result... From an openSER rule, I want to create a 302 message if a specific RURI is used. First I need to change the contact header to specify the URI of the new callee. I tried something like that: avp_printf("$avp(s:100)", "sip:test@christophe.fr;target=$ruri;cause=486;"); avp_pushto("$ct", "$avp(s:100)"); # => error generated on this line ERROR:avpops:fixup_pushto_avp: unsupported destination "$ct"; expected $ru,$du,$br So how can I modify the contact header ? Just after the update on the contact header I envisaged to use this function sl_send_reply(302, "Moved Temporarily") Is it correct ? Is there a better way to 'create' 302 message in this case ? I check UAC_REDIRECT module but I don't think I can use it in this case or perhaps I missed something Before to use a 302 message I succesfully tested to create a new INVITE with the $ruri modified correctly. But I would like to create a 302 message to prevent any problem of authentication. Thanks, Christophe

4 7

RE: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:). Re: Users Digest, Vol 15, Issue 61
by Mark Kent 23 Aug '06

23 Aug '06

>> RE: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:) And no one else might have the same hesitation as you? (i.e., why not post the answer to your own question). This is actually semi-interesting, because if you go here http://ftp.iptel.org/pub/rtpproxy/ you get the impression that rtpproxy-0.3 is the one to get. There is a file there named: LATEST-IS-0.3 But here: http://www.voip-info.org/wiki-Portaone+rtpproxy it gives instructions for how to pull it out of CVS: * Download from CVS note: you should update rtpproxy from CVS in order for it to work: cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser login press enter when prompted for a password cvs -z3 -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy then do the usual: ./configure; make; make install and the code you get is different. Note that the output of "rtpproxy -v" has not changed (yet). So, the CVS version is still 0.3, but it looks like some obvious bug fixes as some arrays have had their dimensions increased by one, plus a couple of new minor features (-T ttl and -L nfiles). But wait, there's more: there is even a man page (rtpproxy.8)!! >> Anyone can explain following options: >> >> 1. -2fv >> 2. -s path >> 3. -L nfiles The man page is there in the distribution from CVS: -2 Send every RTP packet twice in sessions that use low-bitrate codecs. Only packets that are smaller than 128 bytes will be sent twice. This option can improve audio quality on lossy links. -f Rtpproxy will stay in foreground mode if this option is set. -v Show version of program. -s ctrl_socket This parameter configures rtpproxy control socket. The control [snip] -L does not appear to be documented, but looking at the code it sets this: RLIMIT_NOFILE The maximum number of open files for this process. I have other things to say about rtpproxy, but I'll save it for the next email... -mark

2 1

[Users] SMS module (SIP to GSM) question?
by ade darmawan 23 Aug '06

23 Aug '06

ade darmawan <generasi84(a)yahoo.com> wrote: Please help me mr... i have read this module...but i don't understand with 4 question : 1. I found that in the SMS Module (SIP to GSM) don't explained the software which used by the client/user agent or where we should to do testing in client? 2. I wanna know the other facilities should be used, beside openser-1.0.0.tar.gz,mysql,DNS? 3. For the final result, is this module could send SMS by 2 directions? 4. Could you please the step to do the testing? Thank's for all --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. _______________________________________________ Devel mailing list Devel(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/devel --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

1 0

[Users] 403 Forbidden-----onsip configs testing. user not able to register
by ram 23 Aug '06

23 Aug '06

Hi all iam trying to read the getting started document from onsip when iam configuring pstn-gw.cfg with the below config when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of X-lite what is wrong , even i have changed AVP after reading module configs in openser. any one suggest me what is wrong iam doing here Ram ----------------- openser.cfg # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=myip port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", " mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer_avp", "i:25") modparam("tm", "fr_inv_timer_avp", "i:25") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("myproxyip:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); route(5); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("mygateway:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); route(4); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

2 1

[Users] options for rtpproxy.
by Bill Zhang 23 Aug '06

23 Aug '06

Anyone can explain following options: 1. -2fv 2. -s path 3. -L nfiles The rest of them seems to be straight forward. This rtproxy documentation is HARD to find:(. usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] Thanks in advance. Best Regards, Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, August 22, 2006 6:34 PM To: users(a)openser.org Subject: RE: [Users] rtpproxy or RTPProxy,where to download or checkout--found it:). Thanks. Best Regards, Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, August 22, 2006 5:55 PM To: users(a)openser.org Subject: [Users] rtpproxy or RTPProxy, where to download or checkout This could be a simple question but I went through the whole user archive and yet to find out where to download it or cvs checkout, I do see mediaproxy as part of OPENSER module, wonder why rtpproxy or RTPProxy is not:(. Thanks in advance for help. Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Atle Samuelsen Sent: Monday, August 14, 2006 12:56 PM To: Ever Zalazar Cc: users(a)openser.org Subject: Re: [Users] mediaproxy vs. rtpproxy Hi ever, I would say this is not a proper question. Both work like a charm with (Open)SER. The question you should ask yourself is what do you want to accomplish with mediaproxy/rtpproxy. If itÂ´s only calls between nated useragensts, you can use both without any problem. - Atle * Ever Zalazar <zalazar(a)technologymaker.com> [060814 21:41]: > Hi people! Just to know, which is much better for openser mediaproxy or rtpproxy? > > > Best Regards > > > Ever > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

RE: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:).
by Bill Zhang 23 Aug '06

23 Aug '06

Thanks. Best Regards, Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Bill Zhang Sent: Tuesday, August 22, 2006 5:55 PM To: users(a)openser.org Subject: [Users] rtpproxy or RTPProxy, where to download or checkout This could be a simple question but I went through the whole user archive and yet to find out where to download it or cvs checkout, I do see mediaproxy as part of OPENSER module, wonder why rtpproxy or RTPProxy is not:(. Thanks in advance for help. Bill -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Atle Samuelsen Sent: Monday, August 14, 2006 12:56 PM To: Ever Zalazar Cc: users(a)openser.org Subject: Re: [Users] mediaproxy vs. rtpproxy Hi ever, I would say this is not a proper question. Both work like a charm with (Open)SER. The question you should ask yourself is what do you want to accomplish with mediaproxy/rtpproxy. If itÂ´s only calls between nated useragensts, you can use both without any problem. - Atle * Ever Zalazar <zalazar(a)technologymaker.com> [060814 21:41]: > Hi people! Just to know, which is much better for openser mediaproxy or rtpproxy? > > > Best Regards > > > Ever > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] mediaproxy vs. rtpproxy
by Ever Zalazar 23 Aug '06

23 Aug '06

Hi people! Just to know, which is much better for openser mediaproxy or rtpproxy? Best Regards Ever

3 2

[Users] Transaction matching issue?
by H Quintana 22 Aug '06

22 Aug '06

Hello, I'm running OpenSER 1.1.0 in the server xxx.yyy.91.231, I'm also testing a program as 3rd party call control at xxx.yyy.91.230 that is sending "simultaneous" reINVITEs to the 2 parties involved in the call (10252052 and 10092010) through OpenSER. Both messages are accepted by the loose_route function. OpenSER relays the 1st reINVITE (to 10092010) but the second reINVITE (to 10252052) is not relayed. Is openSER considering the 2nd message as a retransmission of the first reINVITE ? I'm having the same behavior by setting tm.ruri_matching to 1 and 0. The results are the same if I call from 10092010 to 10252052, or if I send the 1st reINVITE to 10252052. The only way to make it work properly is sending the 2nd reInvite after some seconds. There is something else that I must add to the reINVITEs to allow OpenSER to differentiate them? The OpenSER debug for the ReINVITEs is at http://pastebin.com/773285 The reInvite to 10092010 is at http://pastebin.com/773286 The reInvite to 10252052 is at http://pastebin.com/773287 Thanks for any suggestion. Regards, Humberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Serusers] avps question
by Luis Silva 22 Aug '06

22 Aug '06

Hi all! Why isn't possible to do "$avp_name[0] = 0;" ? is there another way of doing it? tks in advance

1 0

[Serusers] preferred order of codecs?
by Ricardo Carvalho 22 Aug '06

22 Aug '06

Hi, Is there a way in Ser to stablish a preferred order of codecs to be used by the phones? Thanks, Ricardo.

6 11

[Users] $ru, $rU, $ru/username, $ruri/username, usage
by Greg Fausak 22 Aug '06

22 Aug '06

I want to use the RURI username in an avp query. I've seen a bunch of different ways to select it. Which way is correct? I am currently using $ruri/username, because I got an error with $rU. http://openser.org/docs/pseudo-variables.html shows '$rU', which complains when I try to reference that. Aug 21 06:55:58 www openser[89119]: ERROR:avpops:fixup_pushto_avp: unsupported destination "$rU"; expected $ru,$du,$br Aug 21 06:55:58 www openser[89119]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 82 It's weird, the source code in items.c reads: {{"ru", (sizeof("ru")-1)}, /* */ { XL_RURI, 0, xl_get_ruri, {{0, 0}, 0}, {0, 0}}}, {{"ruri", (sizeof("ruri")-1)}, /* */ { XL_RURI, 0, xl_get_ruri, {{0, 0}, 0}, {0, 0}}}, {{"rU", (sizeof("rU")-1)}, /* */ { XL_RURI_USERNAME, 0, xl_get_ruri_attr, {{0, 1}, 0}, {0, 0}}}, Is there a list of the variables that are available in an avp operation? -g

2 6

[Users] registered("location")
by Helge Waastad 22 Aug '06

22 Aug '06

Hi, I'm just wondering if the 'registered("location")' works when using db only mode for usrloc? br hw

2 5

[Serusers] Modifying the To Field and From field of an INVITE message
by Sammy Touati 22 Aug '06

22 Aug '06

Hi, Can the To and From fields in a SIP INVITE message be modified in SER? I know that the Request URI can be modified but i also need to modify the To field. Thanks. sato.

4 4

[Users] Please help for guiding serweb installation in openser
by Ferianto siregar 22 Aug '06

22 Aug '06

Dear all, Thank you very much to all of you for help me to answer my question about serweb before. So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server. But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet. These are some questions that I need to ask: 1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like? Because when we untar it, there will be a serweb folder in the location where we untar it (for example, when we untar the packet in root there will be a serweb directory in root) . This serweb folder also contain html directory and phplib directory. As I know, There are also html directory in /root/var/www. So, What directory should I use? Should I delete the html directory and phplib directory in /root/var/www/? 2. Is there any relationship between the html directory in /root/var/www and html directory from /root/serweb? Which one should I use? 3. In the installation note I got this message : "install it, unpack the distibution on your webserver so that the directory html is in document root and phplib is not accessible via web" Can anybody help me to explain what this message means? and give me an example? 4. If I can make a wish for all of you, I do hope anybody can share the serweb configuration that she/he has configured to me. So, it can become the guidelines for me. That `s all the question that I got when I tried to install the serweb. I am confused. I do hope anybody can help me because I am new in this system. Thank you very much for your help. Thanks Regards, Ferianto --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

1 0

[Users] t_on_failure doesn't
by Nathan Hawkins 22 Aug '06

22 Aug '06

I'm trying to use the LCR module to do a failover between two gateways. For testing purposes, I have a gateway record pointed at an IP that isn't used. The problem is, I can't seem to get OpenSER to fail over to the other gateway. I want to use TCP (with TLS when I'm done), but I get this: Jul 20 15:36:57 asiago /usr/sbin/openser[11896]: route[6]: routing to INVITE sip:MYDID@MYIPADDRESS:5060;transport=tcp Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR: tcp_blocking_connect: timeout (10) Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR: tcpconn_connect: tcp_blocking_connect failed Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR: tcp_send: connect failed Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: msg_send: ERROR: tcp_send failed Jul 20 15:37:17 asiago /usr/sbin/openser[11896]: ERROR:tm:t_forward_nonack: sending request failed (IP and username censored.) OpenSER sends this to the caller: SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM) Why can't I catch this with either the return code from t_relay or the failure route? It will go to the failure route if I get a reply back from the next hop. But I need to fail to the other gateway. The relevent parts of the configs are shown below. (Addresses censored.) Any help would greatly appreciated. Thanks, ---Nathan Config snippet: # ok, so maybe try a gateway... if( load_gws() ) { xlog("L_INFO","route[3]: gateway routing $rm $ru"); t_on_failure("1"); route(6); return; } } route[6] { xlog("L_INFO","route[6]: $rm $ru"); t_on_failure("1"); t_on_reply("1"); if (!next_gw()) { xlog("L_INFO","route[6]: no more gateways for $ru"); exit; } xlog("L_INFO","route[6]: routing to $rm $ru"); if(!t_relay()) { xlog("L_INFO","route[6]: t_relay returned $rc"); sl_reply_error(); } return; } failure_route[1] { xlog("L_INFO","failure_route[1]: $rm $ru"); t_on_failure("1"); if(t_check_status("404")) { xlog("L_INFO","failure_route[1]: failing call with 404 for $ru"); return; } if (!next_gw()) { xlog("L_INFO","route[6]: no more gateways for $ru"); exit; } append_branch(); xlog("L_INFO","failure_route[1]: sending $rm $ru"); if(!t_relay()) { xlog("L_INFO","route[6]: t_relay returned $rc"); } return; } Gateways: mysql> select * from gw where grp_id=4; +---------+--------+-----------+------+------------+-----------+--------+ | gw_name | grp_id | ip_addr | port | uri_scheme | transport | prefix | +---------+--------+-----------+------+------------+-----------+--------+ | ithaka | 4 | CENSORED | 5060 | NULL | 2 | NULL | | larink | 4 | BAD_ADDRESS | 5060 | NULL | 2 | NULL | +---------+--------+-----------+------+------------+-----------+--------+ 2 rows in set (0.00 sec) ithaka is a working OpenSER proxy. larink is an empty IP address. (There used to be a server named larink.)

3 6

[Users] SER + phone provisioning server - what are the options?
by Mahatma Sarasvati 22 Aug '06

22 Aug '06

Hi All, I'm tired of manually editing text files. What solutions are available for avoiding manual editing and saving of Cisco .cnf files for individual phones while using SER or OpenSER? I've got 80+ phones in my organization and I'm tired of manually managing the phone .cnf files I'm imagining some piece of code that auto-generates the files on command or on a schedule based on entries in the MySQL database. Maybe a column can be added to the Subscriber table or somewhere that holds the MAC address of the phone? Or is there a way to get the phone to get its default config via TFTP and its user information via http? Thanks, M _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] Bug in nathelper module or in the replace() function
by Federico Giannici 22 Aug '06

22 Aug '06

If a message body is modified by both force_rtp_proxy() and replace( "^Content-Length:", "l:" ) then the resulting message is corrupted. The message size is updated but the header field name appear in both the short and long form. Here it is an example: l:Content-Length: 403 The result is the same if I reverse the order of the two operations too. BTW, I use that replace call (and others) when the message is large to prevent it to fragment. Bye. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 6

[Serusers] SER + SEMS in TCP case
by Pat wang 22 Aug '06

22 Aug '06

Hi all, Has anyone tried SER + SEMS in TCP mode yet? I mean the UAs use TCP as transport protocol. I am able to setup SEMS as a conference server. I found "transport=TCP" is missing in the Contact header in the 200ok/SDP after SEMS answers the call. Eyebeam seems not like it and does not response to that 200ok. Result is SER keeps sending that 200ok/SDP till timeout. I tried to use textop module to catch that 200ok then replace the Contact header with a correct one but SER and SEMS communicate via fifo so I am not able to do so. Can anyone help me solve this issue please? cheers, Patrick

1 0

[Users] fail to listen on two addresses
by Jeremie Le Hen 22 Aug '06

22 Aug '06

Hi list, I am new two OpenSER. I installed OpenSER 1.1.0-notls and I need it to listen on both 127.0.0.1 and one of my physical interface's IP address. The manual page states: % -l address Listens on the specified address/interface. Multiple -l % mean listening on multiple addresses. The address format is % [proto:]address[:port], where proto = udp|tcp and address = % host|ip_address|interface_name. Example: -l localhost, -l % udp:127.0.0.1:5080, -l eth0:5062. The default behaviour is % to listen on all the ipv4 interfaces. I tried the following command-line, without success. I cannot connect to either one of the IP address I specified, depending on the order I provided them : % openser -u voipproxy -g voipproxy -l 84.xxx.xxx.xxx -l 127.0.0.1 -D -E And in another terminal: % # lsof -nPi4 -ap `pgrep -d, openser` % COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME % openser 30837 voipproxy 5u IPv4 60272 UDP 84.xxx.xxx.xxx:5060 % openser 30838 voipproxy 5u IPv4 60272 UDP 84.xxx.xxx.xxx:5060 % openser 30839 voipproxy 5u IPv4 60272 UDP 84.xxx.xxx.xxx:5060 If I turn the command-line to: % openser -u voipproxy -g voipproxy -l 127.0.0.1 -l 84.xxx.xxx.xxx -D -E I get: % # lsof -nPi4 -ap `pgrep -d, openser` % COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME % openser 30845 voipproxy 5u IPv4 60382 UDP 127.0.0.1:5060 % openser 30846 voipproxy 5u IPv4 60382 UDP 127.0.0.1:5060 % openser 30847 voipproxy 5u IPv4 60382 UDP 127.0.0.1:5060 Note that in both case I get: % Listening on % udp: 84.xxx.xxx.xxx [84.xxx.xxx.xxx]:5060 % udp: 127.0.0.1 [127.0.0.1]:5060 % tcp: 84.xxx.xxx.xxx [84.xxx.xxx.xxx]:5060 % tcp: 127.0.0.1 [127.0.0.1]:5060 Except the order change, following the command-line. I could have missed something and would be glad if someone pointed it out to me. Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Users] Q:ABOUT:MODULE:UsrLoc
by Alejandro Sanchez 22 Aug '06

22 Aug '06

Hi everybody. I have the next enviroment. OpenSer 1.1.X MySQL 5.0 Linux Fedora 4 Architecture: We have 4 servers one of them work like a LoadBalancer (openser proxy dispatcher), behind of it are the others 2 servers (openser) that have the function to kept the sessions, register, etc... they have the parameter in dbmode to 3 for kept out the cache system and the last is the MySQL DB that share the 2 sip servers. The problem is that, when a phone register it the other server delete the register in table location like if it detect that the register expires. Anybody have an idea of this type of performence? Thank's advance.. --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Serusers] function for lookup contents in a given column of a given table?
by Ricardo Carvalho 22 Aug '06

22 Aug '06

Hi, Is there any Ser function for lookup contents in a given column of the subscriber table, like the *is_user_in* function that looks up in group table? Thanks, Ricardo.

1 0

[Users] auth_db and avps
by Andreas Granig 22 Aug '06

22 Aug '06

Hi, While authorizing a user, auth_db only loads additional AVPs on success. IMO it would also make sense to load them on failure, so you could easily check if it's a known user with wrong credentials or an unknown user. It would especially be useful for fetching the uuid of the user although authorization fails. Any comments? Andy

2 1

[Users] lot of BYEs missing
by Jayesh Nambiar 22 Aug '06

22 Aug '06

Hi folks, I've been using openser for last one year and its working fine. Recently I upgraded my openser to 1.1 stable release which allows me to have more features. But for the last 15 days I have had many discrepencies in the accounting. Among 40 clients to whom we provide service, there was one client who had around 15,000 INVITEs without BYEs in a single day. The UA is a Cisco Gateway. I am aware of BYEs being missed with openser, but there used to be hardly 2% Byes missing. This is very unusual to me and I suffered huge losses due to this behaviour as all the calls were passed to a pstn gateway. The method of authentication that we use for these clients are IP based authentication. I use the function allow_trusted() to get the clients authenticated and make calls. These calls are sent to some Tier1 carriers for pstn termination. I wanted to take some expert advice on this issue. - Is there any way that the UA can bypass the Openser after sending INVITE to openser even after record routing function is incorporated in the script. - Is there a way that the UA can disconnect calls without sending BYEs to the openser. - Are there many risks involved in IP based authentication? I am using IP based authentication as these clients only want to call, they dont want to be called. - If this was a problem with all the UAs, I would have thought that there must be some problem with my openser configuration, but this is with one particular client. The same Cisco gateway at other places work perfectly with hardly 1-2% BYEs missing which is affordable. This indicates that someone must have intentionaly done this. If someone has come across this problem earlier please share it with me. The query I used to find the INVITEs without BYEs is as follows: SELECT t1.from_uri as orig_number, t1.to_uri as term_number, DATE_FORMAT(t1.timestamp, '%a %b %D %Y, %h:%i:%s %p') as calldate, t1.sip_callid as callid FROM acc t1 WHERE t1.sip_method='INVITE' AND t1.sip_callid NOT IN (SELECT t2.sip_callid FROM acc t2 WHERE t2.sip_method = 'BYE'). Thanks a lot in advance... w/regards, jayesh --------------------------------- Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW

3 2

[Users] avp_check in ONREPLY
by Federico Giannici 22 Aug '06

22 Aug '06

Is there a reason why the avp_check() function cannot be used in the ONREPLY_ROUTE, while it can be used for example in the FAILURE_ROUTE? Using OpenSER 1.1.0 from CVS of a few days ago. Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Users] Q:ABOUT:unixodbc
by Alejandro Sanchez 22 Aug '06

22 Aug '06

Hi evrybody. I want to change my implementation of openser 1.1 thati have in mysql with ORACLE and if someone knows what implicates this change for the modules for example "usrloc" i apreciate if give me some feedback. Thank's. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

3 2

[Users] Error reading radius dictionary - Config Openser - radiusclient - freeradius
by Alberto Pacifico 22 Aug '06

22 Aug '06

I like to connect Openser-1.0.1 with Freeradiuus and radiusclient-ng. I'm following the instructions of: http://www.openser.org/docs/openser-radius-1.0.x.html But when I'm starting the Openser-1.0.1 and the modules (like acc) is being initialized, some errors appears related with " error reading radius dictionary". Thanks Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Users] loading modules PA and JABBER fails
by Chris Robson 22 Aug '06

22 Aug '06

No matter what version of openser I build the PA and Jabber modules will fail when loading up modules in Openser start up. Both modules compile without error. Interesting enough is both these modules do not get installed by default with "make install", so I have to install them by hand. Anyone now what I'm missing or what isnt working. loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/dbtext.so" loadmodule "/usr/local/lib/openser/modules/diversion.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/enum.so" loadmodule "/usr/local/lib/openser/modules/flatstore.so" loadmodule "/usr/local/lib/openser/modules/gflags.so" loadmodule "/usr/local/lib/openser/modules/mangler.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/pa.so" loadmodule "/usr/local/lib/openser/modules/options.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/exec.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/msilo.so" loadmodule "/usr/local/lib/openser/modules/textops.so" #oadmodule "/usr/local/lib/openser/modules/jabber.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so"

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[Users] nat-rtpproxy.cfg
by Mark Kent 22 Aug '06

22 Aug '06

A few months ago I downloaded example config files, as a way to learning openser, but I'm not 100% sure where I got them. One of them has this header: # $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ I just found (re-found?) at onsip.org->downloads->Getting Started, a later version of nat-rtpproxy.cfg with this header: # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ and they appear different only in the loose_route handling: OLD: if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; NEW: if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; Now, who wants to take a stab at explaining why this was changed? Why was ACK checked for in the old version and not the new? (my guess: zero length message body messes with force_rtp_proxy) Why does that INVITE logic appear in the new loose_route handler? What case does that handle that the later "if (method == INVITE)" doesn't handle? Does this have something to do with re-INVITES? For convenience, the two files are here: http://a.mainstreet.net/nat-rtpproxy.5.0.cfg http://a.mainstreet.net/nat-rtpproxy-2006.cfg And do we openser people have our own set of sample config files somewhere on openser.org? Thanks, -mark

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[Users] openSER and SEMS: pb with voicemail when phone is busy or on no answer
by Christophe Irles 22 Aug '06

22 Aug '06

Hi, I want to have voicemail feature when phone is busy or on no answer My configuration file is like that: .. # route block used only to have voicemail if phone is busy or on no answer route[6] { append_branch(); avp_db_load("$ruri", "$avp(email)/$email_scheme"); if(!t_write_unix("/tmp/sems_socket", "voicemail/voicemail_headers")) { t_reply("500", "error contacting sems"); return; } route(5); route(4); route(1); } failure_route[1] { # flagset 26 <=> if phone busy we want a voicemail if ( (isflagset(26)) && (t_check_status("486")) ) { resetflag(26); route(6); return; } # flagset 27 <=> on no answer we want a voicemail if ( (isflagset(27)) && (t_check_status("408")) ){ resetflag(27); route(6); return; } end_media_session(); } Now if I use a "default.template" (used by the voicemail module) with the line: from:voicemail@%domain%, SEMS cannot send correctly a email: (3988) INFO: connect (AmSmtpClient.cpp:96): 127.0.0.1 welcomes us (3988) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: HELO 127.0.0.1 (3988) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 250 christophe.rv.fr (3988) DEBUG: run (AmMail.cpp:119): Mail deamon starting its work (3988) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: mail from: <voicemail(a)172.20.250.1> (3988) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 501 Bad address syntax SEMS is trying to send a mail from "voicemail(a)172.20.250.1" ! The SMTP server doesn't accept an IP and this IP is the IP of the callee SIP device . Strange. It's important to notice that "voicemail" works fine when is used on a phone not registered with the line "from:voicemail@%domain%" in file "default.template" . In this case the SEMS log file is: (4494) INFO: connect (AmSmtpClient.cpp:96): 127.0.0.1 welcomes us (4494) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: HELO 127.0.0.1 (4494) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 250 christophe.rv.fr (4494) DEBUG: run (AmMail.cpp:119): Mail deamon starting its work (4494) DEBUG: send_line (AmSmtpClient.cpp:173): SENDED: mail from: <voicemail(a)christophe.rv.fr> (4494) DEBUG: read_line (AmSmtpClient.cpp:145): RECEIVED: 250 Ok So I suspect the "%domain%" in "default.template" file is not correctly read by the module "voicemail" after a "failure_route" or after a "append_branch". What I must change in my OpenSER configuration file in order to resolve this problem ? NOTE: I'm using sems-0.1.1 in revision 68 Thanks in advance, Christophe

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[Users] Sample config
by Pavel D. Kuzin 22 Aug '06

22 Aug '06

Hello! Can anyone provide sample config to proxy any calls. Thank you. -- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk(a)nodex.ru http://nodex.ru

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[Users] SMS Module (SIP to GSM)
by ade darmawan 22 Aug '06

22 Aug '06

Hi all, please help me.... Nowadays, i've been trying to implemented the SMS Module (SIP to GSM) using OpenSER 1.0.0, but never the lles, the tutorials that had given at www.openser.org are so poor... Could you please help me to find more added tutorial guide to implemented the SMS Module (SIP to GSM) ? I wanna know, what kind of facilities that needed to build this module and make it run correctly (such as DNS,etc), step by step to build this module? i hope your opinion? --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

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[Serusers] Is there a function for lookup contents in a given column of the subscriber table?
by Ricardo Carvalho 22 Aug '06

22 Aug '06

Hi, Is there any Ser function for lookup contents in a given column of the subscriber table, like the *is_user_in* function that looks up in group table? Thanks, Ricardo.

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[Users] default behaviour
by Mark Kent 22 Aug '06

22 Aug '06

As an educational tool, I'm trying to write a pure proxy using openser. This proxy doesn't have users or domains or gateways, it just shuffles messages that happen to land on it. I'm starting with REGISTER where I have simply this: t_on_reply("10"); t_on_failure("10"); t_relay(); The onreply and onfailure callbacks just log stuff. The openser cookbook says: The implicit action after execution of the main route block is to drop the SIP request. To send a reply or forward the request, explicit actions must be called inside the route block. To find what happens in other route blocks (onreply, etc.), I searched the cookbook for "implicit" and learned: implicit action for branch_route is to forward the request. implicit action for onreply_route is to send the reply upstream according to Via header And that's all there is. Is there more written somewhere else? What I noticed is that informational responses (1xx) are not relayed by openser, but Final (200) or Client Error (401) are relayed. Is this conclusion correct? This implicit behaviour seems a little odd to me; I would think that if I've got a t_on_reply out there then I'm saying "Hey, I'll handle it" but in fact I cannot handle it (t_relay is not allowed in an onreply block). So, I must be missing something in my understanding... I'ld appreciate it if someone could clarify for me. Thanks, -mark

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[Users] How do I only INVITE if I am registered?
by Mahesh Paolini-Subramanya 22 Aug '06

22 Aug '06

Consider the case where there are two fones, both registered as the same subscriber (foo(a)bar.com) I have max_contacts == 2, therefore the fones actually register I now add a *third* fone, also registered as foo(a)bar.com Since max_contacts == 2, this fone does *not* register. However, if a user makes a call from this fone, the call will go through. I wish to prevent this. What do I do? Something obvious that I am missing? Or something weird with twiddling with the RURI and using "registered(domain)"? Or heavy AVP mucking around? Cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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RE: [Users] Rewriting Contact HF port
by Justin Pearce 21 Aug '06

21 Aug '06

Thanks, Now that I understand what the received avp is for in the registrar module, that seems to make a lot more sense. I was unaware at the time how to go about doing that, and with the small number of replies I got, the only logical conclusion was to modify some functions that performed similarly that I knew worked 100% (all this because of a deadline). The information is helpful, since I will likely have to use it in a future implementation. Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Monday, August 21, 2006 9:04 AM To: Justin Pearce Cc: users(a)openser.org Subject: Re: [Users] Rewriting Contact HF port Justin, just as an idea, you could upload the contact into an AVP (via pseudo variables) and then use a regexp to change the port part....so, you can do it only via script functions. regards, bogdan Justin Pearce wrote: >Hi Bogdan, > >I knew that Nathelper did some rewriting, but the circumstances required >that the thing record the private IP and a certain known port instead of >the public IP as a contact address (using fix_nated_register() to save >the other address). I wound up borrowing the code from the Nathelper >module and modifying it to accept a port argument and made it a separate >module. This way, it only rewrote the port and not left everything else >untouched. > >Thanks. > >Justin Pearce >Information Technology/Programming >Price Video Productions >JustinP(a)PriceVideo.com >361-572-3810 >800-733-3810 >Fax: 361-572-3894 >www.PriceVideo.com > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Monday, August 21, 2006 6:19 AM >To: Justin Pearce >Cc: Klaus Darilion; users(a)openser.org >Subject: Re: [Users] Rewriting Contact HF port > >Hi Justin, > >if you need save(location) to save as "received" a custom uri, you just >need to set an avp with the uri you need to be saved. See: > http://openser.org/docs/modules/1.1.x/registrar.html#AEN211 >It's exactly what nathelper does. > >regards, >bogdan > >Justin Pearce wrote: > > > >>Thanks, Klaus >> >>It looks like I'll have to make something similar to the nathelper >>module to do what I need. >> >>Justin Pearce >> >>-----Original Message----- >>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >>Sent: Monday, August 07, 2006 1:43 AM >>To: Justin Pearce >>Cc: users(a)openser.org >>Subject: Re: [Users] Rewriting Contact HF port >> >>Justin Pearce wrote: >> >> >> >> >>>I am wondering if there is a good way to rewrite the port in the >>> >>> >>> >>> >>contact >> >> >> >> >>>header in OpenSER 1.0.1 >>> >>> >>> >>> >>1. Why do you want to rewrite the port to fixed port? NAT problems? >> >> >> >> >> >>>before it is stored in the location table?(I'm using the MySQL >>> >>> >>> >>> >>database >> >> >> >> >>>backend). >>> >>>Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" >>>just before I do >>> >>>"save("location");", but that does not seem to be working. >>> >>> >>> >>> >>subst will be applied before the message is forwarded. Thus, save() >> >> >sees > > >>the original Contact header. You either loop the modified message to >> >> >the > > >>SIP proxy again or you write your own function (similar to the function >> >> > > > >>in nathelper module) >> >>regards >>klaus >> >> >> >> >> >>>Suggestions? >>> >>> >>> >>>Justin Pearce >>> >>>Information Technology/Programming >>> >>>Price Video Productions >>> >>>JustinP(a)PriceVideo.com <mailto:JustinP@PriceVideo.com> >>> >>>361-572-3810 >>> >>>800-733-3810 >>> >>>Fax: 361-572-3894 >>> >>>www.PriceVideo.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>---------------------------------------------------------------------- - >> >> >- > > >> >> >> >> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> > > >

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Re: [Users] OpenSER administrator issue
by Mike Williams 21 Aug '06

21 Aug '06

I don't think I'm using them, unless there's a bug in Kmail. Mike On Friday 11 August 2006 01:14, Louis Alexander wrote: > Btw, all emails I'm getting on this thread are blank. Are you guys using > PGP signatures? I think my outlook has an issue with that > > > > _____ > > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > Behalf Of Mike Williams > Sent: Friday, August 11, 2006 12:35 PM > To: users(a)openser.org > Subject: Re: [Users] OpenSER administrator issue > > > > > > > > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you.

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[Serusers] SER TCP Bug
by Curt Moore 21 Aug '06

21 Aug '06

Hello all. I've been doing some experimentation with both CVS HEAD and 0.9.6 of SER and version 1.1.0 of OpenSER. It appears that there is a TCP bug in SER which has been fixed in OpenSER. I've seen various posts in the past on the serdev and the serusers lists regarding this issue but with no definitive solution. Are there any plans to merge some of the great work of the OpenSER developers back into SER, such as this fix for TCP? Below is the logfile I receive when I try to place a call from a UDP to a TCP device using CVS HEAD of SER. The same routing logic in ser.cfg works perfectly in openser.cfg. Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcp_blocking_connect: poll error: flags 18 Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcp_blocking_connect: SO_ERROR (113) No route to host Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcpconn_connect: tcp_blocking_connect failed Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: tcp_send: connect failed Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: msg_send: ERROR: tcp_send failed Aug 17 19:03:33 testbox /usr/sbin/ser[32534]: ERROR: t_forward_nonack: sending request failed Aug 17 19:03:33 sg1test /usr/sbin/ser[32534]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Does anyone have any ideas on how to get this fixed in CVS HEAD of SER? I've tried some backporting of the changes to the tcp_* files from OpenSER to SER but have not had any luck as of yet as I'm not fully sure of how the code is supposed to be handling each case. This is to say that I'm just looking at code differences without really understanding how the code is supposed to work. I'm happy to do anything I can to help get this resolved in SER's CVS HEAD. Thanks! -Curt

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[Users] Relay and Redirect
by Andrew And 21 Aug '06

21 Aug '06

I receive "Redirect" with more than one contact destination, and I need to do fork (parallel). I am using "t_relay()" to do this, but I didn´t see correct relay because, now, the Openser sent Invite only one destination. Is there another command to do relay? Could some one help me?

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RE: [Users] Rewriting Contact HF port
by Justin Pearce 21 Aug '06

21 Aug '06

Hi Bogdan, I knew that Nathelper did some rewriting, but the circumstances required that the thing record the private IP and a certain known port instead of the public IP as a contact address (using fix_nated_register() to save the other address). I wound up borrowing the code from the Nathelper module and modifying it to accept a port argument and made it a separate module. This way, it only rewrote the port and not left everything else untouched. Thanks. Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Monday, August 21, 2006 6:19 AM To: Justin Pearce Cc: Klaus Darilion; users(a)openser.org Subject: Re: [Users] Rewriting Contact HF port Hi Justin, if you need save(location) to save as "received" a custom uri, you just need to set an avp with the uri you need to be saved. See: http://openser.org/docs/modules/1.1.x/registrar.html#AEN211 It's exactly what nathelper does. regards, bogdan Justin Pearce wrote: >Thanks, Klaus > >It looks like I'll have to make something similar to the nathelper >module to do what I need. > >Justin Pearce > >-----Original Message----- >From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >Sent: Monday, August 07, 2006 1:43 AM >To: Justin Pearce >Cc: users(a)openser.org >Subject: Re: [Users] Rewriting Contact HF port > >Justin Pearce wrote: > > >>I am wondering if there is a good way to rewrite the port in the >> >> >contact > > >>header in OpenSER 1.0.1 >> >> > >1. Why do you want to rewrite the port to fixed port? NAT problems? > > > >>before it is stored in the location table?(I'm using the MySQL >> >> >database > > >>backend). >> >>Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" >>just before I do >> >>"save("location");", but that does not seem to be working. >> >> > >subst will be applied before the message is forwarded. Thus, save() sees > >the original Contact header. You either loop the modified message to the > >SIP proxy again or you write your own function (similar to the function >in nathelper module) > >regards >klaus > > > >> >> >>Suggestions? >> >> >> >>Justin Pearce >> >>Information Technology/Programming >> >>Price Video Productions >> >>JustinP(a)PriceVideo.com <mailto:JustinP@PriceVideo.com> >> >>361-572-3810 >> >>800-733-3810 >> >>Fax: 361-572-3894 >> >>www.PriceVideo.com >> >> >> >> >> >> >> >----------------------------------------------------------------------- - > > >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

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Re: [Serusers] Multihomed SER & NAT
by Sebastian Gabris 21 Aug '06

21 Aug '06

Hi all, finally I figured out how to change the sdp - part of the message by using force_rtpproxy. All works fine, I can make calls in both directions, but i still got two problems: 1. If someone from outside(VoIP - Provider) calls my UA, i have a ~10 sec. delay before i hear audio. - i checked my dumpfile, and saw that the delay is between the last 180 Ringing and the 200 OK. (- However RTP - Packets are sent before the 200 OK, after a 183-Message. Dont understand why but I dont think it matters that much.) 2. If the call is terminated from outside(VoIP - Provider) the BYE message is not handled correctly by SER. - the previous messages from the Voip provider have all a via HF with the private IP of my UA. The BYE message doesnt have such a Via. All other Via and RR fields are present. - also the BYE uri is somewaht strange. Instead of sip:123456@privateIPofUA it contains sip:123456@publicIPofSER. Thanks in advance for any help Smile Best regards Sebastian. > -----Ursprüngliche Nachricht----- > Von: "Greger V. Teigre" <greger(a)teigre.com> > Gesendet: 10.08.06 15:41:31 > An: Sebastian Gabris <Sebastian.Gabris(a)web.de> > CC: serusers(a)lists.iptel.org > Betreff: Re: [Serusers] Multihomed SER & NAT fix_nated_contact() is meant for changing the IP address of the Contact > header when SER has a public address and the UA has a private (i.e. > behind NAT). This means that it will rewrite Contact to contain the > source ip address:port found in the UDP/TCP header. > This is probably correct, because I assume the UA cannot be reached > from the outside anyway. If you record-route, subsequent requests > should go through your ser. > > If you have problems with responses not being returned, have a look at > Via headers as they control how answers to i.e an INVITE are routed > back to the UA. You may want to have a look at the rr module double > record route parameter to make sure that SER creates one record-route > for the internal and one for the external interface. > g-) > > Sebastian Gabris wrote: > After reading a bit more mails, I came to two possible problems: > 1. fix_nated_contact() needs a registrar to work properly > I dont use a registrar, since i register at the VoIP - Provider. > or > 2. fix_nated_contact() is not ment to be used on this scenario > Any ideas on this? > Sebastian > > > Hi everybody, > need some help on nattraversal. > i use a multihomed SER, one private IP & one public IP. > MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) > If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address. > Instead o writing down his public IP, SER writes his private IP. > Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information. > After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :) > Here are the important bits of my config: > fork=yes > log_stderror=no mhomed=1 > listen=10.x.x.x > listen=193.x.x.x > loadmodule "/usr/lib/ser/modules/nathelper.so" > modparam("usrloc", "db_mode", 0) > modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > if (nat_uac_test("3")) { > if (method == "REGISTER" || ! search("^Record-Route:")) { > log(2,"LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling > if (method == "INVITE") { > fix_nated_sdp("1"); # Add direction=active to SDP > }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > ... and so on. > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > _____________________________________________________________________ > Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! > http://smartsurfer.web.de/?mc=100071&distributionid=000000000071 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

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[Users] acc db_localtime support missing
by Rodrigo P. Telles 21 Aug '06

21 Aug '06

Hi Folks, I tried to use param "db_localtime" on OpenSER 1.0.1 and 1.1.0 but openser complain with 0(28677) set_mod_param_regex: parameter <db_localtime> not found in module <acc> and does not start. openser.cfg looks like: ... loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("acc", "db_localtime", 1) # use GMT time - as OpenSER doc especify ... .. Is that a missing implementation or some thing wrong is happening? Thanks in advance, Telles

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RE: [Users] Rewriting Contact HF port
by Justin Pearce 21 Aug '06

21 Aug '06

Thanks, Klaus It looks like I'll have to make something similar to the nathelper module to do what I need. Justin Pearce -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Monday, August 07, 2006 1:43 AM To: Justin Pearce Cc: users(a)openser.org Subject: Re: [Users] Rewriting Contact HF port Justin Pearce wrote: > I am wondering if there is a good way to rewrite the port in the contact > header in OpenSER 1.0.1 1. Why do you want to rewrite the port to fixed port? NAT problems? > before it is stored in the location table?(I'm using the MySQL database > backend). > > Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" > just before I do > > "save("location");", but that does not seem to be working. subst will be applied before the message is forwarded. Thus, save() sees the original Contact header. You either loop the modified message to the SIP proxy again or you write your own function (similar to the function in nathelper module) regards klaus > > > > Suggestions? > > > > Justin Pearce > > Information Technology/Programming > > Price Video Productions > > JustinP(a)PriceVideo.com <mailto:JustinP@PriceVideo.com> > > 361-572-3810 > > 800-733-3810 > > Fax: 361-572-3894 > > www.PriceVideo.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] BCP: rewrite every header?
by Mark Kent 21 Aug '06

21 Aug '06

I just debugged the following: A PSTN gateway we use is completely non-responsive when the INVITE has this: From: Mark Kent (desk) <sip:markkent@sip.example.com>;tag=foo Contact: Mark Kent (desk) <sip:markkent@192.0.2.216:5060> but things work OK for this: From: "Mark Kent (desk)" <sip:markkent@sip.example.com>;tag=foo Contact: "Mark Kent (desk)" <sip:markkent@192.0.2.216:5060> and also for this: From: Mark Kent <sip:markkent@sip.example.com>;tag=foo Contact: Mark Kent <sip:markkent@192.0.2.216:5060> Now, I know that the EndUser box should put quotes in there to be safe. RFC3261 says "special characters MUST be in a quoted string to be used within a parameter value." But that's not the point of this post... the point is, should I be rewriting every header? Is that standard practice? BTW, I have the same hangup with the SJphone softphone, but there I think it's because of this: From: "Mark Kent(mac)"<sip:MarkKent@sip.example.com>;tag=foo See, no space between the double-quote and the left-angle-bracket. So, should I be re-writing headers to present a single face, independent of the EndUser boxes, to external SIP gateways? Thanks, -mark P.S. FYI, the box that is not putting in the quotes is a SPA3000 running 3.1.10(GWd). A SPA942 running 4.1.10(e) puts in the quotes.

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[Users] check displayname with proxy-authorizationname
by Hakan YASTI 21 Aug '06

21 Aug '06

Dear All, I want to check the "AUTHORIZATION_USERNAME" at proxy-authorization with the "DISPLAY_NAME" of the user agent.( I want them to be same ). How can I do this ? Regards, Hakan. U 2006/08/17 14:50:10.915465 10.0.0.10:9769 -> 10.0.0.15:5060 INVITE sip:DIALEDNUBER@10.0.0.15 SIP/2.0. To: <sip:DIALEDNUBER@10.0.0.15>. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-880502875-1--d87543-;rport. Call-ID: a21d332a7f23d268. CSeq: 1 INVITE. Contact: <sip:USERNAME@10.0.0.10:9769>. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 229. . v=0. o=- 6672092 6672111 IN IP4 10.0.0.10. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 9764 RTP/AVP 18 101. a=alt:1 1 : 4F88BF12 49561D9D 10.0.0.10 9764. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 2006/08/17 14:50:10.917409 10.0.0.15:5060 -> 10.0.0.10:9769 SIP/2.0 407 Proxy Authentication Required. To: <sip:DIALEDNUBER@10.0.0.15>;tag=bf2a561eb1268c60203a720a0640ec5a.1fcb. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-880502875-1--d87543-;rport=9769. Call-ID: a21d332a7f23d268. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="10.0.0.15", nonce="44e4591e0c9257971b55c6113dbcbbdb18552cd3". Server: OpenSer (1.0.1 (i386/freebsd)). Content-Length: 0. Warning: 392 10.0.0.15:5060 "Noisy feedback tells: pid=1378 req_src_ip=10.0.0.10 req_src_port=9769 in_uri=sip:DIALEDNUBER@10.0.0.15 out_uri=sip:DIALEDNUBER@10.0.0.15 via_cnt==1". . # U 2006/08/17 14:50:10.919273 10.0.0.10:9769 -> 10.0.0.15:5060 ACK sip:DIALEDNUBER@10.0.0.15 SIP/2.0. To: <sip:DIALEDNUBER@10.0.0.15>;tag=bf2a561eb1268c60203a720a0640ec5a.1fcb. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-880502875-1--d87543-;rport. Call-ID: a21d332a7f23d268. CSeq: 1 ACK. Content-Length: 0. . # U 2006/08/17 14:50:10.920445 10.0.0.10:9769 -> 10.0.0.15:5060 INVITE sip:DIALEDNUBER@10.0.0.15 SIP/2.0. To: <sip:DIALEDNUBER@10.0.0.15>. From: DISPLAY_NAME<sip:USERNAME@10.0.0.15>;tag=cb1e6d26. Via: SIP/2.0/UDP 10.0.0.10:9769;branch=z9hG4bK-d87543-460538832-1--d87543-;rport. Call-ID: a21d332a7f23d268. CSeq: 2 INVITE. Contact: <sip:USERNAME@10.0.0.10:9769>. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="AUTHORIZATION_USERNAME",realm="10.0.0.15",nonce="44e4591e0c9257971b55c6113dbcbbdb18552cd3",uri="sip:DIALEDNUBER@10.0.0.15",response="93036675b43665e6c94523bf841b7678",algorithm=MD5. User-Agent: eyeBeam release 3004t stamp 16741. Content-Length: 229. . v=0. o=- 6672092 6672111 IN IP4 10.0.0.10. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 9764 RTP/AVP 18 101. a=alt:1 1 : 4F88BF12 49561D9D 10.0.0.10 9764. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv.

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[Users] doubt in Radius and openser fro accounting...
by raviprakash sunkara 21 Aug '06

21 Aug '06

Hi Users, For Me, in Openser + radius Authorize and Acthentication is Succeed Very Good and Nice, But When some to Account, I'm little sucking here When Uac1 dials to Uac2 , ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ radacctID | AcctSessionID | AcctuniqueID | UserName | AcctStartTime | ActStopTime | calledID | callingID | ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- 1 12h34yy 2334juuw45 Uac1 13:10:00 00:00:00 Uac2 uac1 2 12h34yy ko34ji899 Uac2 13:11:11 13:11:11 Uac1 Uac2 --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- In accountting with openser + radius in should insert only one insert for every call made . But I think it insert only two records only. and AcctuniqueD should me same for each call makes then only the AcctStopTime will properly ..... in openser... -------------------- loose record----------------------------------- if(method=="BYE" || method=="CANCEL") { acc_rad_request("Stop"); unforce_rtp_proxy(); } ---------------------------- record route section------------ if( method=="INVITE" || method=="ACK") { acc_rad_request("Start"); forcr_rtp_proxy(); } plz help me in that .. sorry in my english ....... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

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[Serusers] fr_inv_timer fires some interruption?
by Ricardo Carvalho 21 Aug '06

21 Aug '06

Does modparam("tm", "fr_inv_timer", 5) fires some kind of interruption (like a failure_route) when that time (5seconds) of ringing ends? I wish that happens so my Ser be able to send calls to voicemail after ringing with no answer!! Does anybody know if that happens? Regards, Ricardo.

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Re: [Users] Please help for serweb configuration in openser server
by Ferianto siregar 21 Aug '06

21 Aug '06

Dear sunkara, Thank you very much for your kind-hearted for answering my question, Thank you very much. If I can ask you more, would you mind tell me what is the name the application that I can use like "serweb" in openser server ? (the suitable "serweb" for openser server) because I am building openser server. and would you mind tell me where I can get it? Please help me, Sunkara..Please...Please... Thank you very much...Thank you... Regards, Ferianto raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: Hi siregar... I best Suggest is Not to user SERweb in openser. It application to SER only. not For openser , if U want to configure the serweb in openser U need to modify some changes in serweb and openser tables other things.. Perhaps try it..... Best of luck... On 8/19/06, Ferianto siregar <ferianto_voip(a)yahoo.com> wrote: Dear all, Thanks to all of you for your kind-hearted answering my question. Thank you very much. Now, I am building serweb in openser. So, the client can register to become the member of communication in my openser that I have built. 1. Can "serweb" (from iptel.org) be used in openser server for building the webserver of the openser server? Or openser has it`s own webserver except "serweb"? 2.When I build serweb (from iptel.org), I have some problems. I do hope anybody can help me and give me a suggestion. Please.. The error message that I got from client browser is Fatal error: Failed opening required 'var/www/phplib/db_mysql.inc' (include_path='.:/usr/share/pear') in /var/www/html/serweb/user_interface/prepend.php on line 23 The contain of prepend.php that shows line 23 is require($_PHPLIB["libdir"] . "db_mysql.inc"); _/* Change this to match your database. */ I don`t know what is wrong. I do hope anybody can give me a suggestion. Please help me..Please. Thanks Regards, Ferianto Note: This is the contain of db_mysql.inc that I have configured <?php _/* * Session Management for PHP3 * * Copyright (c) 1998-2000 NetUSE AG * Boris Erdmann, Kristian Koehntopp * * $Id: db_mysql.inc,v 1.1.1.1 2002/09/04 18:19:57 kozlik Exp $ * */ class DB_Sql { _/* public: connection parameters */ var $Host = "localhost"; var $Database = "openser"; var $User = "openser"; var $Password = "openserrw"; _/* public: configuration parameters */ var $Auto_Free = 0; ## Set to 1 for automatic mysql_free_result() var $Debug = 0; ## Set to 1 for debugging messages. var $Halt_On_Error = "yes"; ## "yes" (halt with message), "no" (ignore errors quietly), "report" (ignore errror, but spit a warning) var $Seq_Table = "db_sequence"; _/* public: result array and current row number */ var $Record = array(); var $Row; _/* public: current error number and error text */ var $Errno = 0; var $Error = ""; _/* public: this is an api revision, not a CVS revision. */ var $type = "mysql"; var $revision = "1.2"; _/* private: link and query handles */ var $Link_ID = 0; var $Query_ID = 0; _/* public: constructor */ function DB_Sql($query = "") { $this->query($query); } _/* public: some trivial reporting */ function link_id() { return $this->Link_ID; } function query_id() { return $this->Query_ID; } _/* public: connection management */ function connect($Database = "", $Host = "", $User = "", $Password = "") { _/* Handle defaults */ if ("" == $Database) $Database = $this->Database; if ("" == $Host) $Host = $this->Host; if ("" == $User) $User = $this->User; if ("" == $Password) $Password = $this->Password; --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

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Re: [Users] Please help for serweb configuration in openser server
by Ferianto siregar 21 Aug '06

21 Aug '06

Dear sunkara, Thank you very much for your kind-hearted for answering my question, Thank you very much. If I can ask you more, would you mind tell me what is the name the application that I can use like "serweb" in openser server ? (the suitable "serweb" for openser server) because I am building openser server. and would you mind tell me where I can get it? Please help me, Sunkara..Please...Please... Thank you very much...Thank you... Regards, Ferianto raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: Hi siregar... I best Suggest is Not to user SERweb in openser. It application to SER only. not For openser , if U want to configure the serweb in openser U need to modify some changes in serweb and openser tables other things.. Perhaps try it..... Best of luck... On 8/19/06, Ferianto siregar <ferianto_voip(a)yahoo.com> wrote: Dear all, Thanks to all of you for your kind-hearted answering my question. Thank you very much. Now, I am building serweb in openser. So, the client can register to become the member of communication in my openser that I have built. 1. Can "serweb" (from iptel.org) be used in openser server for building the webserver of the openser server? Or openser has it`s own webserver except "serweb"? 2.When I build serweb (from iptel.org), I have some problems. I do hope anybody can help me and give me a suggestion. Please.. The error message that I got from client browser is Fatal error: Failed opening required 'var/www/phplib/db_mysql.inc' (include_path='.:/usr/share/pear') in /var/www/html/serweb/user_interface/prepend.php on line 23 The contain of prepend.php that shows line 23 is require($_PHPLIB["libdir"] . "db_mysql.inc"); /* Change this to match your database. */ I don`t know what is wrong. I do hope anybody can give me a suggestion. Please help me..Please. Thanks Regards, Ferianto Note: This is the contain of db_mysql.inc that I have configured <?php /* * Session Management for PHP3 * * Copyright (c) 1998-2000 NetUSE AG * Boris Erdmann, Kristian Koehntopp * * $Id: db_mysql.inc,v 1.1.1.1 2002/09/04 18:19:57 kozlik Exp $ * */ class DB_Sql { /* public: connection parameters */ var $Host = "localhost"; var $Database = "openser"; var $User = "openser"; var $Password = "openserrw"; /* public: configuration parameters */ var $Auto_Free = 0; ## Set to 1 for automatic mysql_free_result() var $Debug = 0; ## Set to 1 for debugging messages. var $Halt_On_Error = "yes"; ## "yes" (halt with message), "no" (ignore errors quietly), "report" (ignore errror, but spit a warning) var $Seq_Table = "db_sequence"; /* public: result array and current row number */ var $Record = array(); var $Row; /* public: current error number and error text */ var $Errno = 0; var $Error = ""; /* public: this is an api revision, not a CVS revision. */ var $type = "mysql"; var $revision = "1.2"; /* private: link and query handles */ var $Link_ID = 0; var $Query_ID = 0; /* public: constructor */ function DB_Sql($query = "") { $this->query($query); } /* public: some trivial reporting */ function link_id() { return $this->Link_ID; } function query_id() { return $this->Query_ID; } /* public: connection management */ function connect($Database = "", $Host = "", $User = "", $Password = "") { /* Handle defaults */ if ("" == $Database) $Database = $this->Database; if ("" == $Host) $Host = $this->Host; if ("" == $User) $User = $this->User; if ("" == $Password) $Password = $this->Password; /* establish connection, select database */ if ( 0 == $this->Link_ID ) { $this->Link_ID=mysql_pconnect($Host, $User, $Password); if (!$this->Link_ID) { $this->halt("pconnect($Host, $User, \$Password) failed."); return 0; } if (!@mysql_select_db($Database,$this->Link_ID)) { $this->halt("cannot use database ".$Database); return 0; } } return $this->Link_ID; } /* public: discard the query result */ function free() { @mysql_free_result($this->Query_ID); $this->Query_ID = 0; } /* public: perform a query */ function query($Query_String) { /* No empty queries, please, since PHP4 chokes on them. */ if ($Query_String == "") /* The empty query string is passed on from the constructor, * when calling the class without a query, e.g. in situations * like these: '$db = new DB_Sql_Subclass;' */ return 0; if (!$this->connect()) { return 0; /* we already complained in connect() about that. */ }; # New query, discard previous result. if ($this->Query_ID) { $this->free(); } if ($this->Debug) printf("Debug: query = %s \n", $Query_String); $this->Query_ID = @mysql_query($Query_String,$this->Link_ID); $this->Row = 0; $this->Errno = mysql_errno(); $this->Error = mysql_error(); if (!$this->Query_ID) { $this->halt("Invalid SQL: ".$Query_String); } # Will return nada if it fails. That's fine. return $this->Query_ID; } /* public: walk result set */ function next_record() { if (!$this->Query_ID) { $this->halt("next_record called with no query pending."); return 0; } $this->Record = @mysql_fetch_array($this->Query_ID); $this->Row += 1; $this->Errno = mysql_errno(); $this->Error = mysql_error(); $stat = is_array($this->Record); if (!$stat && $this->Auto_Free) { $this->free(); } return $stat; } /* public: position in result set */ function seek($pos = 0) { $status = @mysql_data_seek($this->Query_ID, $pos); if ($status) $this->Row = $pos; else { $this->halt("seek($pos) failed: result has ".$this->num_rows()." rows."); /* half assed attempt to save the day, * but do not consider this documented or even * desireable behaviour. */ @mysql_data_seek($this->Query_ID, $this->num_rows()); $this->Row = $this->num_rows(); return 0; } return 1; } /* public: table locking */ function lock($table, $mode = "write") { $query = "lock tables "; if (is_array($table)) { while (list($key,$value) = each($table)) { if (!is_int($key)) { // texts key are "read", "read local", "write", "low priority write" $query .= "$value $key, "; } else { $query .= "$value $mode, "; } } $query = substr($query,0,-2); } else { $query .= "$table $mode"; } $res = $this->query($query); if (!$res) { $this->halt("lock() failed."); return 0; } return $res; } function unlock() { $res = $this->query("unlock tables"); if (!$res) { $this->halt("unlock() failed."); } return $res; } /* public: evaluate the result (size, width) */ function affected_rows() { return @mysql_affected_rows($this->Link_ID); } function num_rows() { return @mysql_num_rows($this->Query_ID); } function num_fields() { return @mysql_num_fields($this->Query_ID); } /* public: shorthand notation */ function nf() { return $this->num_rows(); } function np() { print $this->num_rows(); } function f($Name) { if (isset($this->Record[$Name])) { return $this->Record[$Name]; } } function p($Name) { if (isset($this->Record[$Name])) { print $this->Record[$Name]; } } /* public: sequence numbers */ function nextid($seq_name) { $this->connect(); if ($this->lock($this->Seq_Table)) { /* get sequence number (locked) and increment */ $q = sprintf("select nextid from %s where seq_name = '%s'", $this->Seq_Table, $seq_name); $id = @mysql_query($q, $this->Link_ID); $res = @mysql_fetch_array($id); /* No current value, make one */ if (!is_array($res)) { $currentid = 0; $q = sprintf("insert into %s values('%s', %s)", $this->Seq_Table, $seq_name, $currentid); $id = @mysql_query($q, $this->Link_ID); } else { $currentid = $res["nextid"]; } $nextid = $currentid + 1; $q = sprintf("update %s set nextid = '%s' where seq_name = '%s'", $this->Seq_Table, $nextid, $seq_name); $id = @mysql_query($q, $this->Link_ID); $this->unlock(); } else { $this->halt("cannot lock ".$this->Seq_Table." - has it been created?"); return 0; } return $nextid; } /* public: return table metadata */ function metadata($table = "", $full = false) { $count = 0; $id = 0; $res = array(); /* * Due to compatibility problems with Table we changed the behavior * of metadata(); * depending on $full, metadata returns the following values: * * - full is false (default): * $result[]: * [0]["table"] table name * [0]["name"] field name * [0]["type"] field type * [0]["len"] field length * [0]["flags"] field flags * * - full is true * $result[]: * ["num_fields"] number of metadata records * [0]["table"] table name * [0]["name"] field name * [0]["type"] field type * [0]["len"] field length * [0]["flags"] field flags * ["meta"][field name] index of field named "field name" * This last one could be used if you have a field name, but no index. * Test: if (isset($result['meta']['myfield'])) { ... */ // if no $table specified, assume that we are working with a query // result if ($table) { $this->connect(); $id = @mysql_list_fields($this->Database, $table); if (!$id) { $this->halt("Metadata query failed."); return false; } } else { $id = $this->Query_ID; if (!$id) { $this->halt("No query specified."); return false; } } $count = @mysql_num_fields($id); // made this IF due to performance (one if is faster than $count if's) if (!$full) { for ($i=0; $i<$count; $i++) { $res[$i]["table"] = @mysql_field_table ($id, $i); $res[$i]["name"] = @mysql_field_name ($id, $i); $res[$i]["type"] = @mysql_field_type ($id, $i); $res[$i]["len"] = @mysql_field_len ($id, $i); $res[$i]["flags"] = @mysql_field_flags ($id, $i); } } else { // full $res["num_fields"]= $count; for ($i=0; $i<$count; $i++) { $res[$i]["table"] = @mysql_field_table ($id, $i); $res[$i]["name"] = @mysql_field_name ($id, $i); $res[$i]["type"] = @mysql_field_type ($id, $i); $res[$i]["len"] = @mysql_field_len ($id, $i); $res[$i]["flags"] = @mysql_field_flags ($id, $i); $res["meta"][$res[$i]["name"]] = $i; } } // free the result only if we were called on a table if ($table) { @mysql_free_result($id); } return $res; } /* public: find available table names */ function table_names() { $this->connect(); $h = @mysql_query("show tables", $this->Link_ID); $i = 0; while ($info = @mysql_fetch_row($h)) { $return[$i]["table_name"] = $info[0]; $return[$i]["tablespace_name"] = $this->Database; $return[$i]["database"] = $this->Database; $i++; } @mysql_free_result($h); return $return; } /* private: error handling */ function halt($msg) { $this->Error = @mysql_error($this->Link_ID); $this->Errno = @mysql_errno($this->Link_ID); if ($this->Halt_On_Error == "no") return; $this->haltmsg($msg); if ($this->Halt_On_Error != "report") die("Session halted."); } function haltmsg($msg) { printf("</td></tr></table>Database error: %s \n", $msg); printf("MySQL Error: %s (%s) \n", $this->Errno, $this->Error); } } ?> --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2ï¿½/min or less. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks and Regards with cheers Sunkara Ravi Prakash Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535 --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

1 0

[Users] Media proxy with getting started Document from onsip.org
by ram 21 Aug '06

21 Aug '06

Hi As per the hello world Document from Onsip.orf ( getting started) iam working with Media proxy with same config i have installed OpenSER 1.1.0 with mysql 4.3 on Cent OS, open SER runnning on Public IP. I have 3 clients Client one Pulic IP Client 2 and client 3 behind NAT ( Linksys Router) Client1 - Sipura Client 2- LeadTek Client 3 my laptop X-Lite Soft Phone the problem here is, when ever iam making call between NATed Clients, some time media proxy using some time not when ever media proxy not using then one way voice and call also not disconnecting, any solution, or suggestion Ram

1 0

[Serusers] provisioning Cisco Phone
by Mahatma Sarasvati 20 Aug '06

20 Aug '06

What solutions are available for avoiding manual editing and saving of Cisco .cnf files for individual phones while using SER or OpenSER? I've got 50-80 phones in my organization and I'm tired of manually managing the phone .cnf files I'm imagining some piece of code that auto-generates the files on command or on a schedule based on entries in the MySQL database. Maybe a column can be added to the Subscriber table or somewhere that holds the MAC address of the phone? Thanks, M

1 0

[Serusers] problem with acc module settings
by Andy Thomas 20 Aug '06

20 Aug '06

Hi, I cant start Ser 0.9.6 I get the following error when I start it up 0(4628) set_mod_param_regex: parameter <db_url> not found in module <acc> 0(4628) parse error (87,20-21): Can't set module parameter 0(4628) set_mod_param_regex: parameter <db_missed_flag> not found in module <acc> 0(4628) parse error (92,19-20): Can't set module parameter 0(4628) set_mod_param_regex: parameter <db_flag> not found in module <acc> 0(4628) parse error (93,19-20): Can't set module parameter ERROR: bad config file (3 errors) Here's my settings for the acc module - can anyone see whats wrong with it?? modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_fmt", "fimos") modparam("acc", "db_missed_flag", 2) modparam("acc", "db_flag", 1)

4 5

[Users] Please help for serweb configuration in openser server
by Ferianto siregar 19 Aug '06

19 Aug '06

Dear all, Thanks to all of you for your kind-hearted answering my question. Thank you very much. Now, I am building serweb in openser. So, the client can register to become the member of communication in my openser that I have built. 1. Can "serweb" (from iptel.org) be used in openser server for building the webserver of the openser server? Or openser has it`s own webserver except "serweb"? 2.When I build serweb (from iptel.org), I have some problems. I do hope anybody can help me and give me a suggestion. Please.. The error message that I got from client browser is Fatal error: Failed opening required 'var/www/phplib/db_mysql.inc' (include_path='.:/usr/share/pear') in /var/www/html/serweb/user_interface/prepend.php on line 23 The contain of prepend.php that shows line 23 is require($_PHPLIB["libdir"] . "db_mysql.inc"); /* Change this to match your database. */ I don`t know what is wrong. I do hope anybody can give me a suggestion. Please help me..Please. Thanks Regards, Ferianto Note: This is the contain of db_mysql.inc that I have configured <?php /* * Session Management for PHP3 * * Copyright (c) 1998-2000 NetUSE AG * Boris Erdmann, Kristian Koehntopp * * $Id: db_mysql.inc,v 1.1.1.1 2002/09/04 18:19:57 kozlik Exp $ * */ class DB_Sql { /* public: connection parameters */ var $Host = "localhost"; var $Database = "openser"; var $User = "openser"; var $Password = "openserrw"; /* public: configuration parameters */ var $Auto_Free = 0; ## Set to 1 for automatic mysql_free_result() var $Debug = 0; ## Set to 1 for debugging messages. var $Halt_On_Error = "yes"; ## "yes" (halt with message), "no" (ignore errors quietly), "report" (ignore errror, but spit a warning) var $Seq_Table = "db_sequence"; /* public: result array and current row number */ var $Record = array(); var $Row; /* public: current error number and error text */ var $Errno = 0; var $Error = ""; /* public: this is an api revision, not a CVS revision. */ var $type = "mysql"; var $revision = "1.2"; /* private: link and query handles */ var $Link_ID = 0; var $Query_ID = 0; /* public: constructor */ function DB_Sql($query = "") { $this->query($query); } /* public: some trivial reporting */ function link_id() { return $this->Link_ID; } function query_id() { return $this->Query_ID; } /* public: connection management */ function connect($Database = "", $Host = "", $User = "", $Password = "") { /* Handle defaults */ if ("" == $Database) $Database = $this->Database; if ("" == $Host) $Host = $this->Host; if ("" == $User) $User = $this->User; if ("" == $Password) $Password = $this->Password; /* establish connection, select database */ if ( 0 == $this->Link_ID ) { $this->Link_ID=mysql_pconnect($Host, $User, $Password); if (!$this->Link_ID) { $this->halt("pconnect($Host, $User, \$Password) failed."); return 0; } if (!@mysql_select_db($Database,$this->Link_ID)) { $this->halt("cannot use database ".$Database); return 0; } } return $this->Link_ID; } /* public: discard the query result */ function free() { @mysql_free_result($this->Query_ID); $this->Query_ID = 0; } /* public: perform a query */ function query($Query_String) { /* No empty queries, please, since PHP4 chokes on them. */ if ($Query_String == "") /* The empty query string is passed on from the constructor, * when calling the class without a query, e.g. in situations * like these: '$db = new DB_Sql_Subclass;' */ return 0; if (!$this->connect()) { return 0; /* we already complained in connect() about that. */ }; # New query, discard previous result. if ($this->Query_ID) { $this->free(); } if ($this->Debug) printf("Debug: query = %s \n", $Query_String); $this->Query_ID = @mysql_query($Query_String,$this->Link_ID); $this->Row = 0; $this->Errno = mysql_errno(); $this->Error = mysql_error(); if (!$this->Query_ID) { $this->halt("Invalid SQL: ".$Query_String); } # Will return nada if it fails. That's fine. return $this->Query_ID; } /* public: walk result set */ function next_record() { if (!$this->Query_ID) { $this->halt("next_record called with no query pending."); return 0; } $this->Record = @mysql_fetch_array($this->Query_ID); $this->Row += 1; $this->Errno = mysql_errno(); $this->Error = mysql_error(); $stat = is_array($this->Record); if (!$stat && $this->Auto_Free) { $this->free(); } return $stat; } /* public: position in result set */ function seek($pos = 0) { $status = @mysql_data_seek($this->Query_ID, $pos); if ($status) $this->Row = $pos; else { $this->halt("seek($pos) failed: result has ".$this->num_rows()." rows."); /* half assed attempt to save the day, * but do not consider this documented or even * desireable behaviour. */ @mysql_data_seek($this->Query_ID, $this->num_rows()); $this->Row = $this->num_rows(); return 0; } return 1; } /* public: table locking */ function lock($table, $mode = "write") { $query = "lock tables "; if (is_array($table)) { while (list($key,$value) = each($table)) { if (!is_int($key)) { // texts key are "read", "read local", "write", "low priority write" $query .= "$value $key, "; } else { $query .= "$value $mode, "; } } $query = substr($query,0,-2); } else { $query .= "$table $mode"; } $res = $this->query($query); if (!$res) { $this->halt("lock() failed."); return 0; } return $res; } function unlock() { $res = $this->query("unlock tables"); if (!$res) { $this->halt("unlock() failed."); } return $res; } /* public: evaluate the result (size, width) */ function affected_rows() { return @mysql_affected_rows($this->Link_ID); } function num_rows() { return @mysql_num_rows($this->Query_ID); } function num_fields() { return @mysql_num_fields($this->Query_ID); } /* public: shorthand notation */ function nf() { return $this->num_rows(); } function np() { print $this->num_rows(); } function f($Name) { if (isset($this->Record[$Name])) { return $this->Record[$Name]; } } function p($Name) { if (isset($this->Record[$Name])) { print $this->Record[$Name]; } } /* public: sequence numbers */ function nextid($seq_name) { $this->connect(); if ($this->lock($this->Seq_Table)) { /* get sequence number (locked) and increment */ $q = sprintf("select nextid from %s where seq_name = '%s'", $this->Seq_Table, $seq_name); $id = @mysql_query($q, $this->Link_ID); $res = @mysql_fetch_array($id); /* No current value, make one */ if (!is_array($res)) { $currentid = 0; $q = sprintf("insert into %s values('%s', %s)", $this->Seq_Table, $seq_name, $currentid); $id = @mysql_query($q, $this->Link_ID); } else { $currentid = $res["nextid"]; } $nextid = $currentid + 1; $q = sprintf("update %s set nextid = '%s' where seq_name = '%s'", $this->Seq_Table, $nextid, $seq_name); $id = @mysql_query($q, $this->Link_ID); $this->unlock(); } else { $this->halt("cannot lock ".$this->Seq_Table." - has it been created?"); return 0; } return $nextid; } /* public: return table metadata */ function metadata($table = "", $full = false) { $count = 0; $id = 0; $res = array(); /* * Due to compatibility problems with Table we changed the behavior * of metadata(); * depending on $full, metadata returns the following values: * * - full is false (default): * $result[]: * [0]["table"] table name * [0]["name"] field name * [0]["type"] field type * [0]["len"] field length * [0]["flags"] field flags * * - full is true * $result[]: * ["num_fields"] number of metadata records * [0]["table"] table name * [0]["name"] field name * [0]["type"] field type * [0]["len"] field length * [0]["flags"] field flags * ["meta"][field name] index of field named "field name" * This last one could be used if you have a field name, but no index. * Test: if (isset($result['meta']['myfield'])) { ... */ // if no $table specified, assume that we are working with a query // result if ($table) { $this->connect(); $id = @mysql_list_fields($this->Database, $table); if (!$id) { $this->halt("Metadata query failed."); return false; } } else { $id = $this->Query_ID; if (!$id) { $this->halt("No query specified."); return false; } } $count = @mysql_num_fields($id); // made this IF due to performance (one if is faster than $count if's) if (!$full) { for ($i=0; $i<$count; $i++) { $res[$i]["table"] = @mysql_field_table ($id, $i); $res[$i]["name"] = @mysql_field_name ($id, $i); $res[$i]["type"] = @mysql_field_type ($id, $i); $res[$i]["len"] = @mysql_field_len ($id, $i); $res[$i]["flags"] = @mysql_field_flags ($id, $i); } } else { // full $res["num_fields"]= $count; for ($i=0; $i<$count; $i++) { $res[$i]["table"] = @mysql_field_table ($id, $i); $res[$i]["name"] = @mysql_field_name ($id, $i); $res[$i]["type"] = @mysql_field_type ($id, $i); $res[$i]["len"] = @mysql_field_len ($id, $i); $res[$i]["flags"] = @mysql_field_flags ($id, $i); $res["meta"][$res[$i]["name"]] = $i; } } // free the result only if we were called on a table if ($table) { @mysql_free_result($id); } return $res; } /* public: find available table names */ function table_names() { $this->connect(); $h = @mysql_query("show tables", $this->Link_ID); $i = 0; while ($info = @mysql_fetch_row($h)) { $return[$i]["table_name"] = $info[0]; $return[$i]["tablespace_name"] = $this->Database; $return[$i]["database"] = $this->Database; $i++; } @mysql_free_result($h); return $return; } /* private: error handling */ function halt($msg) { $this->Error = @mysql_error($this->Link_ID); $this->Errno = @mysql_errno($this->Link_ID); if ($this->Halt_On_Error == "no") return; $this->haltmsg($msg); if ($this->Halt_On_Error != "report") die("Session halted."); } function haltmsg($msg) { printf("</td></tr></table>Database error: %s \n", $msg); printf("MySQL Error: %s (%s) \n", $this->Errno, $this->Error); } } ?> --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

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[Serusers] accounting doesn't seem to work?
by Jean-Michel Hiver 19 Aug '06

19 Aug '06

Hi List, I am trying to get SER to run on FreeBSD 6 For some reason, I can't get the acc module to do anything. I have downloaded SER from CVS, changed ./modules/acc/Makefile to enable SQL accounting, and did a gmake bin include_modules="mysql", then untared the resulting .tgz in / SER starts fine, processes SIP message fine, and yields no errors in /var/log/messages. However I can't see any accounting related information in ser's "acc" table (which remains always empty). Any ideas? PS: Here is my ser.cfg (I have replaced the IP addresses with dummy addresses because I don't want them to be archived...). If somebody would be kind enough to take a look at it... debug=3 fork=yes log_stderror=no listen=123.123.123.123 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" ## ACC loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { sl_send_reply("501", "Not Implemented"); break; }; if (method=="OPTIONS") { sl_send_reply("501", "Not Implemented"); break; }; record_route(); if( (method=="INVITE") || (method=="ACK") || (method=="BYE")) { setflag(1); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; route(3); if (method=="INVITE") { route(2); break; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { if (uri=~"^sip:001#") { rewritehostport ("123.123.123.201:5060"); }; if (uri=~"^sip:002#") { rewritehostport ("123.123.123.202:5060"); }; if (uri=~"^sip:003#") { sl_send_reply ("401", "Unauthorized"); break(); }; if (uri=~"^sip:004#") { rewritehostport ("123.123.123.204:5060"); }; route(1); } route[3] { if ( !( (src_ip == 123.123.123.201) || (src_ip == 123.123.123.202) || (src_ip == 123.123.123.204) || 1 ) ) { sl_send_reply("401", "Unauthorized"); break(); }; }

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[Serusers] Different behaviour from different clients, same firewall.
by sip 19 Aug '06

19 Aug '06

What are some things that might create one-way audio issues from different clients, all using STUN, all NATted behind the same firewall? Some UAs, I have to add port-forwarding for to ensure two-way audio (they're behaving as though they're behind a symmetric NAT). Others, I don't have any issues with at all. For instance, X-Lite? Always works. Snom? Requires port-forwarding in order to function correctly 100% of the time (often works, doesn't always). Grandstream (phones and ATAs)? Always works. UTStarCom F3000? Requires port-forwarding to function correctly. SJPhone? Always works. Etc., etc. All these devices are behind the same firewall, so I don't think it's necessarily an issue with the method of NAT that the firewall does. It's something that varies inside the UA. Any ideas what, inside a UA, could cause such disparity? I thought it might be the SDP headers, but I can't honestly tell MUCH difference between them. I'm at a bit of a loss for all the bizarre variables involved. N.

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[Serusers] iptel.org/ser
by Gaurav Kulshreshtha 19 Aug '06

19 Aug '06

Anybody noticed that the site iptel.org/ser has a new look and the ser admin and ser programmer guides (among many other links) are gone. Are these doc still accessible ?? Thanks Gaurav

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Re: [Serusers] SER on Ubuntu Breezy
by Andrey Kouprianov 19 Aug '06

19 Aug '06

Hi, Oh my. My appologies! I've mixed up with Asterisk. What ser version do you want to install, btw? If 0.9.x then plz take a look at INSTALL file in the tar ball. There should not be any problem with compiling like in any other linux system, if you've installed all the needed packages. When you make, 1. make #builds only ser core, equivalent to make ser 2. make all #builds everything 3. make bin include_modules="mysql jabber cpl-c auth_radius group_radius uri_radius postgres pa" #builds selective modules (you'll definitely need mysql and the rest is up to you) 4. Refer to INSTALL file for more options. Btw, since Ubuntu is Debian based, then also take a look at the "requirements" section for Debian. 5. Make sureto install MySQL first. I used 4.1.x for many installations so far and it seems to be working really fine. Then make install. This should install ser into default directory ( /usr/local ) with configuration file inside the /usr/local/etc/ser/ and shell scripts with binaries in /usr/local/sbin/. Andrey. On 8/19/06, Eddie Johnson Jr <ilo(a)ilo-infosystems.us> wrote: > Are you saying in this e-mail that a package exist in the repository because > I have checked by searching SER and Sip Express Router and nothing. Any > suggestions will be greatly appreciated. > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov > Sent: Thursday, August 17, 2006 10:57 PM > To: serusers(a)iptel.org > Subject: Re: [Serusers] SER on Ubuntu Breezy > > Hi, > > A friend of mine has installed SER on Ubuntu 5.10 before from Ubuntu's > package list. I dont know the details, but it should be fairly easy. > Flex and bison MUST be installed on Ubuntu by default. All u need to > install is MySQL server first, and then SER. Ubuntu should take care > of the rest or at least tell u which package is missing. > > On 8/17/06, Eddie Johnson Jr <ilo(a)ilo-infosystems.us> wrote: > > > > > > > > Has anyone done an install of SER on Ubuntu 5.10. I need some initial > setup > > points and I can take it from there. I installed, flex, bison, > > build-essential, ran make clean nothing, ran make install and I get > makefile > > 80 no directory or file. Where do I go from here? > > > > > > > > Regards, > > > > > > > > iloadmin > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Users] avp_write in 1.1.0
by Stuart Clark 18 Aug '06

18 Aug '06

Hey all.. First time poster, but long time reader. Thanks to all that contribute to this excellent project, this software is very flexible! I just wanted to mention that while upgrading my config to 1.1.0, I found that.. avp_write("$ruri/username", "$avp(myuser)") or avp_write("$ru/username", "$avp(myuser)") does not work as expected, and does not return just the username. Instead I needed to change it to.. avp_write("$rU", "$avp(myuser)") BTW, am I able to update via SVN to a newer bug fixed version of the 1.1.0 branch, without taking on any new things in the HEAD branch? Is it just as simple as this, and I have just not tried it. I wish to use this code in a production environment. Kind regards, Stuart.

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RV: [Serusers] Via RPORT for CANCEL message.
by Ricardo Martinez 18 Aug '06

18 Aug '06

Hello Greger and thanks for your answer I tested sending the CANCEL to route(3), and it seems to be working ok now. But now i'm facing another problem. The OK message for the BYE seems not to be routed back to the NAT'd port, instead is routed to the default port. Please check the file attached (BYE_debug.TXT). As i pointed in my first post, the calling endpoint is NAT'd. Following the BYE sequence seems to be right, it hits the loose_route section and then the route(1), but since the BYE does not contain the rport parameter in the VIA header the OK is routed back to the port advertised in the VIA header. (5061) and not to the received port. Maybe i'm doing something wrong?. I tested a new solution, but i don't know if is correct or is according to the ONSIP document. I added a client_nat_test("3") in the Call Tear Down Section- .......... # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("200.100.100.246:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { if (client_nat_test("3")) { force_rport(); xlog("L_INFO", "[%rm] - From:%fu ; To:%ru ; Call-ID:%ci ; Desde:%is\n - FORCE RPORT ON"); }; end_media_session(); setflag(1); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; ....... Hope that someone could help me here. Thanks Regards, Ricardo Martinez.- -----Mensaje original----- De: Greger V. Teigre [mailto:greger@teigre.com] Enviado el: jueves, 17 de agosto de 2006 4:05 Para: Ricardo Martinez CC: serusers(a)iptel.org Asunto: Re: [Serusers] Via RPORT for CANCEL message. Having a quick look, you may have found a bug in the ONsip.org scripts. A CANCEL is not loose routed (a dialog has not yet been set up) and must be handled the same way as an INVITE, i.e. the only way to find the route for CANCEL is to route the CANCEL the same way you routed the INVITE. However, it seems that you have somehow has done that, as your domain has changed to @gw. in the CANCEL doing downstream (i.e. forwarded by your SER). So, CANCEL should be sent to route(3), the INVITE handler and not route(1). I'm quite sure this was correct in a previous version, we even had some discussions around how to handle it and how to describe it, but it seems that something has happened along the way. g-) Ricardo Martinez wrote: Hello. My main configuration is based on the ser.cfg from the issue 5 plus some modifications included by me. Beside that i'm using version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 11:03:37 Dec 22 2005 with gcc 3.2 I have a question regarding to sending a CANCEL message for an INVITE from a NAT'd endpoint. As far as i know the force_rport() command adds the received IP port to the top most via header in the SIP message, this enables subsequent SIP message to return to the proper port later on in a SIP transaction. My problem is that for a CANCEL message coming from a NAT'd endpoint this command seems not to be working. This is the scenario. NAT'd endpoint : 200.100.100.248 SER : 200.100.100.246 SER-2 : 200.100.100.36 GW : 200.100.100.69 The NAT'd endpoint send an INVITE to the proxy, but then in the middle of the transaction decide to CANCEL the request. As you can see in the "cancel_debug.txt" file included on this mail, the CANCEL message does not contain the "rport" in the Via header, so it seems to be routed back to the default sip port (5060). Is there a way to force the rport in a CANCEL? for a enpoint unable to put the rport by itself? I want to do this according to the onsip document, in a compatible way.. Anyway i made a little test including a force_rport(); in the CANCEL handler .......... # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("200.100.100.246:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); setflag(1); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { force_rport(); route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="ACK") { route(1); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); this solve my problem but it seems not to be so accurate. I hope that someone can help me Thanks! Ricardo Martinez.-

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[Serusers] RTPPROXY doesn't work: HELP!!!
by chiara.alena＠fastwebnet.it 18 Aug '06

18 Aug '06

Hi everybody! I am running SER + RTPPROXY and I am trying to establish a call between a UA with public IP address and a UA with private IP address connecting to the proxy through a Access Point that has an extern ip address that is private too: UA1 (192.168.0.100) <--> (192.168.0.1) AP (192.168.1.1) <--> (192.168.1.2) SER (public IP) <--> (public IP) UA2 I just used the function force_rtp_proxy of nathelper module to force SER to change the sdp body of INVITE and reply messages, and in this way the fields c= and m= of the sdp get changed correctly. I can establish a call and the UAs send RTP packets to the rtpproxy. The problem is that the rtpproxy doesn't forward the RTP packets coming from the UA with public address (UA2) towards the UA with private address (UA1). Are there special flags I have to set in order to make rtpproxy work properly? Do I have to use other functions of nathelper module besides force_rtp_proxy? I am writing my own module for SER and I programmed it in such a way that the force_rtp_proxy function is called (not from the configuration file but from the module itself) for every INVITE request and reply, so that every RTP stream has to pass through the rtpproxy. If I try to use 2 UAs with public addresses everything works properly... Last question: how can I run rtpproxy in debug mode? And where can I see the log files??? Please someone help me! Thanks a lot! Chiara

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[Users] tcp_blocking_connect() in timeout
by Federico Giannici 18 Aug '06

18 Aug '06

I have just upgraded from 1.0.0 to 1.1.0-snapshot of a couple days ago, and now I get the following errors: Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcpconn_connect: tcp_blocking_connect failed Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcp_send: connect failed Aug 18 14:50:56 eowyn OpenSER[16760]: msg_send: ERROR: tcp_send failed Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR:tm:t_forward_nonack: sending request failed I'm using OpenBSD 3.9-stable amd64. What could be the problem? Thanks. P.S. It could be a good idea to add in the 1.0.x to 1.1.x upgrade web page that the "module_exports" structure has changed. P.S. 2 My system is a 64 bit one, but the standard library directory is the usual "lib" and NOT "lib64". -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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FW: [Serusers] Connect Ser on a Asterisk
by Richard C. Thompson 18 Aug '06

18 Aug '06

Sure In sip.conf create a [user] and a [peer] or you can use [friend] if the para's are the same [SER] type=user disallow=all allow=g729 allow=ulaw allow=alaw context=default canreinvite=no permit=10.0.0.2/255.255.255.255 host=10.0.0.2 trustrpid=no [SER] type=peer insecure=yes host=10.0.0.2 context=default nat=no canreinvite=no promiscredir=yes disallow=all allow=g729 allow=ulaw trustrpid=no Rick _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Daniel Mirrha Santos Sent: Friday, August 18, 2006 8:48 AM To: Ser Subject: [Serusers] Connect Ser on a Asterisk Hi, I'm using SER like a public switch and I need to register it on a Asterisk server. Does anybody know how could I do that? Regards, Daniel

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Re: [Serusers] How to send Calls from Ser to Asterisk Voicemail?
by Ricardo Carvalho 18 Aug '06

18 Aug '06

I think Ser also can fire some kind of interruption that calls some routine when the call rings for some time. This routine being called should send that call to voicemail. Anyone knows how can I do that in ser.cfg? Thanks, Ricardo. (Also thanks Juan for your post!) Juan wrote: > Hi Ricardo. > SER doesn't care if the call is answered or not. > Asterisk yes. From the little I know about * you will have to setup a > mailbox for each customer so if he/she is not online or doesn't answer the > call it will be redirected after certain amount of time to the mailbox. > Hope this helps. > Regards > > Juan > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ricardo Carvalho > Sent: Thursday, August 17, 2006 1:33 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] How to send Calls from Ser to Asterisk Voicemail? > > How should I configure ser.cfg to enable Ser to send calls to Asterisk > Voicemail when after ringing for some while, the phone isn't answered? > > Thanks, > > Ricardo. > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] Connect Ser on a Asterisk
by Daniel Mirrha Santos 18 Aug '06

18 Aug '06

Hi, I'm using SER like a public switch and I need to register it on a Asterisk server. Does anybody know how could I do that? Regards, Daniel

1 0

[Serusers] SER with multiple Media Proxy
by Abdul Qadir 18 Aug '06

18 Aug '06

Hi, I would like to know is it possible to setup SER with multiple media proxy. If its not possible than please suggest me what is the best way to solve the problem mentioned below. The problem is I have a setup of SER with Media proxy running in Singapore and I also have some users in US. But I want to use one SER, which is running in singapore, since it connects with DB and doing the AAA. And my users in US also register on the same proxy. But when my US users make IP to IP calls since in current setup their media is first coming to singapore than going to US, they face a latency and poor quality. How can I avoid this without running multiple instances of SER. What I plan to do is setup another media proxy in US and depending on the Nated contact of user agents I give them appropriate media proxy address. Thanking you in advance, Best Regards, Abdul Qadir Best Regards, Abdul Qadir --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta.

4 3

[Users] I have problem in serweb. Please help me..Please..
by Ferianto siregar 18 Aug '06

18 Aug '06

Dear all, First of all, I would like to say thanks to all of you who has been helped me, so I have succeed to build my openser server. Thank you very much. Thanks Now, I am building a serweb. So, the client can register easily from their computer to the openser server. So, they can join the voip communication by using the openser server that I have been built. But, When I tried to build the serweb, I have a problem. This is the error message that I have received from the client browser: Parse error: parse error, unexpected T_VARIABLE in /var/www/html/serweb/config.php on line 217 Fatal error: Failed opening required 'var/www/phplib/db_mysql.inc' (include_path='.:/usr/share/pear') in /var/www/html/serweb/user_interface/prepend.php on line 23 What`s wrong in the configuration? I do hope anyone can help me..Please.. 2. can openser use serweb? because as I know, serweb is used for SER. or the openser has it`s own "serweb"? can anybody can tell me what is it? What`s wrong? I do hope anybody can help me..Please help me..Please Thanks you very much Regards, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

3 2

[Users] I have problem in serweb. Please help me..Please..
by Ferianto siregar 18 Aug '06

18 Aug '06

Dear all, First of all, I would like to say thanks to all of you who has been helped me, so I have succeed to build my openser server. Thank you very much. Thanks Now, I am building a serweb. So, the client can register easily from their computer to the openser server. So, they can join the voip communication by using the openser server that I have been built. But, When I tried to build the serweb, I have a problem. This is the error message that I have received from the client browser: Parse error: parse error, unexpected T_VARIABLE in /var/www/html/serweb/config.php on line 217 Fatal error: Failed opening required 'var/www/phplib/db_mysql.inc' (include_path='.:/usr/share/pear') in /var/www/html/serweb/user_interface/prepend.php on line 23 What`s wrong in the configuration? I do hope anyone can help me..Please.. 2. can openser use serweb? because as I know, serweb is used for SER. or the openser has it`s own "serweb"? can anybody can tell me what is it? Note: This is the configuration in serweb that I have made. I am so sorry because the configuration is too long. Sorry. 1. config.php <? /* * $Id: config.php,v 1.30 2003/12/12 22:03:10 jiri Exp $ */ class Csub_not { var $uri, $desc; function Csub_not($uri, $desc){ $this->uri=$uri; $this->desc=$desc; } } class CREG_list_item { var $reg, $label; function CREG_list_item($reg, $label){ $this->reg=$reg; $this->label=$label; } } class Capplet_params { var $name, $value; function Capplet_params($name, $value){ $this->name=$name; $this->value=$value; } } class Ctab{ var $name, $page, $enabled; function Ctab($enabled, $name, $page){ $this->name=$name; $this->page=$page; $this->enabled=$enabled; } } class Cconfig { var $db_host; var $db_name; var $db_user; var $db_pass; var $table_subscriber; var $table_pending; var $table_grp; var $table_aliases; var $table_location; var $table_missed_calls; var $table_accounting; var $table_phonebook; var $table_event; var $table_netgeo_cache; var $table_ser_mon; var $table_ser_mon_agg; var $table_message_silo; var $table_voice_silo; var $voice_silo_dir; var $greetings_spool_dir; var $show_voicemail_acl; var $show_voice_silo; var $enable_dial_voicemail; var $setup_jabber_account; var $enable_test_firewall; var $stun_applet_width; var $stun_applet_height; var $stun_class; var $stun_archive; var $stun_server; var $stun_port; var $stun_retransmit; var $stun_sourceport; var $grp_values; var $realm; var $domainname; var $first_alias_number; var $new_alias_expires; var $new_alias_q; var $new_alias_callid; var $new_alias_cseq; var $pre_uid_expires; var $psignature; var $web_contact; var $fifo_server; var $reply_fifo_filename; var $reply_fifo_path; var $fifo_aliases_table; var $ul_table; var $ul_priority; var $ul_replication; var $im_length; var $default_domain; var $root_path; var $root_uri; var $img_src_path; var $js_src_path; var $style_src_path; var $zonetab_file; var $charset; var $default_width; var $num_of_showed_items; var $max_showed_rows; var $enable_tabs; var $sub_not; var $mail_header_from; var $mail_forgot_pass; var $forgot_pass_subj; var $mail_register; var $register_subj; // web pages which should be virtually included in beginning and // end of every serweb page var $prolog; var $separator; var $epilog; var $terms_and_conditions; var $metar_event_uri; var $metar_from_sip_uri; var $metar_na_message; var $ser_moni_marginal_period_length; var $ser_moni_aggregation_interval; var $clear_text_pw; var $ul_multidomain; var $ctd_target; var $ctd_uri; function Cconfig(){ //////////////////////////////////////////////////////////////// // configure database /* these are the defaults with which SER installs; if you changed the SER account for MySQl, you need to update here */ $this->db_host="202.95.149.250"; //database host $this->db_name="openser"; //database name $this->db_user="openser"; //database conection user $this->db_pass="openserrw"; //database conection password /* ------------------------------------------------------------*/ /* basic local configuration options */ /* ------------------------------------------------------------*/ /* you need to align these values to your local server settings */ /* the web path bellow which serweb's "admin" and "user_interface" directories begin to spread; If set up to be in root (http://www/admin ...), set just "/" here. Set a specific path otherwise, e.g., "/iptel/html/". Don't forget trailing slash. Hint: if path incorrect, image buttons do not show up */ $this->root_path="/serweb/"; /* roo uri of your server */ /* $this->root_uri="http://192.168.2.16"; */ $this->root_uri="http://".$_SERVER['SERVER_NAME']; /* where is your zone file on your server ? */ $this->zonetab_file = "/usr/share/zoneinfo/zone.tab"; /* serweb will send confirmation emails and SIP IMs -- what sender address should it claim ? should appear in them ? */ $this->mail_header_from="registrar(a)pcr.ac.id"; $this->web_contact="sip:daemon@202.95.149.250"; /* spool directory with voicemail messages */ $this->voice_silo_dir = '/var/spool/voicemail/'; /* directory with voicemail greetings */ $this->greetings_spool_dir = '/var/greetings/'; /* serweb talks to SER via FIFO -- this is FIFO's name, it must have the same value as SER's fifo config param */ $this->fifo_server="/tmp/openser_fifo"; /* these are absolute web paths to HTML documents surrounding serweb pages -- these may typically include banner, trailers, and whatever else appropriate to your web design; make sure the values point to existing files; the files should include at least: prolog: <body> or <body><h1>, etc. separator: may be empty, or </h1><hr> etc. epilog: </body> */ $this->prolog="/serweb/prolog.html"; $this->separator="/serweb/separator.html"; $this->epilog="/serweb/epilog.html"; /* content of html <title> tag */ $this->title="sip.pcr.ac.id, the IP Telephony Site"; /* your domain name */ $this->realm="pcr.ac.id" $this->domainname="pcr.ac.id" $this->default_domain="pcr.ac.id" ereg_replace( "(www\.|sip\.)?(.*)", "\\2", $_SERVER['SERVER_NAME']); /* initial nummerical alias for new subscriber -- don't forget to align your SER routing script to it ! */ $this->first_alias_number=82000; /* info email address */ $this->infomail = "info(a)pcr.ac.id"; /* email address for questions concerning registration */ $this->regmail = "registrar(a)pcr.ac.id"; /* alternate development settings ... $this->zonetab_file = "d:/data/http/iptel/_data/zone.tab"; $this->root_path="/~iptel/"; $this->mail_header_from="php.kk(a)kufr.cz"; $this->voice_silo_dir = 'c:/temp/'; $this->greetings_spool_dir = 'c:/temp/'; $this->fifo_server="d:/temp/tmp"; //path to fifo server */ /* ------------------------------------------------------------*/ /* serweb appearance */ /* ------------------------------------------------------------*/ /* which tabs should show in user's profile ? those set to false by default are experimental features which have not been tested yet */ $this->enable_tabs[1]=true; //enable tab my account $this->enable_tabs[2]=true; //enable tab phonebook $this->enable_tabs[3]=true; //enable tab missed calls $this->enable_tabs[4]=true; //enable tab accounting $this->enable_tabs[5]=true; //enable tab send IM $this->enable_tabs[6]=false; //enable tab notification subscription $this->enable_tabs[7]=true; //enable tab message store /* admin tabs definitions Ctab (enabled, name_of_tab, php_script) */ $this->admin_tabs[]=new Ctab (true, "users", "users.php"); $this->admin_tabs[]=new Ctab (true, "server monitoring", "openser_moni.php"); $this->default_width=564; //width of usable area $this->num_of_showed_items=20; /* num of showed items in the list of users */ $this->max_showed_rows=50; /* maximum of showed items in "user find" */ /* show test firewall/NAT button (see also advanced FW/NAT settings bellow */ $this->enable_test_firewall=false; /* experimental/incomplete features turned off: voicemail and set up a jabber account for each new SIP user too */ $this->show_voicemail_acl=false; /* show voicemail in ACL list */ $this->show_voice_silo=false; /* show voice messages in silo too */ $this->enable_dial_voicemail=false; $this->enable_tabs[8]=false; //enable tab voicemail $this->setup_jabber_account=false; /* ------------------------------------------------------------*/ /* ACLs */ /* ------------------------------------------------------------*/ /* there may be SIP contacts which you wish to prevent from being added through serweb to avoid loops, forwarding to unfriendly domains, etc. use these REGexs to specify which contacts you do not wish; the first value includes banned REs, the second displays error message displayed to users if they attempt to introduce a banned contact */ $this->denny_reg[]=new CREG_list_item("gateway\.pcr.ac.id$","local forwarding prohibited"); $this->denny_reg[]=new CREG_list_item("gateway","gateway contacts prohibited"); /* SER configuration script may check for group membership of users identified using digest authentication; e.g., it may only allow international calls to callers who are members of 'int' group; this is a list of groups that serweb allows to set -- they need to correspond to names of groups used in SER's membership checks */ $this->grp_values[]="voicemail"; $this->grp_values[]="ld"; $this->grp_values[]="local"; $this->grp_values[]="int"; /* ------------------------------------------------------------*/ /* text */ /* ------------------------------------------------------------*/ /* human-readable text containing messages displayed to users in web or sent by email; you may need to hire a lawyer , a word-smith, a diplomat or a translator to get it right :) */ /* text of password-reminder email */ $this->forgot_pass_subj="your login information"; $this->mail_forgot_pass="Hello,\n". "now you can access to your account at the folowing URL within 1 hour:\n". $this->root_uri.$this->root_path."user_interface/my_account.php?#session#\n\n". "We recommend change your password after you login\n\n"; /* text of confirmation email sent during account registration */ $this->register_subj="Your ".$this->domainname." Registration"; $this->mail_register= "Thank you for registering with ".$this->domainname.".\n\n". "We are reserving the following SIP address for you: #sip_address#\n\n". "To finalize your registration please check the following URL within ". "24 hours:\n". $this->root_uri.$this->root_path."user_interface/reg/confirmation.php?nr=#confirm#\n\n". "(If you confirm later you will have to re-register.)\n\n". "Windows Messenger users may look at additional configuration hints at\n". "http://www.iptel.org/phpBB/viewtopic.php?topic=11&forum=1&0\n"; /* terms and conditions as they appear on the subscription webpage */ $this->terms_and_conditions= "BY PRESSING THE 'I ACCEPT' BUTTON, YOU (HEREINAFTER THE 'USER') ARE ". "STATING THAT YOU AGREE TO ACCEPT AND BE BOUND BY ALL OF THE TERMS AND ". "CONDITIONS OF THIS AGREEMENT. DO NOT PROCEED IF YOU ARE UNABLE TO AGREE". " TO THE TERMS AND CONDITIONS OF THIS AGREEMENT. THESE TERMS AND CONDITIONS ". "OF SERVICE FOR USE OF ".$this->domainname." SIP SERVER (THE 'AGREEMENT')". " CONSTITUTE A LEGALLY BINDING CONTRACT BETWEEN ".$this->domainname. " AND THE ENTITY THAT AGREES TO AND ACCEPTS THESE TERMS AND CONDITIONS. ". "ACCESS TO ".$this->domainname."'s SESSION INITIATION PROTOCOL SERVER ". "('SIP SERVER') IS BEING PROVIDED ON AN 'AS IS' AND 'AS AVAILABLE' BASIS, ". "AND ".$this->domainname." MAKES NO REPRESENTATIONS OR WARRANTIES OF ANY ". "KIND, WHETHER EXPRESS OR IMPLIED, WITH RESPECT TO USER'S ACCESS OF THE ". "SIP SERVER, INCLUDING BUT NOT LIMITED TO WARRANTIES OF MERCHANTABILITY, ". "NONINFRINGEMENT, TITLE OR FITNESS FOR A PARTICULAR PURPOSE. FURTHER, ". $this->domainname." MAKES NO REPRESENTATIONS OR WARRANTIES THAT THE SIP ". "SERVER, OR USER'S ACCESS THERETO, WILL BE AVAILABLE AT ANY GIVEN TIME, ". "OR WILL BE FREE FROM ERRORS, DEFECTS, OMISSIONS, INACCURACIES, OR FAILURES". " OR DELAYS IN DELIVERY OF DATA. USER ASSUMES, AND ".$this->domainname. " DISCLAIM, TOTAL RISK, RESPONSIBILITY, AND LIABILITY FOR USER'S ACCESS TO ". "AND USE OF THE SIP SERVER.\n\n". "Access to ".$this->domainname." SIP Server is being provided on a ". "non-exclusive basis. User acknowledges and understands that ". $this->domainname." SIP site is in a developmental stage and that ". $this->domainname." makes no guarantees regarding the availability or ". "functionality thereof. User may not sublicense its access rights to the ". "SIP Server to any third party. \n\n". "USER AGREES TO INDEMNIFY, DEFEND AND HOLD iptel.org, ITS AFFILIATES, ". "DIRECTORS, OFFICERS, EMPLOYEES, AGENTS AND LICENSORS HARMLESS FROM AND ". "AGAINST ANY AND ALL CLAIMS, ACTIONS, EXPENSES, LOSSES, AND LIABILITIES ". "(INCLUDING COURTS COSTS AND REASONABLE ATTORNEYS' FEES), ". "ARISING FROM OR RELATING TO THIS AGREEMENT INCLUDING USER'S ACCESS TO ". "AND USE OF THE SIP SERVER TO THE MAXIMUM EXTENT PERMITTED BY APPLICABLE LAW,". " IN NO EVENT SHALL ".$this->domainname." OR ANY OF ITS LICENSORS, BE LIABLE ". "FOR ANY INDIRECT, SPECIAL, PUNITIVE, EXEMPLARY, OR CONSEQUENTIAL DAMAGES, ". "ARISING OUT OF THE ACCESS TO OR USE OF OR INABILITY TO ACCESS OR USE THE ". "SIP SERVER, OR THAT RESULT FROM MISTAKES, OMISSIONS, INTERRUPTIONS, ". "DELETIONS OF FILES, ERRORS, DEFECTS, DELAYS IN TRANSMISSION OR OPERATION OR ". "ANY FAILURE OF PERFORMANCE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH ". "DAMAGES. \n\n". "If User commits, in ".$this->domainname."'s sole determination, a default ". "of these terms and conditions, ".$this->domainname." may immediately ". "terminate User's access to the SIP Server. Furthermore, ".$this->domainname. " reserves the right to discontinue offering access to the SIP Server at any ". "time. \n\n". "User may not assign its rights hereunder without the prior written ". "consent of ".$this->domainname.". User agrees to comply with all laws, ". "regulations and other legal requirements that apply to these terms and ". "conditions. \n\n". "If any provision of this Agreement is held to be unenforceable for any ". "reason, such provision shall be reformed only to the extent necessary to ". "comply with applicable laws, and the remainder shall remain in full force ". "and effect. \n\n". "Any failure of ".$this->domainname." to enforce any provision of this ". "Agreement shall not constitute a waiver of any rights under such provision ". "or any other provision of this Agreement. \n\n". "USER ACKNOWLEDGES THAT IT HAS READ THIS AGREEMENT, UNDERSTANDS IT, AND ". "AGREES THAT IT IS THE COMPLETE AND EXCLUSIVE STATEMENT OF THE ENTIRE ". "AGREEMENT BETWEEN COMPANY AND ".$this->domainname." WITH RESPECT TO THE ". "SUBJECT MATTER HEREIN, AND SUPERSEDES ALL PRIOR AND CONTEMPORANEOUS ". "PROPOSALS, DISCUSSIONS, AGREEMENTS, UNDERSTANDINGS, AND COMMUNICATIONS, ". "WHETHER WRITTEN OR ORAL AND MAY BE AMENDED ONLY IN A WRITING EXECUTED BY ". "BOTH USER AND ".$this->domainname.". \n\n"; /* =========================================================== */ /* ADVANCED SETTINGS */ /* =========================================================== */ /* ------------------------------------------------------------*/ /* applications (experimental) */ /* ------------------------------------------------------------*/ /* subscribe-notify -- list of events to which a user can subscribe and is then notified with an instant message, if they occur; experimental */ $this->sub_not[]=new Csub_not("sip:weather@pcr.ac.id". ";type=temperature;operator=lt;value=0","temperature is too low"); $this->sub_not[]=new Csub_not("sip:weather@pcr.ac.id". ";type=wind;operator=gt;value=10","wind is too fast"); $this->sub_not[]=new Csub_not("sip:weather@pcr.ac.id;". "type=pressure;operator=lt;value=1000","pressure is too low"); $this->sub_not[]=new Csub_not("sip:weather@pcr.ac.id;type=metar", "send METAR data"); /* metar wheather application */ //this is an identificator in event table for sending METAR data $this->metar_event_uri="sip:weather@pcr.ac.id;type=metar"; //from header in sip message $this->metar_from_sip_uri="sip:daemon@pcr.ac.id"; // N/A message - is sended to user when we can't get his location or METAR data $this->metar_na_message="sorry we can't get your location or METAR data for you"; /* ------------------------------------------------------------*/ /* configure FW/NAT detection applet */ /* ------------------------------------------------------------*/ /* the applet is used to detect whether user is behind firewall or NAT */ //width of NAT detection applet $this->stun_applet_width=350; //height of NAT detection applet $this->stun_applet_height=100; //starting class of NAT detection applet $this->stun_class="STUNClientApplet.class"; //jar archive with NAT detection applet - optional - you can comment // it if you don't use jar archive $this->stun_archive="STUNClientApplet.jar"; /* applet parameters: */ /* STUN server address - must be same as web server address because the java security manager allows only this one */ $this->stun_applet_param[]=new Capplet_params("server", "sip.pcr.ac.id"); /* STUN server port. The Default value is 1221 - optional - you can comment it if you want use default value */ $this->stun_applet_param[]=new Capplet_params("port", 1221); /* destination port for the first probing attempt -- just set up a simple tcp echo server there; we use the first TCP connection to determine local IP address, which can't be learned from systems setting due to security manager ; default is 5060 */ $this->stun_applet_param[]=new Capplet_params("tcp_dummyport", 5061); /* Number of times to resend a STUN message to a STUN server. The Default is 9 times - optional - you can comment it if you want use default value */ // $this->stun_applet_param[]=new Capplet_params("retransmit", 9); /* Specify source port of UDP packet to be sent from. The Default value is 5000 - optional - you can comment it if you want use default value */ // $this->stun_applet_param[]=new Capplet_params("sourceport", 5000); /* ------------------------------------------------------------*/ /* configure server monitoring */ /* ------------------------------------------------------------*/ /* if you change this values, please delete all data from table "table_ser_mon_agg" and "table_ser_mon" by reason that the aggregated data may be calculated bad if you don't do it */ /* length of marginal period in seconds */ $this->ser_moni_marginal_period_length=60*5; //5 minutes /* length of interval (in seconds) for which will data stored, data older then this interval will be deleted */ $this->ser_moni_aggregation_interval=60*15; //15 minut /* ------------------------------------------------------------*/ /* click to dial */ /* ------------------------------------------------------------*/ /* address of the final destination to which we want to transfer initial CSeq and CallId */ $this->ctd_target="sip:23@202.95.14.250"; /* address of user wishing to initiate conversation */ $this->ctd_uri="sip:44@202.95.149.250"; /* from header for click-to-dial request */ $this->ctd_from = "sip:controller@pcr.ac.id"; /* ------------------------------------------------------------*/ /* Values you typically do NOT want to change unless you know * /* well what you are doing * /* ------------------------------------------------------------*/ /* Unless you used brute-force to change SER table names */ $this->table_subscriber="subscriber"; $this->table_pending="pending"; $this->table_grp="grp"; $this->table_aliases="aliases"; $this->table_location="location"; $this->table_missed_calls="missed_calls"; $this->table_accounting="acc"; $this->table_phonebook="phonebook"; $this->table_event="event"; $this->table_netgeo_cache="netgeo_cache"; $this->table_ser_mon="server_monitoring"; $this->table_ser_mon_agg="server_monitoring_agg"; $this->table_message_silo="silo"; $this->table_voice_silo="voice_silo"; /* these are table names as reffered from script and via FIFO */ $this->ul_table="location"; $this->fifo_aliases_table="aliases"; /* relative paths of serweb tree */ $this->img_src_path = $this->root_path."img/"; $this->js_src_path = $this->root_path."styles/"; $this->style_src_path = $this->root_path."styles/"; $this->user_pages_path = $this->root_path."user_interface/"; $this->admin_pages_path = $this->root_path."admin/"; /* values used for names of reply fifos -- they change radnomly */ $this->reply_fifo_filename="webfifo_".rand(); $this->reply_fifo_path="/tmp/".$this->reply_fifo_filename; /* development value $this->reply_fifo_path="d:/temp/".$this->reply_fifo_filename; */ /* serweb version */ $this->psignature="Web_interface_Karel_Kozlik-0.9"; /* IM paging configuration */ $this->charset="windows-1250"; $this->im_length=1300; /* expiration times, priorities, etc. for usrloc/alias contacts */ $this->new_alias_expires='567648000'; $this->new_alias_q=1.00; $this->new_alias_callid="web_call_id@fox"; $this->new_alias_cseq=1; $this->ul_priority="1.00"; /* replication support ? (a new ser feature) */ $this->ul_replication=1; /* fifo expects usernames AND domainnames ? make sure this option is synchronized with usrloc's use_domain option */ $this->ul_multidomain=0; /* seconds in which expires "get pass session" */ $this->pre_uid_expires=3600; /* is the sql database query for user authentication formed with clear text password or a hashed one; the former is less secure the latter works even if password hash is incorrect, which sometimes happens, when it is calculated from an incorrect domain during installation process */ $this->clear_text_pw=1; } } $config=new Cconfig(); ?> 2. db_mysql.inc <?php /* * Session Management for PHP3 * * Copyright (c) 1998-2000 NetUSE AG * Boris Erdmann, Kristian Koehntopp * * $Id: db_mysql.inc,v 1.1.1.1 2002/09/04 18:19:57 kozlik Exp $ * */ class DB_Sql { /* public: connection parameters */ var $Host = "localhost"; var $Database = "openser"; var $User = "openser"; var $Password = "openserrw"; /* public: configuration parameters */ var $Auto_Free = 0; ## Set to 1 for automatic mysql_free_result() var $Debug = 0; ## Set to 1 for debugging messages. var $Halt_On_Error = "yes"; ## "yes" (halt with message), "no" (ignore errors quietly), "report" (ignore errror, but spit a warning) var $Seq_Table = "db_sequence"; /* public: result array and current row number */ var $Record = array(); var $Row; /* public: current error number and error text */ var $Errno = 0; var $Error = ""; /* public: this is an api revision, not a CVS revision. */ var $type = "mysql"; var $revision = "1.2"; /* private: link and query handles */ var $Link_ID = 0; var $Query_ID = 0; /* public: constructor */ function DB_Sql($query = "") { $this->query($query); } /* public: some trivial reporting */ function link_id() { return $this->Link_ID; } function query_id() { return $this->Query_ID; } /* public: connection management */ function connect($Database = "", $Host = "", $User = "", $Password = "") { /* Handle defaults */ if ("" == $Database) $Database = $this->Database; if ("" == $Host) $Host = $this->Host; if ("" == $User) $User = $this->User; if ("" == $Password) $Password = $this->Password; /* establish connection, select database */ if ( 0 == $this->Link_ID ) { $this->Link_ID=mysql_pconnect($Host, $User, $Password); if (!$this->Link_ID) { $this->halt("pconnect($Host, $User, \$Password) failed."); return 0; } if (!@mysql_select_db($Database,$this->Link_ID)) { $this->halt("cannot use database ".$Database); return 0; } } return $this->Link_ID; } /* public: discard the query result */ function free() { @mysql_free_result($this->Query_ID); $this->Query_ID = 0; } /* public: perform a query */ function query($Query_String) { /* No empty queries, please, since PHP4 chokes on them. */ if ($Query_String == "") /* The empty query string is passed on from the constructor, * when calling the class without a query, e.g. in situations * like these: '$db = new DB_Sql_Subclass;' */ return 0; if (!$this->connect()) { return 0; /* we already complained in connect() about that. */ }; # New query, discard previous result. if ($this->Query_ID) { $this->free(); } if ($this->Debug) printf("Debug: query = %s \n", $Query_String); $this->Query_ID = @mysql_query($Query_String,$this->Link_ID); $this->Row = 0; $this->Errno = mysql_errno(); $this->Error = mysql_error(); if (!$this->Query_ID) { $this->halt("Invalid SQL: ".$Query_String); } # Will return nada if it fails. That's fine. return $this->Query_ID; } /* public: walk result set */ function next_record() { if (!$this->Query_ID) { $this->halt("next_record called with no query pending."); return 0; } $this->Record = @mysql_fetch_array($this->Query_ID); $this->Row += 1; $this->Errno = mysql_errno(); $this->Error = mysql_error(); $stat = is_array($this->Record); if (!$stat && $this->Auto_Free) { $this->free(); } return $stat; } /* public: position in result set */ function seek($pos = 0) { $status = @mysql_data_seek($this->Query_ID, $pos); if ($status) $this->Row = $pos; else { $this->halt("seek($pos) failed: result has ".$this->num_rows()." rows."); /* half assed attempt to save the day, * but do not consider this documented or even * desireable behaviour. */ @mysql_data_seek($this->Query_ID, $this->num_rows()); $this->Row = $this->num_rows(); return 0; } return 1; } /* public: table locking */ function lock($table, $mode = "write") { $query = "lock tables "; if (is_array($table)) { while (list($key,$value) = each($table)) { if (!is_int($key)) { // texts key are "read", "read local", "write", "low priority write" $query .= "$value $key, "; } else { $query .= "$value $mode, "; } } $query = substr($query,0,-2); } else { $query .= "$table $mode"; } $res = $this->query($query); if (!$res) { $this->halt("lock() failed."); return 0; } return $res; } function unlock() { $res = $this->query("unlock tables"); if (!$res) { $this->halt("unlock() failed."); } return $res; } /* public: evaluate the result (size, width) */ function affected_rows() { return @mysql_affected_rows($this->Link_ID); } function num_rows() { return @mysql_num_rows($this->Query_ID); } function num_fields() { return @mysql_num_fields($this->Query_ID); } /* public: shorthand notation */ function nf() { return $this->num_rows(); } function np() { print $this->num_rows(); } function f($Name) { if (isset($this->Record[$Name])) { return $this->Record[$Name]; } } function p($Name) { if (isset($this->Record[$Name])) { print $this->Record[$Name]; } } /* public: sequence numbers */ function nextid($seq_name) { $this->connect(); if ($this->lock($this->Seq_Table)) { /* get sequence number (locked) and increment */ $q = sprintf("select nextid from %s where seq_name = '%s'", $this->Seq_Table, $seq_name); $id = @mysql_query($q, $this->Link_ID); $res = @mysql_fetch_array($id); /* No current value, make one */ if (!is_array($res)) { $currentid = 0; $q = sprintf("insert into %s values('%s', %s)", $this->Seq_Table, $seq_name, $currentid); $id = @mysql_query($q, $this->Link_ID); } else { $currentid = $res["nextid"]; } $nextid = $currentid + 1; $q = sprintf("update %s set nextid = '%s' where seq_name = '%s'", $this->Seq_Table, $nextid, $seq_name); $id = @mysql_query($q, $this->Link_ID); $this->unlock(); } else { $this->halt("cannot lock ".$this->Seq_Table." - has it been created?"); return 0; } return $nextid; } /* public: return table metadata */ function metadata($table = "", $full = false) { $count = 0; $id = 0; $res = array(); /* * Due to compatibility problems with Table we changed the behavior * of metadata(); * depending on $full, metadata returns the following values: * * - full is false (default): * $result[]: * [0]["table"] table name * [0]["name"] field name * [0]["type"] field type * [0]["len"] field length * [0]["flags"] field flags * * - full is true * $result[]: * ["num_fields"] number of metadata records * [0]["table"] table name * [0]["name"] field name * [0]["type"] field type * [0]["len"] field length * [0]["flags"] field flags * ["meta"][field name] index of field named "field name" * This last one could be used if you have a field name, but no index. * Test: if (isset($result['meta']['myfield'])) { ... */ // if no $table specified, assume that we are working with a query // result if ($table) { $this->connect(); $id = @mysql_list_fields($this->Database, $table); if (!$id) { $this->halt("Metadata query failed."); return false; } } else { $id = $this->Query_ID; if (!$id) { $this->halt("No query specified."); return false; } } $count = @mysql_num_fields($id); // made this IF due to performance (one if is faster than $count if's) if (!$full) { for ($i=0; $i<$count; $i++) { $res[$i]["table"] = @mysql_field_table ($id, $i); $res[$i]["name"] = @mysql_field_name ($id, $i); $res[$i]["type"] = @mysql_field_type ($id, $i); $res[$i]["len"] = @mysql_field_len ($id, $i); $res[$i]["flags"] = @mysql_field_flags ($id, $i); } } else { // full $res["num_fields"]= $count; for ($i=0; $i<$count; $i++) { $res[$i]["table"] = @mysql_field_table ($id, $i); $res[$i]["name"] = @mysql_field_name ($id, $i); $res[$i]["type"] = @mysql_field_type ($id, $i); $res[$i]["len"] = @mysql_field_len ($id, $i); $res[$i]["flags"] = @mysql_field_flags ($id, $i); $res["meta"][$res[$i]["name"]] = $i; } } // free the result only if we were called on a table if ($table) { @mysql_free_result($id); } return $res; } /* public: find available table names */ function table_names() { $this->connect(); $h = @mysql_query("show tables", $this->Link_ID); $i = 0; while ($info = @mysql_fetch_row($h)) { $return[$i]["table_name"] = $info[0]; $return[$i]["tablespace_name"] = $this->Database; $return[$i]["database"] = $this->Database; $i++; } @mysql_free_result($h); return $return; } /* private: error handling */ function halt($msg) { $this->Error = @mysql_error($this->Link_ID); $this->Errno = @mysql_errno($this->Link_ID); if ($this->Halt_On_Error == "no") return; $this->haltmsg($msg); if ($this->Halt_On_Error != "report") die("Session halted."); } function haltmsg($msg) { printf("</td></tr></table>Database error: %s \n", $msg); printf("MySQL Error: %s (%s) \n", $this->Errno, $this->Error); } } ?> 3. prepend.php <?php /* * Session Management for PHP3 * * Copyright (c) 1998-2000 NetUSE AG * Boris Erdmann, Kristian Koehntopp * * $Id: prepend.php,v 1.1.1.1 2002/09/04 18:19:41 kozlik Exp $ * */ $_PHPLIB = array(); # Can't control your include path? # Point this to your PHPLIB base directory. Use a trailing "/"! $_PHPLIB["libdir"] = "var/www/phplib/"; require "../config.php"; require "../functions.php"; require "../page.php"; require($_PHPLIB["libdir"] . "db_mysql.inc"); /* Change this to match your database. */ require($_PHPLIB["libdir"] . "ct_sql.inc"); /* Change this to match your data storage container */ require($_PHPLIB["libdir"] . "session.inc"); /* Required for everything below. */ require($_PHPLIB["libdir"] . "auth.inc"); /* Disable this, if you are not using authentication. */ require($_PHPLIB["libdir"] . "perm.inc"); /* Disable this, if you are not using permission checks. */ require($_PHPLIB["libdir"] . "user.inc"); /* Disable this, if you are not using per-user variables. */ /* Additional require statements go below this line */ # require($_PHPLIB["libdir"] . "menu.inc"); /* Enable to use Menu */ /* Additional require statements go before this line */ require($_PHPLIB["libdir"] . "local.inc"); /* Required, contains your local configuration. */ require($_PHPLIB["libdir"] . "page.inc"); /* Required, contains the page management functions. */ ?> I do hope anybody can help me..I am so sorry because this message is too long..Sorry..Thank you very much.. Thank you Regards, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta. --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

1 0

[Serusers] SER on Ubuntu Breezy
by Eddie Johnson Jr 18 Aug '06

18 Aug '06

Has anyone done an install of SER on Ubuntu 5.10. I need some initial setup points and I can take it from there. I installed, flex, bison, build-essential, ran make clean nothing, ran make install and I get makefile 80 no directory or file. Where do I go from here? Regards, iloadmin

2 1

[Users] 'socket' field in location db table
by Christian Schlatter 18 Aug '06

18 Aug '06

hi all, I'm running two openser instances accessing one central MySQL database using the usrloc db-only mode. Now I'm getting warnings in the log like WARNING:usrloc:get_all_db_ucontacts: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring Although this is only a 'warning' level message, I'm a little concerned what effects this could have. Does anybody know what the 'socket' field in the location table is actually good for? thanks, Christian

2 2

[Users] haze about the REGISTER method
by Jeremie Le Hen 17 Aug '06

17 Aug '06

Hi, first of all, excuse me for posting these questions twice, but the previous mail was bloated with too verbose sentences. I'll be straightforward then. 1) Why do all the configuration files I was given to see on the web call record_route() when the method is *not* a REGISTER ? 2) What is the harm of calling fix_nated_contact() in addition to fix_nated_register() in a REGISTER message ? The configuration file [1] carefully avoids doing this. [1] http://voip-info.org/wiki/view/OpenSER+And+RTPProxy Thank you very much. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

3 2

[Serusers] How to send Calls from Ser to Asterisk Voicemail?
by Ricardo Carvalho 17 Aug '06

17 Aug '06

How should I configure ser.cfg to enable Ser to send calls to Asterisk Voicemail when after ringing for some while, the phone isn't answered? Thanks, Ricardo.

2 1

[Serusers] iptel.org problem
by Klaus Darilion 17 Aug '06

17 Aug '06

Hi! The SIP service of iptel.org does not work correctly - calls to other SIP domains do not work (404). regards klaus

2 1

[Users] heders in route & failure_route blocks
by Jose Silva 17 Aug '06

17 Aug '06

>Message: 1 >Date: Tue, 01 Aug 2006 15:27:20 +0300 >From: OpenSer Users <openser-users(a)list.coretech.ro> >Subject: [Users] heders in route & failure_route blocks >To: users(a)openser.org >Message-ID: <44CF48A8.4020503(a)list.coretech.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed >hello, >I want to add Remote-Party-ID header in route block and change it in >failure_route block. >the problem is that after the failure_route block the message sent out >has two Remote-Party-ID headers. >in failure_route I am using remove_hf("Remote-Party-ID") >scenario: >original invite has no Remote-Party-ID header >in route block I use: remove_hf("Remote-Party-ID"); append_rpid_hf("<", >">;party=calling;privacy=full") >in failure_route block I use: remove_hf("Remote-Party-ID"); >append_rpid_hf("<", ">;party=calling;privacy=off") >the invite sent out after failure_route block has two Remote-Party-ID >headers, one with privacy=full and another with privacy=off. >is there a way to correct the problem ? >from what I understand the failure_route block works on a copy of the >original message, but the original message has no Remote-Party-ID >header, so why the header from the route block is still visible ? >on the other hand if the header added from the route block is visible >why is it no seen by the remove_hf function ? >thanks, >Razvan Radu Hi I all so have a similar problem! Is this behavior normal? Is it possible to have this type of manipulation of the message on the failure-route? Thanks in advance, Jose Silva

3 2

[Serusers] SIP MESSAGE method revisited.
by sip 17 Aug '06

17 Aug '06

Okay, Since I've been unable to pass SIP MESSAGEs back and forth through our server, but knowing full well that I once was able, I decided to take our configuration and remove everything in it that we've added over time (slowly regressing back to the beginning of our SVN code repository for the SER config) and see if I could tell what had changed so that I can no longer pass a SIP MESSAGE back and forth between clients. It didn't work. SO... I decided to take the hello-world config and try with it. No luck. What am I missing here? Is there some special handling that is needed to pass a SIP MESSAGE in SER 0.9.6? Is there a chance that something basic would have changed between 0.9.0 (our original server) and 0.9.6 that would have broken our ability to pass the MESSAGE method without additional tinkering? Sure, if I put in a block like: if(method=="MESSAGE") { sl_send_reply("200", "OK"); }; ... it's clear the message gets there and I receive the 200 OK, but of course it doesn't go anywhere after that. Without that block in there, it ends up getting a 513 Message too big response (which seems to be somewhat of a default for messages that don't end up getting handled in any other way that SER understands). I'm now completely at a loss. Pointers. Tips. Snide remarks. Anything helpful would be greatly appreciated. N.

5 8

[Serusers] Restrict user for one login
by ram 17 Aug '06

17 Aug '06

Hi I have installed SER and testing But in my both sipura config same Extension both are registering, how can i restrict the extensions. If the user already loging, reject other user to login using same extension. any sample rules. Ram

4 8

[Users] Problem with a NATing proxy
by Jeremie Le Hen 17 Aug '06

17 Aug '06

Hi, first of all, I am new to the SIP world, so please forgive me if I ask anything dumb. I am trying to set up a NAT-traversal SIP proxy, with the help of the nathelper module and rtpproxy. I only have access to the proxy and one softphone behind it. I am facing two problems: when someone in the outer world wants to call my softphone (which is behind the proxy), the proxy forwards the INVITE message to itself and recurses until the mf_process_maxfwd_header() statement stops further processing. I am not sure how to handle this. My guess is that I should use the registrar module and when the proxy receives an INVITE message, it should use registered() method in order to know where to forward the request ; but I don't know how to do that. Help would be welcome. My second problem is that when the peer hangs off the phone, I should receive a BYE message but I can't see it. Given I have no access on the other end, I guessed that the BYE message had been sent elsewhere. That's why I fixed the Contact header field. However, I still can't see actions coming from the outer phone. Any idea ? (I didn't attached any packet dump for now because I think it can be more misleading than informative because there are two proxies on the same machine: one that is a tunnel broker to encapsulate VoIP traffic in HTTP, and the other one is OpenSER. Basically, every SIP session is being tunneled in HTTP, then the datagram in unwrapped by the first proxy and then forwarded to the second one, which is OpenSER. Nonetheless I would be glad to provide any packet dump upon request.) My configuration file is attached. If you find any other mistakes or inconsistencies in this configuration file, please tell me. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Serusers] Ser with Asterisk for Voicemail
by Ricardo Carvalho 17 Aug '06

17 Aug '06

Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users database and sip routing, and Asterisk is used for voicemail and PSTN gateway. How should I edit ser.cfg to make Ser redirect calls to Asterisk when the user doesn't answer his phone or when the user exists but it hasn't any phone registered? If I'm using also Asterisk for Gateway purpose, I guess all calls to be sent to PSTN, Ser redirects them to Asterisk, but it will try to find a mailbox for every number dialled, when it should distinguish calls for PSTN not to be sent to Voicemail. How can Asterisk distinguish between a call sent for voicemail and a call sent by ser to be redirected to PSTN? It depends on the configuration made in ser.cfg? Thanks, Ricardo.

1 0

[Users] External Script execution?
by Dominik Bay 17 Aug '06

17 Aug '06

Hi, how can i execute an external script to lookup if a prefix exists in a mysqltable? I want to have prefix(""); filled from a database. -- Mit freundlichen Grüßen / Kind regards Dominik Bay Cablesurf Technik

1 1

[Users] ACK Routing: Best practices?
by Carsten Bock 17 Aug '06

17 Aug '06

Hi everybody, Short question: Are there any best practices regarding the routing of ACK Messages? Currently we have the following logic implemented: route { [...] ################################################################################################################## # Check for Re-Transmissions (not ACK/CANCEL) ############################################################################################################### if ((method != "CANCEL") && (method != "ACK")) { if (t_check_trans()) { log(1, "Re-Transmission detected, message dropped.\n"); # Drop the message silently. return; } } ################################################################################################################## # Loose-Routing (RFC3261) ############################################################################################################### if (loose_route()) { route(10); return; }; # if (loose_route()) { ################################################################################################################## # ACK/CANCEL Messages may be relayed ############################################################################################################### if ((method == "CANCEL") || (method == "ACK")) { if (uri != myself) { # Und das Paket entsprechend weiterleiten if (!t_relay("udp:outbound_proxy:outbound_proxy_port")) { log(1, "Not possible to relay\n"); # Fehler melden sl_reply_error(); return; } return; } } [...] ################################################################################################################## # Rest of "normal" Routing logic with number manipulation / Routing to different Gateways / User Authorization/Authentication ############################################################################################################### } Is there any other method, to improve the routing of ACK/CANCEL Messages? I would like to avoid all the number manipulation, checking and choosing of a proper gateway.... I thought, this must (somehow) be possible since from a technical perspective the ACK and CANCEL Message can be associated with a corresponding INVITE Message (and the corresponding Transaction). I've looked in the manual of the TM-Module and found the following example: if ( is_method("CANCEL") ) { if ( t_check_trans() ) t_relay(); exit; } (http://openser.org/docs/modules/1.1.x/tm.html#AEN460) I guess, this example does not mean, that the URI gets rewritten properly and the CANCEL-/ACK-Message would get properly relayed to the destination, where it is supposed to end. Is there any way to do it best? I've seen, that some clients do proper loose-routing for ACK-Messages, but not all clients, so this is not a solution. Are there any other suggestions? Thanks in advance, Carsten ____________ Virus checked by G DATA AntiVirusKit

4 5

[Serusers] iptelorg
by Sebastian Gabris 17 Aug '06

17 Aug '06

Hi everybody, does somebody know if the iptelorg GmbH is still existent.Visited their homepage www.iptelorg.com and it look pretty unmaintained to me. Is there any other company in Germany who offers support & installation services for SER? Best regards, Sebastian P.S. Congrats to iptel.org for their new homepage, will look closer at it tommorow :-) > -----Ursprüngliche Nachricht----- > Von: Sebastian Gabris <Sebastian.Gabris(a)web.de> > Gesendet: 11.08.06 16:21:36 > An: ratif(a)burraqtel.com.p, serusers(a)iptel.org > Betreff: [Serusers] ser SDP rewrite problem, rtpproxy is not working !!! > Hi, > > got also this problem, did you find a solution? > > Sebastian > > hello there, > > I am working on SER with two network interfaces. > > EP1--------------->(172.16.16.160) SER (192.168.5.1) -------------> EP2 > > what happens, SER receives invite from EP1, it reinitiates invite for > EP2. now what it does is that it writes the external IP i.e. > 172.16.16.160 in the SDP contact info, where as it should be the > internal one i.e. 192.168.5.1 ... > now I want to overwrite the SDP contact info in that invite. I have got > two options to do it one option is to use force_rtp_proxy("ei", > "192.168.5.1") or use sdp_mangle_ip() function of mangler.so . but they > are not working for me. when I play around with force_rtp_proxy function > rtpproxy get crashed and sdp_mangle_ip() is not behaving properly ...... > whats your experience ? > > I have used ser-0.8.14 and now using ser-0.9.0 with cvs rtpproxy. I am > running rtpproxy like > rtpproxy -s udp:192.168.5.1 -l 172.16.16.160 > and I am using this line in ser.cfg > modparam("nathelper", "rtpproxy_sock", "udp:192.168.5.1") > > I am really stucked please help !!!! > > thanking you > > regards, > -- > Atif > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

4 5

[Serusers] About the alias call
by AR. 17 Aug '06

17 Aug '06

Dear All I have a problem about the alias call When I use my PC and hardphone (cisco7960) , I can call someone by alias name But, It can't if I use the NB or use other PCs, Could anyone help me? thanks AR.

2 2

Re: [Serusers] IMPORANT ANNOUNCEMENT: Launch of new iptel.org site!!
by Sebastian Gabris 17 Aug '06

17 Aug '06

Hi all, I cannot register as new user. The reply email from iptel has a different username but my password(from the registration formular) in it. Tried it two times, and got two different usernames as reply. Also if I tried to login with my email-address instead of username but it didnt work. best regards, Sebastian > -----Ursprüngliche Nachricht----- > Von: "Greger V. Teigre" <greger(a)teigre.com> > Gesendet: 16.08.06 14:59:10 > An: sip <sip(a)arcdiv.com> > CC: serdev(a)iptel.org, serusers(a)iptel.org, serusers-bounces(a)lists.iptel.org > Betreff: Re: [Serusers] IMPORANT ANNOUNCEMENT: Launch of new iptel.org site!! sip wrote: > I actually just happened to hit the main site yesterday and noticed the > change. Much glitzier looking, although it's going to take me a little time to > get used to the new layout. > ;-) at least there is no flash... > One thing I didn't notice on there (perhaps because I just missed it) was a > link to the CVS repository to browse. Not a huge deal, it just used to be nice > to browse the READMEs for the modules (since the documentation hasn't all been > in once place). > Good point! Feel free to add it! > Maybe here?! > http://www.iptel.org/doc > > There is a need to do something about the module documentation. The > xml-based based documentation is a bit hard to maintain and I don't > think anybody has really gotton used to it. One suggestion is to move > the authoritative module documention to the new iptel.org site and keep > them maintained there. It will then be easier to maintain, more easily > accessible, and we can keep documentation for different versions easily > accessible. > > I'm looking forward to seeing the future of SER. > Me too! > g-) > > > N. > On Wed, 16 Aug 2006 09:50:09 +0200, Greger V. Teigre wrote > > > Dear all! > We are doing this the soft way: Launching when everybody is on vacation :-D > We have for some time now been working on putting up a new community-based framework for the iptel.org open source projects. Our aim is to step-wise address lack of documentation, roadmap, > release plans, transparent management and so on. > The new http://iptel.org/ site contains quite a lot of content. There is still plenty to migrate from the old iptel.org site, > ONsip.org, as well as better organize documentation around SER development and releases. You can help out with that by adding a small FAQ, how-to, copy something from ONsip.org that you found particularly helpful and so on! > We have created some basic content that we hope you will help comment and develop further. Here are some pointers to content on the new site: > ======================================== > About the new site and how to contribute content > http://www.iptel.org/iptelorgsite/intro > How to get started with SER? > http://www.iptel.org/ser/using > Overview of documentation > http://www.iptel.org/doc > The old ONsip.org Getting Started document > http://www.iptel.org/ser/doc/gettingstarted > Contributed modules and patches > http://www.iptel.org/ser/contributions > A beginning skeleton for a roadmap > http://www.iptel.org/ser/roadmap > A wishlist for future versions > http://www.iptel.org/ser/wishlist > Release policies > http://www.iptel.org/ser/development/release_policies > ====================================================== > Below is the launch announcement. > Best regards, > Greger Teigre > g-) > --------------------------------------------------------- > The new http://iptel.org/, home of SER, SEMS, and SERWeb is now up. If you have an account from ONsip.org, you can log in using your old ONsip.org username/password. However, all content is available without registration. > The new site is supported by the joint efforts of the iptel.org and ONsip.org teams. Our aim is to dramatically improve the project communication, documentation, and development process. The start is already up there, and we have many plans. The best thing of all: The new http://iptel.org site is a great basis for contributions from anyone. > We are working on migrating and organizing the remaining content from the old site, but meanwhile you can access everything from the old site from http://dog.iptel.org/ > We look forward to a revitalised SER open source project! > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ----------------------------------------------------------------- > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

2 1

[Serusers] Can ser listen on two or more then two ports?
by Charles Wang 16 Aug '06

16 Aug '06

Hi, ALL: Can ser listen on 2 or more than 2 ports on the same IP of one server? -- Best Regards Charles

2 1

[Users] SIP general question about via header
by Ricardo Martinez 16 Aug '06

16 Aug '06

Hello Is permitted to mix the two via forms (normal form and compact form) in one SIP message? For example i have an INVITE with this VIA's : INVITE sip:005623341876@10.0.0.67:5060 SIP/2.0 Record-Route: <sip:10.0.0.36;ftag=2179110362;lr=on> Record-Route: <sip:10.0.0.35;ftag=2179110362;lr=on> Via: SIP/2.0/UDP 10.0.0.36;branch=z9hG4bK58f4.cdc602d2.0 Via: SIP/2.0/UDP 10.0.0.35;branch=z9hG4bK58f4.83fe5745.0 v: SIP/2.0/UDP 64.76.148.234:5060;branch=z9hG4bK2881566366 .... If the answer is yes where can i found some documents to support this information? Thanks again! Best Regards, Ricardo Martinez.-

3 2

[Users] sip:+5555;npdi=yes@domain.com issue with enum_query()
by Greg Fausak 16 Aug '06

16 Aug '06

Howdy! How can I get rid of the tel: number crap on the end of a RURI like the one in the header? When I do my enum lookup it is failing, because it is looking up +5555;npdi=yes when I want it to look up +5555 Any advice from the wise would be greatly appreciated! Thanks, ---greg -- Greg Fausak greg(a)thursday.com

2 2

[Users] Need help to migrate OpenSer v1.0.x to v1.1.x
by Rafo RGV 16 Aug '06

16 Aug '06

Hi I am migrating OpenSER v1.0.9 to v1.1.0 using the config file I send below that works perfect in 1.0.9 and I modified to run in 1.1.x , I´ve just needed to change naming schema in parameter given to exported functions of avpops module to run without errors in this new version but when running can´t see basic logs and can´t register any client even if I install DB schema and create subscriber using its openserctl , ngrep just show me a 'forbidden' message but dont know why, please someone send me some advice. thanks rafael # ----------------- global configuration parameters -------------------- debug=4 fork=yes log_stderror=yes listen=192.168.2.136 # OPENSER IP ADDRESS HERE port=5060 # Hostname matching an alias will satisfy the condition uri==myself". # alias=mydomain.com.pe:5060 # alias=mydomain.com.pe alias=192.168.2.136:5060 alias=192.168.2.136 check_via=yes # (cmd. line: -v) children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" # fifo="FIFO" fifo_mode=0666 # Fifo permissions can be changes from here. fifo_db_url="mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser" # ----------------------------------------------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" ## loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/uac_redirect.so" loadmodule "/usr/local/lib/openser/modules/diversion.so" # ----------------------------------------------------------------------- modparam("auth_db|permissions|group|uri_db|usrloc", "db_url", "mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser") modparam("avpops", "avp_table", "usr_preferences") # ------------- Mysql Accounting parameters modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://dbuser:dbpass@dbt2.mydomain.com.pe/openser") modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc (para INVITE y ACK) modparam("acc", "log_fmt", "miocfsputdr") modparam("acc", "detect_direction", 1) # only in 1.1.x or cvs modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) # ------------- Domain parameters modparam("usrloc", "use_domain", 1) modparam("auth_db", "use_domain", 1) modparam("registrar", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # ------------------------------------------------------------------------- route { # log(1, "-------------------------------------------------------\n"); # log(1, "entering main loop\n"); ## populate the avps (for Multi call leg accounting): if (!method=="REGISTER" && !method=="NOTIFY") { avp_write("$from/username", "$avp(i:110)"); avp_write("$ruri/username", "$avp(i:111)"); }; # set flag for Missed calls for Accounting: if (!method=="OPTIONS") setflag(3); if (method=="INVITE") { log(1, " INVITE MESSAGE RECEIVED - START ACC\n"); }; if (method=="BYE") { log (1, " BYE MESSAGE RECEIVED - STOP ACCOUNTING\n"); }; if (method=="CANCEL") { log (1, " CANCEL MESSAGE RECEIVED - STOP ACCOUNTING\n"); }; if (method=="OPTIONS") { log(1, " OPTIONS MESSAGE RECEIVED\n"); return; #Do Nothing for 'Options' }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Handle NOTIFY requests from Sipura's to keep-alive NAT # ----------------------------------------------------------------- if ((is_method("NOTIFY")) && (! uri=~"sip:.*[@]+.*")) { sl_send_reply("200", "OK keep-alive"); log(1, " OK keep-alive\n"); return; }; # ----------------------------------------------------------------- # Only allow known sip messages # ----------------------------------------------------------------- if (!is_method("INVITE|REGISTER|BYE|NOTIFY|ACK|CANCEL|REFER|SUBSCRIBE")) { sl_send_reply("501", "Not implemented here"); log(1, " 501 - Not implemented here\n"); return; }; # ----------------------------------------------------------------- # SUBSCRIBE cleanup # ----------------------------------------------------------------- if (is_method("SUBSCRIBE")) { replace(" talk", "talk"); }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && nat_uac_test("19")) { # OPENSER IP ADDRESS OR DOMAIN HERE: record_route_preset("192.168.2.136:5060;nat=yes"); setflag(1); ## Set Flag 1 for Acc: test from this location!! } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="REFER" || method=="CANCEL") { unforce_rtp_proxy(); setflag(1); ### Set Flag 1 for Acounting }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(7); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { setflag(1); # Set Flag for Acounting route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location") && !method=="ACK") { if (does_uri_exist()) { log(1, " uri_exist but its Temporarily Unavailable 1 \n"); route(7); # To Voicemail return; } else { log(1, " unable to locate user - User Not Exist 1 \n"); sl_send_reply("404", "User Not Exist"); acc_db_request("404 User Not Exist", "missed_calls"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (is_user_in("from", "desactivado")) { sl_send_reply("402", "desactivada temporalmente"); log(1," Cta. desactivada temporalmente \n"); return; }; if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { log(1," Fails to Register \n"); www_challenge("","0"); return; }; if (!check_to()) { log(1," Unauthorized registration attempt \n"); sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); log(1," Registered! \n"); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- log(1," route[3]---> INVITE Message Handler \n"); if ( allow_trusted() ) log(1, " Call from pstn or *pbx, no authentication is required. \n"); if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); log(1, " Sorry, only registered users are allowed \n"); return; }; consume_credentials(); }; append_rpid_hf(); # some digits rules here.... lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return; }; if (nat_uac_test("19")) { setflag(7); force_rport(); }; # ----------------------------------------------------------------- # to PSTN Routes!!! # ----------------------------------------------------------------- route(6); # To PSTN & LDI Routes block!!! # ----------------------------------------------------------------- if ( is_uri_host_local() ) { # (Is he in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc: if (is_user_in("Request-URI", "voicemail")) { log(1, " requested user is in voicemail group \n"); setflag(4); }; # if user is on-line and is in Voicemail group, enable redirection # Pendiente: # ver como meter voicemail al avp junto a las otras funciones... if (method == "INVITE" && isflagset(4)) { log(1, " invite for voicemail user->initiate failureroute[1]\n"); t_on_failure("1"); }; # flag 22 determines if we need to call the append_branch() for Blind CallFwd. if (avp_db_load("$ruri/username", "$avp(s:callfwd)")) { setflag(22); avp_pushto("$ruri", "$avp(s:callfwd)"); route(6); return; }; # native SIP destinations are handled using our USRLOC DB: if (!lookup("location") ) { if (does_uri_exist()) { log(1," uri_exist but its Temporarily Unavailable 2 \n"); route(7); # To Voicemail return; } else { log(1," Unable to locate xcompany user - User Not Exist 2X \n"); prefix("iu"); ## in asterisk: iu: exten => _iu1780XXXX,1,Playback(invalid,noanswer) acc_db_request("404 User Invalid", "missed_calls"); rewritehostport("192.168.2.131:5070"); t_relay(); return; }; }; # t_on_failure() informs SER that we want to perform special handling when a failure condition occurs. # Failure conditions in this context refer to 4xx and 5xx response codes. # By setting t_on_failure(1) before calling t_relay(), oSER will pass control to the failure_route[1] t_on_failure("1"); route(4); route(1); }; } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- ### First check and resetflags if UAS are behind the same NAT system: ### if (isflagset(6) && isflagset(7)){ log(2, " Both Clients are behind NAT"); # Store the destination domain into an AVP avp_printf("$avp(i:450)", "$dd"); if (avp_check("$avp(i:450)", "eq/$src_ip/g")){ log(3, " Detected Two Clients Behind the Same NAT - Disabling Mediaproxy"); # Do not use media-proxy as the clients seem to be behind the same NAT resetflag(6); resetflag(7); }; }; if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); force_rport(); fix_nated_contact(); force_rtp_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # ILD - PSTN Handler # ----------------------------------------------------------------- # Forward international calls to Asterisk (a SIP LDI Providers): if(uri=~"^sip:011"){ log(1, " 011N match - Larga Distancia Internacional a USA \n"); if (!is_user_in("from", "ldiusa")) { log(1," No permission for international calls to USA \n"); sl_send_reply("403", "No permission for international calls to USA"); acc_db_request("403 Forbidden", "missed_calls"); return; }; strip(2); }; if(uri=~"^sip:011"){ log(1," 011N match - Larga Distancia Internacional \n"); if (!is_user_in("from", "ldix")) { log(1, " No permission for international calls \n"); sl_send_reply("403", "No permission for international calls"); acc_db_request("403 Forbidden", "missed_calls"); return; }; }; rewritehostport("192.168.2.137:5070"); # Asterisk or PSTN GATEWAY IP ADDRESS t_on_failure("1"); route(4); route(1); } route[6] { # ----------------- Free Calls ------------------- # To Info numbers 102, 104, 123 y 080011236 if(uri=~"^sip:102@" || uri=~"^sip:104@" || uri=~"^sip:123@" || uri=~"^sip:0800123456@"){ log(1," To CallCenter xcompany!! \n"); rewriteuri("sip:123456@mydomain.com.pe"); rewritehostport("callcenter.mydomain.com.pe:5060"); route(4); route(1); return; }; if (is_user_in("from", "restringido")) { sl_send_reply("402", "Restriccion temporal"); log(1," 402 - Restriccion temporal \n"); acc_db_request("402 Restricted", "missed_calls"); return; }; lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); return; }; # -------------------------- Seccion de Rutas a LDI ---------------------------------------- # Forward international calls to Asterisk (To SIP LDI Providers): if (uri=~"^sip:011[0-9]*@" && !uri=~"^sip:01152*@") { route(5); return; }; if( uri=~"^sip:[2-8][0-9]{6}@" || uri=~"^sip:1[2-8][0-9]{6}@" || uri=~"^sip:01[2-8][0-9]{6}@" ) { log(1, " A Fijos Mexico PSTN \n"); if (uri=~"^sip:01[2-8][0-9][0-9][0-9][0-9][0-9][0-9]@") { strip(1); }; if (uri=~"^sip:[2-8][0-9][0-9][0-9][0-9][0-9][0-9]@") { prefix("1"); }; if (!is_user_in("from", "fijomexico")) { sl_send_reply("403", "No permission for local fixed calls"); log(1, " 403 - No permission for local fixed calls \n"); acc_db_request("403 Forbidden", "missed_calls"); return; }; rewritehostport("192.168.2.137:5070"); prefix("52"); route(4); route(1); return; }; # no existe? (debe ir al final siempre) if ( uri=~"^sip:[0-9]" && !does_uri_exist() ) { log(1, " Unable to locate route for this dnid - User Not Exist 3X \n"); prefix("mc"); acc_db_request("404 Number Invalid or Incomplete", "missed_calls"); rewritehostport("192.168.2.131:5070"); t_relay(); return; }; } route[7] { # ---------------------- Handling of Unavailable user ---------------------- # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL" || method=="BYE" || method=="OPTIONS")) { sl_send_reply("404", "Not Found"); acc_db_request("404 Not Found", "missed_calls"); log(1, " 404 Not Found & non-Voip -- just send off-line \n"); return; }; # Not voicemail subscriber: if (!isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE" && !method=="CANCEL") { #sl_send_reply("404", "Not Found and no VMail turned on"); acc_db_request("480 Temporarily Unavailable", "missed_calls"); # A locucion (Usuario temporalmente fuera de servicio 'ss-noservice'); log(1, " 404 Not Found and no voicemail \n"); prefix("fu"); ## su: exten => _su1780XXXX,1,Playback(ss-noservice,noanswer) rewritehostport("192.168.2.131:5070"); t_relay(); return; }; # forward to voicemail adding prefix to simplify asterisk "extension.conf" :: if (isflagset(4) && !method=="OPTIONS" && !method=="ACK" && !method=="BYE" && !method=="CANCEL") { prefix("vm"); acc_db_request("480 Temporarily Unavailable -> Vm", "missed_calls"); rewritehostport ("192.168.2.131:5070"); t_relay(); return; }; } # -------------------------------------------------------------------------- onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { if (t_check_status("487")) { return; }; if (t_check_status("486")){ prefix("vmb"); acc_db_request("486 Busy -> Vm", "missed_calls"); rewritehostport ("192.168.2.131:5070"); append_branch(); t_relay(); return; }; if (t_check_status("408")){ prefix("vmu"); acc_db_request("408 Timeout -> Vm", "missed_calls"); rewritehostport ("192.168.2.131:5070"); append_branch(); t_relay(); return; }; unforce_rtp_proxy(); } failure_route[2] { # handle 302 redirects if (t_check_status("3[0-9][0-9]")) { get_redirects("*"); t_relay(); }; } ### The End ###

2 2

[Serusers] Via RPORT for CANCEL message.
by Ricardo Martinez 16 Aug '06

16 Aug '06

Hello. My main configuration is based on the ser.cfg from the issue 5 plus some modifications included by me. Beside that i'm using version: ser 0.9.3 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 11:03:37 Dec 22 2005 with gcc 3.2 I have a question regarding to sending a CANCEL message for an INVITE from a NAT'd endpoint. As far as i know the force_rport() command adds the received IP port to the top most via header in the SIP message, this enables subsequent SIP message to return to the proper port later on in a SIP transaction. My problem is that for a CANCEL message coming from a NAT'd endpoint this command seems not to be working. This is the scenario. NAT'd endpoint : 200.100.100.248 SER : 200.100.100.246 SER-2 : 200.100.100.36 GW : 200.100.100.69 The NAT'd endpoint send an INVITE to the proxy, but then in the middle of the transaction decide to CANCEL the request. As you can see in the "cancel_debug.txt" file included on this mail, the CANCEL message does not contain the "rport" in the Via header, so it seems to be routed back to the default sip port (5060). Is there a way to force the rport in a CANCEL? for a enpoint unable to put the rport by itself? I want to do this according to the onsip document, in a compatible way.. Anyway i made a little test including a force_rport(); in the CANCEL handler .......... # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("200.100.100.246:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); setflag(1); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { force_rport(); route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; } else if (method=="ACK") { route(1); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); this solve my problem but it seems not to be so accurate. I hope that someone can help me Thanks! Ricardo Martinez.- <<cancel_debug.txt>>

1 0

SV: [Users] Route-header DNS lookup?
by Kerker Staffan 16 Aug '06

16 Aug '06

hi is this issue (maddr=) fixed in the 1.1.0 release? do you know if the db_mode=3 is in the 1.1.0 release? best regards, /staffan > -----Ursprungligt meddelande----- > Från: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Skickat: den 31 juli 2006 09:36 > Till: Simon Morvan > Kopia: Kerker Staffan; users(a)openser.org > Ämne: Re: [Users] Route-header DNS lookup? > > Hi Simon! > > Please make a unified diff (cvs diff -u) and post it at the > bugtracker on sourceforge: > http://sourceforge.net/tracker/?group_id=139143&atid=743022 > > This way it can't get lost under thousands of emails. > > regards > klaus > > Simon Morvan wrote: > > modified file : modules/rr/loose.c > > > > 2006/7/28, Simon Morvan <garphy(a)gmail.com>: > >> after a quick check, the answer is : no > >> > >> this is a quick hack i've made as a workaround. feedback > welcome :) > >> > >> 781,783c781,803 > >> < if (set_dst_uri(_m, uri) < 0) { > >> < LOG(L_ERR, "after_loose: Error > while setting > >> dst_uri\n"); > >> < return RR_ERROR; > >> --- > >> > if( puri.maddr.s != NULL ){ > >> > str builturistr; > >> > char builturi[150]; > >> > > >> > memcpy( builturi, "sip:", 4 ); > >> > memcpy( builturi+4, puri.maddr.s+6, > >> puri.maddr.len-6 ); > >> > builturi[4+puri.maddr.len-6] ='\0'; > >> > > >> > LOG( L_ERR, "MaddrURI is %p\n", > builturi ); > >> > > >> > builturistr.s = builturi; > >> > builturistr.len = 4+puri.maddr.len-6; > >> > > >> > if (set_dst_uri(_m, &builturistr) < 0) { > >> > LOG(L_ERR, "after_loose: > Error while > >> setting dst_uri\n"); > >> > return RR_ERROR; > >> > } > >> > > >> > }else{ > >> > if (set_dst_uri(_m, uri) < 0) { > >> > LOG(L_ERR, "after_loose: > Error while > >> setting dst_uri\n"); > >> > return RR_ERROR; > >> > } > >> > >> > >> 2006/7/28, Simon Morvan <garphy(a)gmail.com>: > >> > Is this issue solved in Openser 1.1.0 ? > >> > > >> > 2006/7/7, Simon Morvan <garphy(a)gmail.com>: > >> > > As Daniel said on the ml thread : > >> > > > it makes sense, we will analyze the implications > and include it > >> on the > >> > > > to-do list. > >> > > > > >> > > > Cheers, > >> > > > Daniel > >> > > Has it been included in the developpement branch yet ? > >> > > > >> > > -- > >> > > Simon. > >> > > > >> > > 2006/7/7, Klaus Darilion <klaus.mailinglists(a)pernau.at>: > >> > > > I'm not sure about the maddr parameter. doesn't it > mean that > >> > > > the > >> request > >> > > > should be forwarded to this IP regardsless of the domain? If > >> yes, the > >> > > > DNS lookup can be skipped. > >> > > > > >> > > > regards > >> > > > klaus > >> > > > > >> > > > Kerker Staffan wrote: > >> > > > > hi > >> > > > > i recently bounced into this problem, and i'm not > sure here. > >> > > > > i'm running the openser-devel, with the cacheless > db_mode=3. > >> (works fine btw) > >> > > > > > >> > > > > the record-route header received by the proxy on the other > >> side (SNOM4S), inserts > >> > > > > the domain name (iptel1.ipatl.se) and not the hostname > >> (sip.iptel1.ipatl.se) in the > >> > > > > record-route header, and uses the > maddr=<ip_of_server> with > >> the actual server IP address. > >> > > > > > >> > > > > now, when my client (behind the OpenSER) replies > with an ACK > >> to the incomming OK, > >> > > > > it uses the Route-header recieved in the > RR-header, and sends > >> the ACK to OpenSER. i > >> > > > > then get the following errors in OpenSER. > >> > > > > > >> > > > > --- > >> > > > > /usr/local/sbin/openser[3583]: ERROR: mk_proxy: could not > >> resolve hostname: "iptel1.ipatl.se" > >> > > > > /usr/local/sbin/openser[3583]: ERROR: uri2proxy: bad host > >> > > > > name > >> in URI > >> > <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;maddr=172. > >> 28.248.66;transport=udp;lr> > >> > >> > > > > --- > >> > > > > > >> > > > > the ACK i sent look like this: > >> > > > > > >> > > > > --- > >> > > > > Request-Line: ACK > sip:2307@iptel1.ipatl.se;gruu=6fg9n6dl SIP/2.0 > >> > > > > Via: SIP/2.0/UDP > >> 172.28.248.52:2051;branch=z9hG4bK-d96b1fvapkyn;rport > >> > > > > Route: <sip:172.28.248.10;lr=on;ftag=li9buf1i4p> > >> > > > > Route: > >> > <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;ma > ddr= > >> 172.28.248.66;transport=udp;lr> > >> > > > > From: "Snom 2652" <sip:2652@ipatl.se>;tag=li9buf1i4p > >> > > > > To: <sip:2307@ipatl.se>;tag=hvseiz7kgb > >> > > > > Call-ID: 3c269d83900b-xj3ild14y880@snom360 > >> > > > > CSeq: 1 ACK > >> > > > > Max-Forwards: 70 > >> > > > > Contact: <sip:2652@172.28.248.52:2051;line=cp4a7ljd> > >> > > > > Content-Length: 0 > >> > > > > --- > >> > > > > > >> > > > > as far as i understand, according the rfc 3263, the > >> route-header may contain domain name that > >> > > > > has to be resolved using SRV. > >> > > > > > >> > > > > --- > >> > > > > "6 Constructing SIP URIs > >> > > > > > >> > > > > In many cases, an element needs to construct a > SIP URI for > >> inclusion > >> > > > > in a Contact header in a REGISTER, or in a Record-Route > >> header in an > >> > > > > INVITE. According to RFC 3261 [1], these URIs have to > >> > > > > have > >> the > >> > > > > property that they resolve to the specific element that > >> inserted > >> > > > > them. However, if they are constructed with just an IP > >> address, for > >> > > > > example: > >> > > > > > >> > > > > sip:1.2.3.4 > >> > > > > > >> > > > > then should the element fail, there is no way > to route the > >> request or > >> > > > > response through a backup. > >> > > > > > >> > > > > SRV provides a way to fix this. Instead of using an IP > >> address, a > >> > > > > domain name that resolves to an SRV record can be used: > >> > > > > > >> > > > > sip:server23.provider.com" > >> > > > > --- > >> > > > > > >> > > > > now, OpenSER only asks DNS for an A record of the name > >> recieved in the route header, > >> > > > > and since that's a domain name, it's > unresolvable, and so the > >> ACK is never sent. > >> > > > > > >> > > > > any hints or clues? > >> > > > > > >> > > > > best regards, > >> > > > > /Staffan Kerker > >> > > > > > >> > > > > > >> > > > > -- > >> > > > > Staffan Kerker > >> > > > > Saab Communications, Växjö > >> > > > > p. +46 470 42185 > >> > > > > c. +46 705 391365 > >> > > > > m. staffan.kerker(a)saabgroup.com > >> > > > > > >> > > > > > >> > > > > _______________________________________________ > >> > > > > Users mailing list > >> > > > > Users(a)openser.org > >> > > > > http://openser.org/cgi-bin/mailman/listinfo/users > >> > > > > >> > > > > >> > > > _______________________________________________ > >> > > > Users mailing list > >> > > > Users(a)openser.org > >> > > > http://openser.org/cgi-bin/mailman/listinfo/users > >> > > > > >> > > > >> > > >> > >

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[Serusers] IMPORANT ANNOUNCEMENT: Launch of new iptel.org site!!
by Greger V. Teigre 16 Aug '06

16 Aug '06

Dear all! We are doing this the soft way: Launching when everybody is on vacation :-D We have for some time now been working on putting up a new community-based framework for the iptel.org open source projects. Our aim is to step-wise address lack of documentation, roadmap, release plans, transparent management and so on. The new http://iptel.org/ site contains quite a lot of content. There is still plenty to migrate from the old iptel.org site, ONsip.org, as well as better organize documentation around SER development and releases. You can help out with that by adding a small FAQ, how-to, copy something from ONsip.org that you found particularly helpful and so on! We have created some basic content that we hope you will help comment and develop further. Here are some pointers to content on the new site: ======================================== About the new site and how to contribute content http://www.iptel.org/iptelorgsite/intro How to get started with SER? http://www.iptel.org/ser/using Overview of documentation http://www.iptel.org/doc The old ONsip.org Getting Started document http://www.iptel.org/ser/doc/gettingstarted Contributed modules and patches http://www.iptel.org/ser/contributions A beginning skeleton for a roadmap http://www.iptel.org/ser/roadmap A wishlist for future versions http://www.iptel.org/ser/wishlist Release policies http://www.iptel.org/ser/development/release_policies ====================================================== Below is the launch announcement. Best regards, Greger Teigre g-) --------------------------------------------------------- The new http://iptel.org/, home of SER, SEMS, and SERWeb is now up. If you have an account from ONsip.org, you can log in using your old ONsip.org username/password. However, all content is available without registration. The new site is supported by the joint efforts of the iptel.org and ONsip.org teams. Our aim is to dramatically improve the project communication, documentation, and development process. The start is already up there, and we have many plans. The best thing of all: The new http://iptel.org site is a great basis for contributions from anyone. We are working on migrating and organizing the remaining content from the old site, but meanwhile you can access everything from the old site from http://dog.iptel.org/ We look forward to a revitalised SER open source project!

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RE: [Users] Error REGISTERing and adding users to OpenSER 1.1.0 on Solaris
by peter.3.edwards＠bt.com 16 Aug '06

16 Aug '06

> yes, it was fixed in the past, but I reintroduced by mistake with the > new version of openserctl. It was reported on bug tracker as > well and it > is now fixed on CVS (both head and rel_1_1_0 branches). Yep, I unconsciously posted to the bug tracker with my non-work pseudonym ... :) I've followed the posts on there, as well. > You can use the /usr/local/etc/openser/openserctlrc (or > ~/.openserctlrc) > to set the AWK variable to the appropriate path to gmd5sum. > Most of the > tool used inside openserctl can be set via env variables. > > Cheers, > Daniel Yes, but it makes more sense on my system to have an md5sum available, so I'm happy to symlink. Cheers for the assistance, Peter. -- This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

1 0

[Serusers] OPTIONS PING functionality in nathelper patch for 0.9.x - was - Re: [Serdev] Memory problem in t_request
by Greger V. Teigre 16 Aug '06

16 Aug '06

Hi all, I have now had the nathelper 0.9.x patch for OPTIONS keep-alive running successfully for a while. So, if anyone is interested, here it is: http://www.iptel.org/nathelper_sip_message_options_ping_functionality g-)

1 0

[Users] Developed modules
by Andrew And 16 Aug '06

16 Aug '06

I need to write modules (developed module) and I didn´t find documentation in www.openser.org about this, where could I find it?

2 1

[Serusers] new user for SER
by vijay tiwari 15 Aug '06

15 Aug '06

hi all hi guys please help me out how to start SER and what will be best way to start SER thanks vijay --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] Module for Config files
by ronn100200＠aim.com 15 Aug '06

15 Aug '06

Hi All, I am using the following if statement to allow certain calls from pre authorized IPs. if ( method == "INVITE" && src_ip != 195.xxx.xxx.xx && src_ip != 195.xxx.xxx.xx && src_ip != 195.xxx.xxx.xx && src_ip != 195.xxx.xxx.xx) { My issues is the number of IPs that need to be authorized this way is getting large and was wondering if there's a module that can be used to read config date from a config file ? regards, Ron ________________________________________________________________________ Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection.

5 8

RE: [Users] Error REGISTERing and adding users to OpenSER 1.1.0 on Solaris
by peter.3.edwards＠bt.com 15 Aug '06

15 Aug '06

> Hello, > > you need a tool called md5sum. On Linux/Debian it comes with > coreutils > package. It computes the md5 of a string and it used mainly > to check the > files if they were corrupted after a download. Maybe it exists under > other name in solaris, although nobody reported so far. > > For solaris 8 I found the following: > http://www.sun.com/software/solaris/freeware/pkgs_download.xml#md5sum > > Hope it helps. > > Cheers, > Daniel Thanks, Daniel. I was skipping over the md5sum errors (I already had it installed, but under Solaris it becomes gmd5sum - a quick symlink sorted it out) because I thought the awk errors were quite serious. Should've looked a bit closer before posting because it turns out, again, to be another Linux-biased script ... Lines 135 and 136 in /usr/local/lib64/openser/openserctl/openserctl.base need the space removing before the @ sign. Solaris awk does not allow whitespace before the parameter. After sorting that out and symlinking gmd5sum, the add user script worked and I can now register a SIP User Agent - which is a great leap fwd ... Many thanks, Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

2 1

[Serusers] ser-0.9.7pre3
by Steve Blair 15 Aug '06

15 Aug '06

I'm trying to append an rpid header to enable caller ID blocking at my gateway. I started by adding the following commands but a quick check with ser -c -f <filename> produces the following errors. Everything in the documentation suggests use_rpid does indeed exist in the auth_db module. Does anyone know what might be causing this error? Thanks,Steve modparam("auth_db", "rpid_column", "remote_party_id") modparam("auth_db", "use_rpid", 1) 0(20424) set_mod_param_regex: parameter <rpid_column> not found in module <auth_db> 0(20424) parse error (162,20-21): Can't set module parameter 0(20424) set_mod_param_regex: parameter <use_rpid> not found in module <auth_db> 0(20424) parse error (163,19-20): Can't set module parameter -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

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[Serusers] RTPPROXY & lookup request failed: session
by Sebastian Gabris 15 Aug '06

15 Aug '06

Hi again, I start rtpproxy like this: rtpproxy -2f -l 193.55.55.55/10.245.x.x can someone give me a quick hint what problem could cause this output: outgoing calls work & have this output: received command "U 5e04ce62ad145445YTczODQwZGYxMDVhODNjODYwZWE0YzQxOTA3OGEwMmM.10.245.16.55 45746 6879e139" new session 5e04ce62ad145445YTczODQwZGYxMDVhODNjODYwZWE0YzQxOTA3OGEwMmM., tag 68 9 requested, type strong new session on a port 35000 created, tag 6879e139 pre-filling caller's address with 10.245.16.55:45746 sending reply "35000 193.77.54.68 " incoming calls dont work & have this output: received command "L 30e096903015a6b321b8fcbe29eafda8(a)217.10.67.4 10.245.16.55 44 224 as15976aa3 71195064" lookup request failed: session 30e096903015a6b321b8fcbe29eafda8(a)217.10.67.4, tags as15976aa3/71195064 not found sending reply "0 193.77.54.68 " Thanks Sebastian > -----Ursprüngliche Nachricht----- > Von: Sebastian Gabris <Sebastian.Gabris(a)web.de> > Gesendet: 11.08.06 13:47:11 > An: serusers(a)lists.iptel.org > Betreff: [Serusers] Problem with incoming calls > Hi everybody, > > Like told in the subject i got problems with incomnig calls, SIP signalling works but RTP doesnt. > When I make a call from my UA to the VoIP-Provider all works well, sip & rtp. > I start rtpproxy in bridged mode: rtpproxy -l publicIP/privateIP. > > I use this scenario: > > UA(private IP) <-> (privateIP) SER (publicIP) <-> VoIP-Provider > > i try to figure out where my mistake is in this config file: > > route[1] > { > if(dst_ip == 10.245.104.68) > { > force_rtp_proxy("","193.77.54.68"); > } > else > { > force_rtp_proxy("","10.245.104.68"); > }; > > t_on_reply("1"); > > if (!t_relay()) { > sl_reply_error(); > }; > } > > onreply_route[1] { > if (status =~ "(183)|2[0-9][0-9]") { > if(dst_ip == 10.245.104.68) > { > force_rtp_proxy("","193.77.54.68"); > } > else > { > force_rtp_proxy("","10.245.104.68"); > }; > > }; > } > > > By the way, is there a option to view rtpproxy debug info? > > > Thanks in advance! > > Sebastian > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

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RE: [Serusers] Adding area codes to dialed number
by Rosa De Santis 14 Aug '06

14 Aug '06

Thanks for your quick reply, Weiter And how is that local number format? Thanks again. ----------------------------------------------- > Date: Mon, 14 Aug 2006 23:07:47 +0200 > From: bp4mls(a)googlemail.com > To: rosadesantis(a)hotmail.com > Subject: Re: [Serusers] Adding area codes to dialed number > CC: serusers(a)lists.iptel.org > > Yes, it is possible. > If geographical user distribution is mapped to domains, using domain attributes. > Otherwise, you would have to store this information (country code, area code) per user, as user attributes. When the user originates a call, check if using local number format and load his attributes and fix the uri otherwise. > On 8/14/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > Hello all. > Please, I'd like to know if there is a way in SER to add the appropriates area codes (country and city) to a PSTN dialed number, based in the location of the calling user, using avpops and/or databases values, so that the user can dial local numbers without adding country and city codes. > For example, the user has location prefixes for his country=11 and city=222, when he dials the PSTN number 123456 that is local based in his location, then the SER action adds 11 + 222 to this dialed number, to dial the complete number 11222123456 ? > Is this possible? How can be done? > Please, any hint is appreciated... > Thanks > _________________________________________________________________ > Descubre Live.com - tu mundo en línea reunido: noticias, deportes, el tiempo, y mucho más. > http://www.live.com/getstarted_____________________________________________… > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Descubre Live.com - tu propia página de inicio, personalizada para ver rápidamente todo lo que te interesa en un mismo sitio. http://www.live.com/getstarted

1 0

[Serusers] Adding area codes to dialed number
by Rosa De Santis 14 Aug '06

14 Aug '06

Hello all. Please, I'd like to know if there is a way in SER to add the appropriates area codes (country and city) to a PSTN dialed number, based in the location of the calling user, using avpops and/or databases values, so that the user can dial local numbers without adding country and city codes. For example, the user has location prefixes for his country=11 and city=222, when he dials the PSTN number 123456 that is local based in his location, then the SER action adds 11 + 222 to this dialed number, to dial the complete number 11222123456 ? Is this possible? How can be done? Please, any hint is appreciated... Thanks _________________________________________________________________ Descubre Live.com - tu mundo en línea reunido: noticias, deportes, el tiempo, y mucho más. http://www.live.com/getstarted

2 1

[Users] nathelper & fax = bug ?
by Dmitry Lyubimkov 14 Aug '06

14 Aug '06

Connection scheme: UA - router with NAT - OpenSER with nathelper - PSTN gateway (Cisco AS5350) (192.168.13.109) (217.107.59.194) (62.33.22.14) (62.33.22.11) Both incoming and outgoing calls work right. Openser uses the nathelper module for proxing of rtp stream of NAT UA. Here is example of SIP messages (call from PSTN through a gateway): 15:37:07.406529 IP 62.33.22.11.54581 > 62.33.22.14.5060: UDP, length 1121 E..}........>!..>!...5...i.hINVITE sip:78142799233@voapp.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone" From: <sip:78142764164@62.33.22.11>;tag=A515D068-227D To: <sip:78142799233@voapp.ru> Date: Fri, 04 Aug 2006 11:37:07 GMT Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)195.161.136.114 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1688609156-585372123-2332753710-1711444852 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:78142764164@62.33.22.11>;party=calling;screen=yes;privacy=off Timestamp: 1154691427 Contact: <sip:78142764164@62.33.22.11:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 316 v=0 o=CiscoSystemsSIP-GW-UserAgent 4330 9654 IN IP4 62.33.22.11 s=SIP Call c=IN IP4 62.33.22.11 t=0 0 m=audio 17088 RTP/AVP 3 18 8 0 4 c=IN IP4 62.33.22.11 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes Nathelper works right and in the message sent to UA you can see already IP address of Openser (62.33.22.14) instead of the address of a gateway (62.33.22.11): 15:37:07.407463 IP 62.33.22.14.5060 > 217.107.59.194.47331: UDP, length 1256 E.....@.@..|>!...k;.......n^INVITE sip:ngul@217.107.59.194:47331 SIP/2.0 Record-Route: <sip:62.33.22.14;lr;ftag=A515D068-227D> Via: SIP/2.0/UDP voapp.ru:5060;branch=z9hG4bK2d06.d63c8585.0 Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone" From: <sip:78142764164@62.33.22.11>;tag=A515D068-227D To: <sip:78142799233@voapp.ru> Date: Fri, 04 Aug 2006 11:37:07 GMT Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)195.161.136.114 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1688609156-585372123-2332753710-1711444852 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: <sip:78142764164@62.33.22.11>;party=calling;screen=yes;privacy=off Timestamp: 1154691427 Contact: <sip:78142764164@62.33.22.11:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 334 v=0 o=CiscoSystemsSIP-GW-UserAgent 4330 9654 IN IP4 62.33.22.11 s=SIP Call c=IN IP4 62.33.22.14 t=0 0 m=audio 35858 RTP/AVP 3 18 8 0 4 c=IN IP4 62.33.22.14 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=nortpproxy:yes After some talking the subscriber from PSTN tries to send a fax. PSTN gateway detects it and sends this message: 15:37:22.512722 IP 62.33.22.11.51655 > 62.33.22.14.5060: UDP, length 1276 E..........z>!..>!..........INVITE sip:62.33.22.14:5060;from-tag=A515D068-227D;lr SIP/2.0 Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone" From: <sip:78142764164@62.33.22.11>;tag=A515D068-227D To: <sip:78142799233@voapp.ru>;tag=bbaac0e818284ff5 Date: Fri, 04 Aug 2006 11:37:22 GMT Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)195.161.136.114 Route: <sip:ngul@217.107.59.194:47331> Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1688609156-585372123-2332753710-1711444852 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 102 INVITE Max-Forwards: 6 Remote-Party-ID: <sip:78142764164@62.33.22.11>;party=calling;screen=yes;privacy=off Timestamp: 1154691442 Contact: <sip:78142764164@62.33.22.11:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 393 v=0 o=CiscoSystemsSIP-GW-UserAgent 4330 9656 IN IP4 62.33.22.11 s=SIP Call c=IN IP4 62.33.22.11 t=0 0 m=image 17088 udptl t38 c=IN IP4 62.33.22.11 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy Openser processes is and sends to UA: 15:37:22.513017 IP 62.33.22.14.5060 > 217.107.59.194.47331: UDP, length 1336 E..T..@.@..,>!...k;......(a)n.INVITE sip:ngul@217.107.59.194:47331 SIP/2.0 Record-Route: <sip:62.33.22.14;lr;ftag=A515D068-227D> Via: SIP/2.0/UDP voapp.ru:5060;branch=z9hG4bKfc06.4b118272.0 Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone" From: <sip:78142764164@62.33.22.11>;tag=A515D068-227D To: <sip:78142799233@voapp.ru>;tag=bbaac0e818284ff5 Date: Fri, 04 Aug 2006 11:37:22 GMT Call-ID: 64A759D3-22E411DB-8B0DFF2E-66029374(a)195.161.136.114 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1688609156-585372123-2332753710-1711444852 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 102 INVITE Max-Forwards: 5 Remote-Party-ID: <sip:78142764164@62.33.22.11>;party=calling;screen=yes;privacy=off Timestamp: 1154691442 Contact: <sip:78142764164@62.33.22.11:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 393 v=0 o=CiscoSystemsSIP-GW-UserAgent 4330 9656 IN IP4 62.33.22.11 s=SIP Call c=IN IP4 62.33.22.11 t=0 0 m=image 17088 udptl t38 c=IN IP4 62.33.22.11 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy As you can see the nathelper module has not worked since the field c=IN IP4 62.33.22.11 has not changed. Probably it has taken place because m=image instead of m=audio as usual. As a result of transfer of a fax has not taken place. If to place UA outside for NAT router all works that once again confirms that bug is in the nathelper module. Questions: Why the module behaves so? What difference that to proxing (what byte stream and in what format)? How it can be bypassed? Also that the most interesting - UA refuses to accept T38 and suggests to use instead of it G.711 codec and the gateway agrees i.e. in result we have audio stream. Dmitry

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[Users] Error REGISTERing and adding users to OpenSER 1.1.0 on Solaris
by peter.3.edwards＠bt.com 14 Aug '06

14 Aug '06

Hi, I'm trying to install OpenSER 1.1.0 (rel_1_1_0 checked out from CVS) on SPARC/Solaris 9. I'm dutifully following a combination of the wiki instructions: Install And Maintain OpenSER From CVS http://www.openser.org/dokuwiki/doku.php?id=install_and_maintain_openser _from_cvs and section D) OpenSER with Persistent Data Storage from the Install page of the main site. I've installed OpenSER with MySQL / Auth support correctly at each stage, as far as I can tell, and it is now running (can be seen with "ps -ef | grep openser"). My first attempt at running "openserctl moni" failed because of line 119 in /usr/local/lib64/openser/openserctl/openserctl.fifo: attempt=$(($attempt + 1)) So I changed this to to the more Solaris-friendly: attempt=`expr $attempt + 1` which seems to work. However, I'm unable to REGISTER the admin:openserrw user with my eyeBeam SIP user agent (#401 Unauthorised), and when I try to add a new user I get the following error: # openserctl add testuser 1234 test(a)mydomain.org awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 /usr/local/sbin/openserctl: md5sum: not found /usr/local/sbin/openserctl: md5sum: not found awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 /usr/local/sbin/openserctl: md5sum: not found MySql password for user 'openser@localhost': ERROR 1062 (23000) at line 1: Duplicate entry '' for key 2 -e \E[37;31mERROR: introducing the new user 'testuser' to the database failed I'm running out of ideas, now. Can anyone give me some pointers? Has anyone compiled from src with Solaris and got this working? I think this looks like there maybe a MySQL interop issue in openserctl? Any help appreciated, Peter. -- Peter This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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RE: [Users] Handle_subscription crashes openser
by Pletli Antal 14 Aug '06

14 Aug '06

Thank you for your quick answer. I have tried with db_use=0. In this case it does work but my clients don't show the change of presence. I think it depends on capability of SIP client too... -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: Monday, August 14, 2006 4:29 PM To: Pletli Antal Cc: users(a)openser.org Subject: Re: [Users] Handle_subscription crashes openser Hello, On 08/14/06 14:01, Pletli Antal wrote: > Hi, > > I use the latest openser from cvs with presence support with pa module. > I have experienced that the SUBSCRIBE message of the new x-lite (based > on eyeBeam) crashes the proxy in handle_subscription('registra > <<crash.txt>> r') function. > > It is a known problem in openser or it is a X-Lite bug? > might be a problem of OpenSER. The presence support is being refurbished completely. Expect new code to be uploaded in the future, as we get the basics working. The old pa module is deprecated, none in working on it. Now we are following the latest specifications in this area with the new development, specifications which are quite heavy. Best regards, Daniel > Best regards, > > Antal Pletli > > > ---------------------------------------------------------------------- > -- > > Aug 14 11:13:28 ship1 openser[32283]: Maxfwd module- initializing Aug > 14 11:13:28 ship1 openser[32283]: Parsed mimetype text/xml+msrtcp.pidf > got 100ff expected 1000c Aug 14 11:13:28 ship1 openser[32283]: > db_url=0x9c870d0 Aug 14 11:13:28 ship1 openser[32283]: > db_url=mysql://openser:openserrw@localhost/openser > Aug 14 11:13:28 ship1 openser[32283]: db_url.len=43 Aug 14 11:13:28 > ship1 openser[32283]: pa_mod: use_db=1 > db_url.s=mysql://openser:openserrw@localhost/openser > Aug 14 11:13:28 ship1 openser[32283]: pa_mod_init done Aug 14 11:13:28 > ship1 openser[32283]: subscribe_fixup: pdomain name is registrar Aug > 14 11:13:28 ship1 openser[32283]: subscribe_fixup: pdomain name is > registrar Aug 14 11:13:30 ship1 openser[32299]: DEBUG - MSILO: offline > messages dumped - if they were Aug 14 11:13:30 ship1 openser[32290]: > handle_subscription() entered Aug 14 11:13:30 ship1 openser[32290]: > check_message -0- _m=0x8130408 Aug 14 11:13:30 ship1 openser[32290]: > check_message -1- Aug 14 11:13:30 ship1 openser[32290]: check_message > -2- accepts_mimes=(nil) Aug 14 11:13:30 ship1 openser[32290]: > check_message -3- Aug 14 11:13:30 ship1 openser[32290]: check_message > -4- parsed_event=0x81241d0 Aug 14 11:13:30 ship1 openser[32290]: > check_message -5- Aug 14 11:13:30 ship1 openser[32290]: get_pres_uri: > _puri=sip:700046@xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32290]: > handle_subscription(): -1- Aug 14 11:13:30 ship1 openser[32296]: > handle_subscription() entered Aug 14 11:13:30 ship1 openser[32296]: > parsing accept header Aug 14 11:13:30 ship1 openser[32296]: > check_message -0- _m=0x8130408 Aug 14 11:13:30 ship1 openser[32296]: > check_message -1- Aug 14 11:13:30 ship1 openser[32290]: > new_presentity_no_wb=0xbd6851c0 for uri=700053(a)xxx.xxx.xxx.xxx Aug 14 > 11:13:30 ship1 openser[32296]: pa check_message: > accept=application/watcherinfo+xml parsed=:0x3000a Aug 14 11:13:30 > ship1 openser[32290]: add_presentity _p=0xbd6851c0 > p_uri=700053(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: > check_message -2- accepts_mimes=0x8123cb8 Aug 14 11:13:30 ship1 > openser[32290]: new_presentity_no_wb=0xbd685218 for > uri=700053-list(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: > check_message -3- Aug 14 11:13:30 ship1 openser[32290]: add_presentity > _p=0xbd685218 p_uri=700053-list(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 > openser[32296]: check_message -4- parsed_event=0x8123c88 Aug 14 > 11:13:30 ship1 openser[32290]: new_presentity_no_wb=0xbd685270 for > uri=700046(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: > check_message -4- eventtype=0x0002 Aug 14 11:13:30 ship1 > openser[32290]: add_presentity _p=0xbd685270 > p_uri=700046(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: > check_message -4a- eventtype=0x2 epm[i].event_type=0x1 Aug 14 11:13:30 > ship1 openser[32290]: new_presentity_no_wb=0xbd685310 for > uri=700046(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: > check_message -4a- eventtype=0x2 epm[i].event_type=0x2 Aug 14 11:13:30 > ship1 openser[32290]: add_presentity _p=0xbd685310 > p_uri=700046(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: check_message -4c- eventtype=0x2 mimetype=0x3000a accepts_mimes[k]=0x3000a Aug 14 11:13:30 ship1 openser[32290]: new_presentity_no_wb=0xbd685368 for uri=700053(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: check_message -4b- eventtype=0x2 accepts_mime=0x3000a Aug 14 11:13:30 ship1 openser[32290]: add_presentity _p=0xbd685368 p_uri=700053(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32296]: get_pres_uri: _puri=sip:700046@xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32290]: new_presentity_no_wb=0xbd6853c0 for uri=700053-list(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32290]: add_presentity _p=0xbd6853c0 p_uri=700053-list(a)xxx.xxx.xxx.xxx Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: _p->uri='700053(a)xxx.xxx.xxx.xxx' > Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: > _p->uri='700053(a)xxx.xxx.xxx.xxx' done Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: _p->uri='700053-list(a)xxx.xxx.xxx.xxx' > Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: > _p->uri='700053-list(a)xxx.xxx.xxx.xxx' done Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: _p->uri='700046(a)xxx.xxx.xxx.xxx' > Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: > _p->uri='700046(a)xxx.xxx.xxx.xxx' done Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: _p->uri='700046(a)xxx.xxx.xxx.xxx' > Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: > _p->uri='700046(a)xxx.xxx.xxx.xxx' done Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: _p->uri='700053(a)xxx.xxx.xxx.xxx' > Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: > _p->uri='700053(a)xxx.xxx.xxx.xxx' done Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: _p->uri='700053-list(a)xxx.xxx.xxx.xxx' > Aug 14 11:13:30 ship1 openser[32290]: db_read_watcherinfo: > _p->uri='700053-list(a)xxx.xxx.xxx.xxx' done Aug 14 11:13:30 ship1 > openser[32290]: handle_subscription(): -3- Aug 14 11:13:30 ship1 > openser[32290]: new_watcher starting Aug 14 11:13:30 ship1 > openser[32290]: new_watcher: _p->uri=700046(a)xxx.xxx.xxx.xxx Aug 14 > 11:13:30 ship1 openser[32290]: db_new_watcher: > watcher->uri=700046(a)xxx.xxx.xxx.xxx > Aug 14 11:13:30 ship1 openser[32290]: new_watcher: > watcher->package=unknown Aug 14 11:13:31 ship1 openser[32317]: ERROR: > receive_fd: EOF on 10 Aug 14 11:14:31 ship1 openser[32283]: BUG: shutdown timeout triggered, dying... > > > > ---------------------------------------------------------------------- > -- > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] Handle_subscription crashes openser
by Pletli Antal 14 Aug '06

14 Aug '06

Hi, I use the latest openser from cvs with presence support with pa module. I have experienced that the SUBSCRIBE message of the new x-lite (based on eyeBeam) crashes the proxy in handle_subscription('registra <<crash.txt>> r') function. It is a known problem in openser or it is a X-Lite bug? Best regards, Antal Pletli

2 1

[Serusers] Mediaproxy problem
by ram 14 Aug '06

14 Aug '06

Hi iam trying to integrate with SER and Media proxy for my NAt users Till now i have tested with all Public IP but my new requirement come with NAT users so install Media Server and when i start run this is the error i see in my messages.. iam running SER with Mysql Support. Aug 14 14:07:22 asterisk1 mediaproxy[4725]: warning: accounting is enabled but the accounting module is missing. accounting is not available! Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: Traceback (most recent call last): Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: File "/usr/local/mediaproxy/proxydispatcher.py", line 65, in ? Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: import dispatcher Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 39, in ? Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: from accounting import accounting, StopRecordSerializer, UnserializeError Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: File "/usr/local/mediaproxy/modules/accounting.py", line 14, in ? Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: from mysql import * Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: File "/usr/local/mediaproxy/modules/mysql.py", line 8, in ? Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: import MySQLdb Aug 14 14:07:22 asterisk1 mediaproxy[4725]: Listening for commands on local socket `/var/run/mediaproxy.sock' Aug 14 14:07:22 asterisk1 proxydispatcher[4728]: ImportError Aug 14 14:07:22 asterisk1 mediaproxy[4725]: Listening for remote commands is disabled Ram

1 0

[Serusers] How to build in SER modules statically for profiling?
by 邹嘉 12 Aug '06

12 Aug '06

Hi, all! I am currently involved in a research work which aims at optimizing the SIP stack, and we chose SER -0.9.6 as our researching platform. we need to profile SER to find out which part(parser? network? transactions? ) consumes the most CPU cycles. However, tools like gprof and Intel Vtune's sampling tool can not show any information about the functions in dynamically loaded modules, such as tm, registrar. So, a possible solution seems to build in these modules into the core statically. I tried to do what the INSTALL doc says, compling with the following command:"make static_modules=tm PROFILE=-pg all", the final part of the result shows: ...... make[1]:***[callid.o] Error 1 make[1]: Leaving directory '/....../SER/ser-0.9.6/modules/tm' gcc -Wl,-02 -Wl, -E -pg action.o crc.o .....modules/tm/dlg.o, modules/tm/h_table.o .....modules/tm/uac_unixsock.o - gcc: modules/tm/callid.o:No such file or directory gcc: modules/tm/h_table.o:No such file or directory ... gcc: modules/tm/uac_unixsock.o:No such file or directory make: **[ser] Error 1 And if I proceed to type: make install the result shows: sr_module.o(.text+0xbc): In function 'register_builtin_modules': /.../ser-0.9.6/sr_module.c:89: undefined reference to 'tm_exports' collect2:ld returned 1 exit status make: ***[ser] Error 1 I am working with SER 0.9.6 And the linux kernel used in my machine is: Linux 2.6.9-22.ELsmp the OS used is Redhat 9 if anyone has tried the same thing or knows the solution, please let me know.Thanks very very much! Jacqueline Zou

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Re: [Serusers] Error using serctl monitor
by Jiri Kuthan 12 Aug '06

12 Aug '06

which version of ser are you using? which version of serctl are you using? whats the output of 'serctl fifo which'? Could you set the "-x" option for serctl so that we see where serctl failed? Thanks -jiri At 14:24 01/08/2006, Josel Layno wrote: >Hi guys, > >I got my SER running "so far" and i trying to explore some commands using serctl. >Using serctl moni gives me error ... > > [: -ne: unexpected operator > >and using serctl domain show gives me... > > $ serctl domain show > 500 command 'domain_dump' not available > >Is this all crucial on my server? Did i missed something? >I appreciate any kind of assistance...:) > >regards, > >josel >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] How to build in SER modules statically for profiling?
by Jia Zou 12 Aug '06

12 Aug '06

Thank you for your kind help. However,my problem is to profile the shared libraries rather than to profile forking processes. gprof can not generate any information about these shared libraries, and i also seems problem when using sprof, which is said to be able to offer profiling statistics about dynamic loaded libraries. So, I think another way is to build these modules statically, but don't know how to. So can anyone help me with that? thanks for your time! Jacqueline Zou ======== 2006-08-11 14:56:12 您在来信中写道： ======== Your post belongs to serdev. Here is a serdev post from Hendrik Scholz some time back. g-) Hi! I'm about to profile our SER installation and since profiling doesn't run out of the box here's a rough guide: - you'll need to build SER from sources, thus get sources, apply your patches and so on - before compilation get this patch (for CVS HEAD): http://dump.wormulon.net/patches/ser-profiling.diff Call $ patch -p0 < ser-profiling.diff from the SER base directory. What it does it invoking monstartup() in each child process since the profiling information get lost when fork()ing. I added all childs including timer and fifoserver. If you are only interested in certain worker processes you can cut down the resource impact by selectively activating profiling. - enable profiling in Makefile.defs - Set both -DPROFILING and PROFILE=-pg - compile and deploy as usual - Before starting SER using your standard init script you need to set an environment variable that acts as a prefix to all profiling data files. If you don't set it you'll only get one file for the main process. # export GMON_OUT_PREFIX=testrun The gmon files will be named testrun.<PID> with the PID matching $ serctl fifo ps SER may change the current working directory and drop privileges on the way therefore I suggest you invoke the init script from the SER working directory (-w option) so you'll end up with all files in one place. - Run SER for a while. Not all gmon files will show up immediately, though. - You should now be able to get per-process statistics using $ gprof /usr/sbin/ser testrun.<PID> - If cummulative statistics are what you are looking for use $ gprof -s /usr/sbin/ser testrun.* to create a new gmon.out which contains statistics over all processes. I have yet to fully analyze the results. Cheers (and YMMV), Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de _______________________________________________ Serdev mailing list Serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev 邹嘉 wrote: Hi, all! I am currently involved in a research work which aims at optimizing the SIP stack, and we chose SER -0.9.6 as our researching platform. we need to profile SER to find out which part(parser? network? transactions? ) consumes the most CPU cycles. However, tools like gprof and Intel Vtune's sampling tool can not show any information about the functions in dynamically loaded modules, such as tm, registrar. So, a possible solution seems to build in these modules into the core statically. I tried to do what the INSTALL doc says, compling with the following command:"make static_modules=tm PROFILE=-pg all", the final part of the result shows: ...... make[1]:***[callid.o] Error 1 make[1]: Leaving directory '/....../SER/ser-0.9.6/modules/tm' gcc -Wl,-02 -Wl, -E -pg action.o crc.o .....modules/tm/dlg.o, modules/tm/h_table.o .....modules/tm/uac_unixsock.o - gcc: modules/tm/callid.o:No such file or directory gcc: modules/tm/h_table.o:No such file or directory ... gcc: modules/tm/uac_unixsock.o:No such file or directory make: **[ser] Error 1 And if I proceed to type: make install the result shows: sr_module.o(.text+0xbc): In function 'register_builtin_modules': /.../ser-0.9.6/sr_module.c:89: undefined reference to 'tm_exports' collect2:ld returned 1 exit status make: ***[ser] Error 1 I am working with SER 0.9.6 And the linux kernel used in my machine is: Linux 2.6.9-22.ELsmp the OS used is Redhat 9 if anyone has tried the same thing or knows the solution, please let me know.Thanks very very much! Jacqueline Zou _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers = = = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　　　　　　　Jia Zou 　　　　　　　　　　　　　　zouj03(a)mails.tsinghua.edu.cn 　　　　　　　　　　　　　　　2006-08-12

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[Users] LCR load_contacts and load_gw
by Helge Waastad 11 Aug '06

11 Aug '06

Hei, I wonder if someone could comment on this: I have several cisco gateways distributed all over and are using the lcr module for this. Is it possible to combine both load_contacts/next_contact and load_gw/next_gw (I have softphones also registered with the same numbers that are behind the gateways.....) Would it be interesting to have regexp in prefix matching aswell? ex: 4675[1-2] ? br hw

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[Users] lcr load_gws()
by Sharon 11 Aug '06

11 Aug '06

Hey all, Can anyone post any example config with successful use of load_gws() and next_gw() . I am trying to set it up for my system here but don't know what I am missing.Also if possible how do you log the loaded load_gws() and next_gw() in your logs. Thanks, -Sharon

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RE: [Serusers] Looking for Call termination service provider
by John Kim 11 Aug '06

11 Aug '06

We are in USA and we are targeting the calls terminated in South Korea. -----Original Message----- From: ram [mailto:talk2ram@gmail.com] Sent: Thursday, August 10, 2006 11:27 AM To: John Kim Subject: Re: [Serusers] Looking for Call termination service provider Hi where are you from ? I can asissts you Ram On 8/10/06, John Kim <johnk(a)koreanet.us> wrote: Hi, I am looking for a VoIP wholesaler who will allow us to operate our own SIP server infrastructure while we can forward all PSTN destined calls to their gateway. I really didn't think that it would be hard to find one but I have not been successful. I have checked so far with iCallGlobe.com and Deltathree.com <http://Deltathree.com> . Can any of you recommend us some VoIP wholesaler with quality service and competitive rate who allows this business setup? Below is logical business relation setup: Our subscriber <--->ATA setup to connect sip.ours.com<----->PSTN gateway of wholesaler.com<--->PSTN Basically we are looking for good " wholesaler.com <http://wholesaler.com> ". Any comments will be appreciated. John K. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] ser SDP rewrite problem, rtpproxy is not working !!!
by Sebastian Gabris 11 Aug '06

11 Aug '06

Hi, got also this problem, did you find a solution? Sebastian hello there, I am working on SER with two network interfaces. EP1--------------->(172.16.16.160) SER (192.168.5.1) -------------> EP2 what happens, SER receives invite from EP1, it reinitiates invite for EP2. now what it does is that it writes the external IP i.e. 172.16.16.160 in the SDP contact info, where as it should be the internal one i.e. 192.168.5.1 ... now I want to overwrite the SDP contact info in that invite. I have got two options to do it one option is to use force_rtp_proxy("ei", "192.168.5.1") or use sdp_mangle_ip() function of mangler.so . but they are not working for me. when I play around with force_rtp_proxy function rtpproxy get crashed and sdp_mangle_ip() is not behaving properly ...... whats your experience ? I have used ser-0.8.14 and now using ser-0.9.0 with cvs rtpproxy. I am running rtpproxy like rtpproxy -s udp:192.168.5.1 -l 172.16.16.160 and I am using this line in ser.cfg modparam("nathelper", "rtpproxy_sock", "udp:192.168.5.1") I am really stucked please help !!!! thanking you regards, -- Atif ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

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[Users] extract substring?
by zm23 11 Aug '06

11 Aug '06

Hi all, Is there a way to extract a substring from uri and save that value? I see subst function in textops module but that is only to substitute. thanks. -- Zahid Mehmood CUIT Network Systems

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[Serusers] Problem with incoming calls
by Sebastian Gabris 11 Aug '06

11 Aug '06

Hi everybody, Like told in the subject i got problems with incomnig calls, SIP signalling works but RTP doesnt. When I make a call from my UA to the VoIP-Provider all works well, sip & rtp. I start rtpproxy in bridged mode: rtpproxy -l publicIP/privateIP. I use this scenario: UA(private IP) <-> (privateIP) SER (publicIP) <-> VoIP-Provider i try to figure out where my mistake is in this config file: route[1] { if(dst_ip == 10.245.104.68) { force_rtp_proxy("","193.77.54.68"); } else { force_rtp_proxy("","10.245.104.68"); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { if(dst_ip == 10.245.104.68) { force_rtp_proxy("","193.77.54.68"); } else { force_rtp_proxy("","10.245.104.68"); }; }; } By the way, is there a option to view rtpproxy debug info? Thanks in advance! Sebastian ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

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[Users] I can`t make a calling by using xlite at the client. Please help me...Please
by Ferianto siregar 11 Aug '06

11 Aug '06

Dear all, I would like to say thanks for everybody help and kind-hearted who always give me some helps to solve my problem. Thank you very much. Now, I have a problem in my openser. I can`t make a calling from client using the xlite. It is meant that the client can not make a call by using xlite ( It can not connect to server) . In xlite screen, it always shows "403 forbidden" message. Does anybody can give me a suggesstion what is meant? Where the problem comes from? Please help me ...Please. For the information: 1. The client use IP 202.95.149.2 2. This is the log message that I have got from xlite diagnostic: SEND TIME: 3836510 SEND >> 202.95.149.250:5060 REGISTER sip:pcr.ac.id SIP/2.0 Via: SIP/2.0/UDP 202.95.149.40:5060;rport;branch=z9hG4bKC098998FE862469BBE0A3F218390D2B1 From: aldi <sip:123@pcr.ac.id>;tag=4145118495 To: aldi <sip:123@pcr.ac.id> Contact: "aldi" <sip:123@202.95.149.40:5060> Call-ID: AF8EEFDE9AD94CC5B910233C4AFA09BF(a)pcr.ac.id CSeq: 5795 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105x Content-Length: 0 RECEIVE TIME: 3836512 RECEIVE << 202.95.149.250:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 202.95.149.40:5060;rport=5060;branch=z9hG4bKC098998FE862469BBE0A3F218390D2B1 From: aldi <sip:123@pcr.ac.id>;tag=4145118495 To: aldi <sip:123@pcr.ac.id>;tag=98a2bfb09e790c090877fa710657efb6.0d98 Call-ID: AF8EEFDE9AD94CC5B910233C4AFA09BF(a)pcr.ac.id CSeq: 5795 REGISTER Server: OpenSer (1.1.0-tls (i386/linux)) Content-Length: 0 Warning: 392 202.95.149.250:5060 "Noisy feedback tells: pid=4439 req_src_ip=202.95.149.40 req_src_port=5060 in_uri=sip:pcr.ac.id out_uri=sip:pcr.ac.id via_cnt==1" 3.What is the meaning : " 2(4454) is_local(): Realm 'pcr.ac.id' is not local"? I got it when I run "openser" command. This is the message that I got: get_hdr_field: cseq <CSeq>: <1497> <REGISTER> 2(4454) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(4454) parse_headers: flags=200 2(4454) DEBUG: get_hdr_body : content_length=0 2(4454) found end of header 2(4454) find_first_route: No Route headers found 2(4454) loose_route: There is no Route HF 2(4454) is_local(): Realm 'pcr.ac.id' is not local 2(4454) DEBUG: add_param: tag=888227406 2(4454) DEBUG:parse_to:end of header reached, state=29 2(4454) DBUG:parse_to: display={123}, ruri={sip:123@pcr.ac.id} 2(4454) is_local(): Realm 'pcr.ac.id' is not local 2(4454) parse_headers: flags=ffffffffffffffff 2(4454) check_via_address(202.95.149.2, 202.95.149.2, 0) 2(4454) DEBUG:destroy_avp_list: destroying list (nil) 2(4454) receive_msg: cleaning up 4(4460) udp_rcv_loop: probing packet received from 202.95.149.2 50195 5(4461) udp_rcv_loop: probing packet received from 202.95.149.2 50195 2(4454) udp_rcv_loop: probing packet received from 202.95.149.2 50195 4.This is the contain of mysql: mysql> select * from aliases; +----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+| username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | user_agent | socket | methods |+----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+| 123 | pcr.ac.id | sip:coba@pcr.ac.id | NULL | NULL | 2006-08-09 02:53:59 | 1.00 | The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything | 42 | 1970-01-01 07:00:00 | 128 | OpenSER Server FIFO | NULL | NULL || 456 | pcr.ac.id | sip:feri@pcr.ac.id | NULL | NULL | 2006-08-09 06:57:28 | 1.00 | The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything | 42 | 1970-01-01 07:00:00 | 128 | OpenSER Server FIFO | NULL | NULL |+----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+2 rows in set (0.00 sec) I do hope anybody can help me. I have tried to solve this problem 2 days but I can`t. Please help me..please.. Regards with cheers, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1ï¿½/min. --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

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Re: [Serusers] presence_b2b UA
by Vaclav Kubart 11 Aug '06

11 Aug '06

Hi Curt, are presence subscriptions enough? PA module is prepared to generate internal (QSA) subscription when it is processing SUBSCRIBE request. But it still can not handle responses - this will be done soon. I think that this will be what do you need. I hope that I will have time for it and finish it till the end of August. Vaclav On Wed, Aug 09, 2006 at 02:18:07PM -0500, Curt Moore wrote: > Hi Vaclav. > > Thank you for your reply. > > This sounds very close to what I'm trying to accomplish for presence. > I have 1 SER server setup as the main proxy/registrar. If a SUBSCRIBE > message comes into this server, it is forwarded over to a second SER > server which maintains the watcher info, a presence server. This > currently works perfectly. > > What I'd like to happen is for the B2B UA on the presence server to > generate a new SUBSCRIBE message and send it out to an Asterisk > server, essentially having the SER presence server subscribe to the > presence of the SIP user on the Asterisk server. If the presence > changes on the Asterisk server, it will send a NOTIFY back to the > presence SER server. This should then, as you mention, trigger a > notification through QSA back to any watchers watching this particuar > resource. > > My confusion is how to get the outbound SUBSCRIBE generated through > QSA and sent to the Asterisk server from the presence server after the > initial inbound SUBSCRIBE comes into the SER presence server. > > How would I get the SER server to generate the outbound SUBSCRIBE? > > Thanks! > -Curt > > On 8/9/06, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > >Hi, > >presence_b2b processes INTERNAL subscriptions (through QSA - internal > >API for status querying) and generates SIP subscriptions according to > >them (now only presence subscriptions). Each arriving NOTIFY triggers > >notification through QSA. > > > >It works quite good with one SER as resource list server and another as > >presence server. > > > > Vaclav > > > >On Wed, Aug 09, 2006 at 12:01:59PM -0500, Curt Moore wrote: > >> Hello all. > >> > >> I have been experimenting with the ser-0.10.99-xxx branch(es) and have > >> a question about the presence_b2b module. > >> > >> I realize that the module is not yet fully complete but according to > >> the documentation, when finished, the module should be able to take an > >> inbound SUBSCRIBE request and then create a new SUBSCRIBE request > >> outbound to another SIP entity or multiple SIP entities, ie Asterisk > >> servers, hence the back to back functionality. What is the status of > >> this functionality? Is anyone currently working to complete this > >> functionality? I would be interested in helping complete or > >> sponsoring the completion of this functionality. > >> > >> My intent is to have a SER server act as the central watcher database > >> server and have all SUBSCRIBE and NOTIFY requests channeled through > >> this SER server. This SER server will then generate new SUBSCRIBE > >> requests out to each server in a farm of Asterisk servers and the > >> Asterisk servers will then send NOTIFIES back to the SER watcher > >> server. The watcher server will then take care of sending the NOTIFY > >> messages to the correct watcher(s), according to the data in its > >> watcher database. > >> > >> Is anyone currently doing anything similar to this using a different > >> method or is using the yet to be completed B2B UA the correct/only way > >> to accomplish this? > >> > >> Thanks, > >> -Curt > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] Accounting Calls not by extension
by Nhadie Ramos 11 Aug '06

11 Aug '06

Hi All, I'm doing accounting via the usage of per extension. Can i also do it when a client is a gateway and would forward calls to my SER? e.g. from their gateway, the will append everything infront of their dialled digits with 12345#, then forward it to my SER. then i'll strip the 12345# then forward it to my gateway provider. how can i account anything that has 12345# or can I account it via the IP address of their gateway? TIA Regards, nhadie

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[Serusers] to use SER stateless proxy..................
by dillip parida 11 Aug '06

11 Aug '06

hi all, I am newbie to SER. actually I want to use SER as stateless proxy.. i will explain with a example......user agent A and User agent B are two SIP phones. if A calls , B's number initially it reaches at SER, then SER route the call stalelessly to particular URI. BUT I DONT NEED THAT USER 'A' and USER 'B' SHOULD NOT GET REGISTERED AT SER. without registering at SER we need to route the call to that particular URI....i mean to that user B..... please help me.......i have attached the ser.cfg for ur refrence.

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Re: Re: [Users] two phones ringing
by Mark Kent 10 Aug '06

10 Aug '06

>> By the way, when you answer very quickly, did the other phone ring at >> least once before? The phone I answered rang once, the slow phone did not. If they happen to both ring at the same time then it works, it's just the times that the slow phone lags that causes the problem. Thanks, -mark

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Re: [Serusers] Multihomed SER & NAT
by Sebastian Gabris 10 Aug '06

10 Aug '06

Thanks for the response, will try it and see what happens. :-) While waiting for help, i tried a new config, without fix_nat_contact. It works (Audio & signalling) for outgoing calls.(private -> public) Contact header has a private IP but SDP information is correct due to RTPProxy. At incoming calls the force_rtp_proxy command didn't change anything, meaning that SDP Information is not correct -> no audio, only signalling. I tried to start rtpproxy without options and in bridged mode. Also in bridged mode i tried both possibilities -l private/public & -l public/private, but it didn't influence the outcome. Perhaps someone could help me out on this. :-) Here are the changes to my config. route[1] { if(dst_ip == 10.x.x.x) { force_rtp_proxy("","193.x.x.x"); }; if (dst_ip == 193.x.x.x) { force_rtp_proxy("","10.x.x.x"); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { if(dst_ip == 10.x.x.x) { force_rtp_proxy("e"); log(2,"RTP_PROXY: Reply-Route1 dst 10.x.x.x\n"); }; if (dst_ip == 193.x.x.x) { force_rtp_proxy("i"); log(2,"RTP_PROXY: Reply-Route1 dst 193.x.x.x\n"); }; }; } fix_nated_contact() is meant for changing the IP address of the Contact > header when SER has a public address and the UA has a private (i.e. > behind NAT). This means that it will rewrite Contact to contain the > source ip address:port found in the UDP/TCP header. > This is probably correct, because I assume the UA cannot be reached > from the outside anyway. If you record-route, subsequent requests > should go through your ser. > > If you have problems with responses not being returned, have a look at > Via headers as they control how answers to i.e an INVITE are routed > back to the UA. You may want to have a look at the rr module double > record route parameter to make sure that SER creates one record-route > for the internal and one for the external interface. > g-) > > Sebastian Gabris wrote: > After reading a bit more mails, I came to two possible problems: > 1. fix_nated_contact() needs a registrar to work properly > I dont use a registrar, since i register at the VoIP - Provider. > or > 2. fix_nated_contact() is not ment to be used on this scenario > Any ideas on this? > Sebastian > > > Hi everybody, > need some help on nattraversal. > i use a multihomed SER, one private IP & one public IP. > MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) > If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address. > Instead o writing down his public IP, SER writes his private IP. > Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information. > After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :) > Here are the important bits of my config: > fork=yes > log_stderror=no mhomed=1 > listen=10.x.x.x > listen=193.x.x.x > loadmodule "/usr/lib/ser/modules/nathelper.so" > modparam("usrloc", "db_mode", 0) > modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > if (nat_uac_test("3")) { > if (method == "REGISTER" || ! search("^Record-Route:")) { > log(2,"LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling > if (method == "INVITE") { > fix_nated_sdp("1"); # Add direction=active to SDP > }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > ... and so on. > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > _____________________________________________________________________ > Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! > http://smartsurfer.web.de/?mc=100071&distributionid=000000000071 > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

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[Serusers] Same aliases for diferent subscribers
by Ricardo Carvalho 10 Aug '06

10 Aug '06

Hi, Is it possible to give same alias to different subscribers? Trying to add same alias in serctl isn't well succeeded because it says that that alias is already in use by other user. Is it any other way to do it? Regards, Ricardo.

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[Serusers] How to change display name
by Steve Blair 10 Aug '06

10 Aug '06

Is it possible to change just the display name portion of the from header? I'm trying to implement caller ID blocking but we cannot remove the actual from URI because that breaks our gateway based CDR generation. I'm hoping changing the from header to something like: from="Anonymous" <sip:376826@myschool.edu>;tag=003094 Thanks, Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

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Re: [Serusers] Multihomed SER & NAT
by Sebastian Gabris 10 Aug '06

10 Aug '06

After reading a bit more mails, I came to two possible problems: 1. fix_nated_contact() needs a registrar to work properly I dont use a registrar, since i register at the VoIP - Provider. or 2. fix_nated_contact() is not ment to be used on this scenario Any ideas on this? Sebastian > Hi everybody, > > need some help on nattraversal. > > i use a multihomed SER, one private IP & one public IP. > > MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) > > If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address. > Instead o writing down his public IP, SER writes his private IP. > Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information. > > After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :) > > Here are the important bits of my config: > > fork=yes > log_stderror=no > > mhomed=1 > listen=10.x.x.x > listen=193.x.x.x > > loadmodule "/usr/lib/ser/modules/nathelper.so" > modparam("usrloc", "db_mode", 0) > > modparam("registrar", "nat_flag", 6) > modparam("nathelper", "natping_interval", 30) # Ping interval 30 s > modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT > > if (nat_uac_test("3")) { > > if (method == "REGISTER" || ! search("^Record-Route:")) { > log(2,"LOG: Someone trying to register from private IP, rewriting\n"); > > fix_nated_contact(); # Rewrite contact with source IP of signalling > if (method == "INVITE") { > fix_nated_sdp("1"); # Add direction=active to SDP > }; > force_rport(); # Add rport parameter to topmost Via > setflag(6); # Mark as NATed > }; > }; > > ... and so on. > > > > ______________________________________________________________ > Verschicken Sie romantische, coole und witzige Bilder per SMS! > Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193 > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

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[Users] Troubles parsing headers
by Lars Sundqvist 10 Aug '06

10 Aug '06

Hi everyone, We have been having a problem with our openser installation for a while, certain REGISTER requests generates errors. The openser version we are running is 1.0.0 from cvs. Does anyone have a clue why the following register generates the error below? Register: xxx.xxx.xx.xx:5061 -> yyy.yyy.yyy.yyy:5060 REGISTER sip:sip.xxxxx.xxx SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xx.xx:5061;branch=z9hG4bK-39f541a2..From: aaaa.bbbbb <sip: username123(a)sip.xxxxx.xxx>;tag=ea 8cef4b108a99bco1..To: aaaa.bbbbb <sip: username123(a)sip.xxxxx.xxx>..Call-ID: b96fead3-f03493d4@xxx.xxx.xx.xx..CSeq: 2508 REGISTER..Max-Forwards: 70..Conta ct: aaaa.bbbbb <sip: username123@xxx.xxx.xx.xx:5061>;expires=3600..User-Agent: Linksys/RT31P2-2.0.10(LIc)..Content-Length: 0..Allow: ACK, BYE, CANCEL, IN FO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura.... Error: /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR:parse_from_header: bad from header /sbin/openser[18198]: xl_get_from: ERROR cannot parse FROM header /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: xl_get_to: ERROR cannot parse TO header /sbin/openser[18198]: [REGISTER] from [<null>] to [<null>] /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! /sbin/openser[18198]: ERROR:parse_from_header: bad from header /sbin/openser[18198]: insert_RR(): From parsing failed /sbin/openser[18198]: record_route(): Error while inserting Record-Route line /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: find_first_route: Error while parsing headers /sbin/openser[18198]: analyzing REGISTER request /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: find_credentials(): Error while parsing headers /sbin/openser[18198]: pre_auth(): Error while looking for credentials /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed /sbin/openser[18198]: pre_auth(): Error while sending 400 reply Many thanks, Lars Sundqvist

2 6

[Serusers] Multihomed SER & NAT
by Sebastian Gabris 10 Aug '06

10 Aug '06

Hi everybody, need some help on nattraversal. i use a multihomed SER, one private IP & one public IP. MyUA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) If i try to make a call from private net to public net, SER doesnt change the contact address. He changes the sdp - connection field but he, puts in the wrong IP - address. Instead o writing down his public IP, SER writes his private IP. Then SER forwards the call on his public interface to the provider, which will get wrong Contact & SDP information. After reviewing my logs i saw that SER does the nat_uac_test, but the fix_nat_contact & fix_sdp dont work dont do what they should do :) Here are the important bits of my config: fork=yes log_stderror=no mhomed=1 listen=10.x.x.x listen=193.x.x.x loadmodule "/usr/lib/ser/modules/nathelper.so" modparam("usrloc", "db_mode", 0) modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT if (nat_uac_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { log(2,"LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; ... and so on. ______________________________________________________________ Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=021193

1 0

[Users] two phones ringing
by Mark Kent 10 Aug '06

10 Aug '06

I've got two Linksys phones (one spa3000, one spa942) and registered markkent@... from both on the openser1.0.1 box. >From a third device I call markkent@... and both ring. If I take my time answering one phone then everything works as expected. The CANCEL gets sent to the second phone and that phone says OK and maybe I get one spurious ring but it stops. I noticed that if I answer one phone really quickly then the other phone will keep on ringing. So, I looked at the traffic and this is what happens: The players are ORIGINATOR, OPENSER, ANSWEREDPHONE and IGNOREDPHONE ORIGINATOR sends the INVITE to OPENSER where it gets branched out to both ANSWEREDPHONE and IGNOREDPHONE Both phones send back "100 Trying" ANSWEREDPHONE sends "180 Ringing" and OPENSER relays that to ORIGINATOR ANSWEREDPHONE sends "200 OK" when I answer, and OPENSER relays that to ORIGINATOR OPENSER sends "CANCEL sip:markkent@IGNOREDPHONE-IP:5062" to IGNOREDPHONE about 0.6 seconds later OPENSER repeats the CANCEL IGNOREDPHONE sends back to OPENSER 481 Call Leg/Transaction Does Not Exist and does it twice, perhaps once for each CANCEL Then IGNOREDPHONE sends back "180 Ringing" IGNOREDPHONE rings, a lot, but does not send back "180 Ringing" messages There are a lot of "200 OK" messages exchanged between the IGNOREDPHONE and the OPENSER box. The IGNOREDPHONE took 5 seconds to send back the first "180 Ringing", these phones are side by side, although IGNOREDPHONE (the 942) is NAT'ed and ANSWEREDPHONE is not. The ngrep output is here: http://a.mainstreet.net/double-ring.txt What I would like to know is whether this is something I can and/or should handle in the openser configuration? If so, how? Thanks, -mark

2 1

Re: [Serusers] Call between two SER Server
by Andrey Kouprianov 10 Aug '06

10 Aug '06

Hi, 1. In your "domain" table (i.e. MySQL) in both SER's you'd have to add domain entries. E.g. # serctl domain add you.domain.name.net 2. Then into" trusted" tables add entries as follows: a. For 192.168.100 --> Login to MySQL, and insert entries into SER's trusted table: mysql> insert into trusted values ('192.168.200.200', 'any', '^sip:.*$'); b. And vice versa. 3. After that you need to modify ur ser.cfg whenever INVITE method is encountered: E.g. ... if(!allow_trusted() && !is_uri_host_local() ) { if (!proxy_authorize("you.domain.name.net","subscriber")) { proxy_challenge("you.domain.name.net","0"); break; } else if (!check_from()) { xlog("L_WARN", "[%Tf]:%rm: possible uri spoofing attempt!; IP=%is; from=%fu; to=%tu\n"); sl_send_reply("403", "Forbidden"); break; }; consume_credentials(); }; .... Before saving make sure these modules are properly loaded: uri_db.so, domain.so, permissions.so and xlog.so. 4. Then restart your restart your SER Hope that works. Andrey. On 8/1/06, Asep Saifudin <aseps2000(a)gmail.com> wrote: > Andrey, > > I have try to put in my ser.cfg like this below but still doesnt work. > in my first SER Server : > > if (!proxy_authorize("192.168.200.200", "subscriber")) { > proxy_challenge("192.168.200.200", "0"); > break; > }; > > and in my secondary SER server : > > if (!proxy_authorize(" 192.168.100.100", "subscriber")) { > proxy_challenge("192.168.100.100", "0"); > break; > }; > > Really need further advice from you or anyone who have implement for call > between 2 SER Server ? > > Thanks, > Asep > > 2006/8/1, Asep Saifudin <aseps2000(a)gmail.com>: > > > > can you give me step by step what should I do for this ? or sample for > this scheme ? > > And should I add my first SER server domain in my secondary SER server > domain table (in MySQL server) or how because I still dont undestand with > your explanation ? > > > > > > > > 2006/8/1, Andrey Kouprianov <andrey.kouprianov(a)gmail.com >: > > > > > Make your ser.cfg on receiving side configured so that it does NOT > > > authorize incoming calls from different domains. Otherwise, the caller > > > will always be rejected. > > > > > > On 8/1/06, Asep Saifudin < aseps2000(a)gmail.com> wrote: > > > > Hi, > > > > > > > > I have two running SER Server but I stil got problem to make call from > my > > > > first SER server to my secondary SER server. > > > > > > > > My first SER server info : > > > > 1. IP address = 192.168.100.100 > > > > 2. sample subscriber number is 10999 > > > > > > > > My second SER Server info : > > > > 1. IP address = 192.168.200.200 > > > > 2. sample subscriber number is 20099 > > > > > > > > My question is : > > > > How suberiber 10099 can call to subscriber 20099 ? > > > > What additional config should I put in each ser.cfg ? or > > > > should I use gatekeeper/softswitch ? > > > > > > > > Thanks > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > >

2 1

[Users] I can`t make a calling by using xlite at the client. Please help me...Please
by Ferianto siregar 10 Aug '06

10 Aug '06

Dear all, I would like to say thanks for everybody help and kind-hearted who always give me some helps to solve my problem. Thank you very much. Now, I have a problem in my openser. I can`t make a calling from client using the xlite. It is meant that the client can not make a call by using xlite ( It can not connect to server) . In xlite screen, it always shows "403 forbidden" message. Does anybody can give me a suggesstion what is meant? Where the problem comes from? Please help me ...Please. For the information: 1. The client use IP 202.95.149.2 2. This is the log message that I have got from xlite diagnostic: SEND TIME: 3836510 SEND >> 202.95.149.250:5060 REGISTER sip:pcr.ac.id SIP/2.0 Via: SIP/2.0/UDP 202.95.149.40:5060;rport;branch=z9hG4bKC098998FE862469BBE0A3F218390D2B1 From: aldi <sip:123@pcr.ac.id>;tag=4145118495 To: aldi <sip:123@pcr.ac.id> Contact: "aldi" <sip:123@202.95.149.40:5060> Call-ID: AF8EEFDE9AD94CC5B910233C4AFA09BF(a)pcr.ac.id CSeq: 5795 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105x Content-Length: 0 RECEIVE TIME: 3836512 RECEIVE << 202.95.149.250:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 202.95.149.40:5060;rport=5060;branch=z9hG4bKC098998FE862469BBE0A3F218390D2B1 From: aldi <sip:123@pcr.ac.id>;tag=4145118495 To: aldi <sip:123@pcr.ac.id>;tag=98a2bfb09e790c090877fa710657efb6.0d98 Call-ID: AF8EEFDE9AD94CC5B910233C4AFA09BF(a)pcr.ac.id CSeq: 5795 REGISTER Server: OpenSer (1.1.0-tls (i386/linux)) Content-Length: 0 Warning: 392 202.95.149.250:5060 "Noisy feedback tells: pid=4439 req_src_ip=202.95.149.40 req_src_port=5060 in_uri=sip:pcr.ac.id out_uri=sip:pcr.ac.id via_cnt==1" 3.What is the meaning : " 2(4454) is_local(): Realm 'pcr.ac.id' is not local"? I got it when I run "openser" command. This is the message that I got: get_hdr_field: cseq <CSeq>: <1497> <REGISTER> 2(4454) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(4454) parse_headers: flags=200 2(4454) DEBUG: get_hdr_body : content_length=0 2(4454) found end of header 2(4454) find_first_route: No Route headers found 2(4454) loose_route: There is no Route HF 2(4454) is_local(): Realm 'pcr.ac.id' is not local 2(4454) DEBUG: add_param: tag=888227406 2(4454) DEBUG:parse_to:end of header reached, state=29 2(4454) DBUG:parse_to: display={123}, ruri={sip:123@pcr.ac.id} 2(4454) is_local(): Realm 'pcr.ac.id' is not local 2(4454) parse_headers: flags=ffffffffffffffff 2(4454) check_via_address(202.95.149.2, 202.95.149.2, 0) 2(4454) DEBUG:destroy_avp_list: destroying list (nil) 2(4454) receive_msg: cleaning up 4(4460) udp_rcv_loop: probing packet received from 202.95.149.2 50195 5(4461) udp_rcv_loop: probing packet received from 202.95.149.2 50195 2(4454) udp_rcv_loop: probing packet received from 202.95.149.2 50195 4.This is the contain of mysql: mysql> select * from aliases; +----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+| username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | user_agent | socket | methods |+----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+| 123 | pcr.ac.id | sip:coba@pcr.ac.id | NULL | NULL | 2006-08-09 02:53:59 | 1.00 | The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything | 42 | 1970-01-01 07:00:00 | 128 | OpenSER Server FIFO | NULL | NULL || 456 | pcr.ac.id | sip:feri@pcr.ac.id | NULL | NULL | 2006-08-09 06:57:28 | 1.00 | The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything | 42 | 1970-01-01 07:00:00 | 128 | OpenSER Server FIFO | NULL | NULL |+----------+-----------+--------------------+----------+------+---------------------+------+---------------------------------------------------------------------+------+---------------------+-------+---------------------+--------+---------+2 rows in set (0.00 sec) I do hope anybody can help me. I have tried to solve this problem 2 days but I can`t. Please help me..please.. Regards with cheers, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta. --------------------------------- Get your email and more, right on the new Yahoo.com

1 0

[Serusers] Crash Issues
by Suresh Arunachalam 10 Aug '06

10 Aug '06

Hello All, I am using ser-0.8.14 as the proxy and seeing the below given crash frequently. Please let me know what is causing the issue. The pool size is around 16mb. Loaded symbols for /lib/libnss_dns.so.2 #0 qm_malloc (qm=0x80cea20, size=16) at mem/q_malloc.c:239 239 FRAG_END(next)->prev_free=prev; (gdb) bt #0 qm_malloc (qm=0x80cea20, size=16) at mem/q_malloc.c:239 #1 0x080cea20 in mem_block () #2 0x08069dce in hostent_cpy (dst=0x80d0d74, src=0x80b8ce0) at proxy.c:133 #3 0x08069576 in mk_proxy (name=0xbfffe540, port=5060, proto=1) at proxy.c :228 #4 0x40296c41 in uri2proxy (uri=0xbfffe798, proto=16) at ut.h:120 #5 0x40295ab9 in uri2sock (uri=0x10, to_su=0xbfffe6b0, proto=16) at ut.h :141 #6 0x4029476a in build_dlg_ack (rpl=0xbfffe7a0, Trans=0x42533318, branch=16, to=0xbfffe748, len=0xbfffe7a0, next_hop=0xbfffe798) at t_msgbuilder.c:410 #7 0x402992a1 in build_local_ack (rpl=0x80d18c8, trans=0x10, branch=16, ret_len=0x10, next_hop=0x10) at t_reply.c:274 #8 0x40298621 in reply_received (p_msg=0x80d18c8) at t_reply.c:1217 #9 0x080569a0 in forward_reply (msg=0x80d18c8) at forward.c:504 #10 0x0806bea1 in receive_msg ( buf=0x80b9700 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 4.71.12.219:5068 ;branch=z9hG4bKe921.67e8dc61.0\r\nFrom: <sip:+xxxxxx@xx.xx.xx.xxx:xxxx>;tag=5df86c24f8e00fb2ae17ab041c094b88-7a49\r\nTo: <sip:+xxxxx@xx.xx.xx.xxx:"..., len=592, rcv_info=0x80d18c8) at receive.c:201 #11 0x080795ef in udp_rcv_loop () at udp_server.c:370 #12 0x0805aa89 in main_loop () at main.c:1107 #13 0x0805ce8e in main (argc=1108556168, argv=0xbfffed14) at main.c:2130 #14 0x42015574 in __libc_start_main () from /lib/tls/libc.so.6 (gdb) f 0 #0 qm_malloc (qm=0x80cea20, size=16) at mem/q_malloc.c:239 239 FRAG_END(next)->prev_free=prev; (gdb) list 239 234 235 prev=FRAG_END(frag)->prev_free; 236 next=frag->u.nxt_free; 237 prev->u.nxt_free=next; 238 if(prev) 239 FRAG_END(next)->prev_free=prev; 240 } 241 242 243 #ifdef DBG_QM_MALLOC (gdb) Looking forward for the reply Thanks Suresh

1 0

[Serusers] forwarded calls got timeout when called part hangup
by ashou han 10 Aug '06

10 Aug '06

Hi there, I use a t_on_failure("1") in route[1] for a unavailable server to another sip server, but always get timeout (code=408) when the called part hangup, no error (code=200) when calling part hangup. Please help!! here is my ser.cfg and log: ============================== # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=yes # (cmd line: -E) listen=127.0.0.1 /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "../usr/local/lib/ser/modules/mysql.so" #loadmodule "/usr/local/lib/ser/modules/postgres.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("acc", "log_missed_flag", 2) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "log_fmt", "miocfsdrtup") #modparam("acc", "db_flag", 3) #modparam("acc", "db_missed_flag", 4) #modparam("acc", "failed_transactions", 1) #modparam("acc", "db_url", "postgres://gatekeeper:mm1ch0t@localhost/ser") #modparam("acc", "log_extra" ,"ua=hdr/User-Agent") #modparam("acc", "db_extra" ,"uuid=avp/i:888; src_ip=avp/i:999") modparam("avpops", "avp_aliases", "c_uri=i:888; s_ip=i:999" ) # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # — tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE" || method=="CANCEL"){ ## Accounting for all calls setflag(1); # setflag(2); }; if (method=="INVITE") { record_route(); } # write the source IP as avp "s_ip" for accounting purposes # avp_write( "$src_ip", "$s_ip"); if (loose_route() ) { log("------- LOOSE ROUTE"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); log("------- IF ROUTE"); #route(1); break; } else { # write the current RURI as avp "c_uri" for accounting purposes #avp_write( "$ruri", "$c_uri"); append_hf("P-hint: outbound alias\r\n"); log("------- ELSE ROUTE"); route(1); break; }; append_hf("P-hint: usrloc applied\r\n"); log("------- LAST ROUTE"); #route(1); setflag(4); break; } route[1] { if ((src_ip==192.168.0.199)) { if ((uri=~"^sip:1617.*@127\.0\.0\.1") or (uri=~"^sip:1508.*@127\.0\.0\.1")) { log(1, "----------- Forwarding to OUR GW\n"); rewritehost("192.168.0.201"); log("----------- Host rewrited\n"); setflag(1); # if we do not get a positive reply, continue at reply_route[1] t_on_failure("1"); }else if ((uri=~"^sip:1.*@127\.0\.0\.1")) { log("----------- Forwarding to GW\n"); rewritehost("192.168.0.100"); log("----------- Host rewrited\n"); setflag(1); }; t_relay(); break; } } failure_route[1] { if (t_check_status("487")) { break; }; # forwarding failed -- try again at another destination if (method=="INVITE" && t_check_status("403|408|500|503")) { # revert_uri(); log("----------- Next gateway\n"); rewritehost("192.168.0.100"); log("----------- Host rewrited\n"); append_branch(); setflag(1); t_relay(); } } ========================================= log: Jul 31 15:17:38 localhost /usr/local/sbin/ser[3468]: ACC: transaction answered: method=INVITE, i-uri=sip:16175552222@127.0.0.1:5060, o-uri=sip:16175552222@192.168.0.201:5060, call_id=C3ECE80-200011DB-9498E149-4F3DC2BB(a)199.232.41.199, from=<sip:192.168.0.199>;tag=318DE490-1872, code=200, totag=SDa3kv999-78f0001f-0-2058902615, fromtag=318DE490-1872, to=<sip:16175552222@127.0.0.1>;tag=SDa3kv999-78f0001f-0-2058902615, uid=n/a, userpart=16175552222 Jul 31 15:17:38 localhost /usr/local/sbin/ser[3463]: ACC: request acknowledged: method=ACK, i-uri=sip:127.0.0.1:5060;ftag=318DE490-1872;lr=on, o-uri=sip:16175552222@192.168.0.100:5060;endpoint=127.2.0.43;transport=udp, call_id=C3ECE80-200011DB-9498E149-4F3DC2BB(a)192.168.0.199, from=<sip:192.168.0.199>;tag=318DE490-1872, code=200, totag=SDa3kv999-78f0001f-0-2058902615, fromtag=318DE490-1872, to=<sip:16175552222@127.0.0.1>;tag=SDa3kv999-78f0001f-0-2058902615, uid=n/a, userpart=n/a Jul 31 15:18:13 localhost /usr/local/sbin/ser[3471]: ACC: transaction answered: method=BYE, i-uri=sip:127.0.0.1:5060;ftag=318DE490-1872;lr=on, o-uri=sip:16175552222@192.168.0.100:5060;endpoint=127.2.0.43;transport=udp, call_id=C3ECE80-200011DB-9498E149-4F3DC2BB(a)192.168.0.199, from=<sip:192.168.0.199>;tag=318DE490-1872, code=408, totag=SDa3kv999-78f0001f-0-2058902615, fromtag=318DE490-1872, to=<sip:16175552222@127.0.0.1>;tag=SDa3kv999-78f0001f-0-2058902615, uid=n/a, userpart=n/a

2 1

[Serusers] SER avpops avp_pushto
by Shaun Hofer 10 Aug '06

10 Aug '06

When I try to use avp_pushto() in one section of ser.cfg it works and in another section it doesn't. When I use it to out right forward a call it works fine (callfwd). When it's used in the failure_route[1] it doesn't seem to push a value onto ruri (I have made sure that avp_db_load() has been called/used) . I'm wondering if avp_pushto() one of those methods that can't be called from a failed route ? I get the following error when I use it: ERROR: t_forward_nonack: no branched for forwarding ERROR: w_t_relay (failure mode): forwarding failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) I tried taking away voicemail section in failure_route[1] and made sure I was phoning extension with a number instead of 'voicemail' in value column of the database. That made no difference. When I tried adding append_branch() after avp_pushto() it just ends up ringing the original phone number again. Thanks in advance -Shaun route[1] { # ---------------------------------------------------------- # Default Message Handler # ---------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[3] { # ---------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { if (uri=~"^sip:0[1-9][0-9]{8}") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; log(1,"t on fail \n"); t_on_failure("1"); route(4); route(1); } route[4] { # ---------------------------------------------------------- # NAT Traversal Section # ---------------------------------------------------------- if (isflagset(6) || isflagset(7) || isflagset(11)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[6] { # ---------------------------------------------------------- # Call Forwarding Handler # ---------------------------------------------------------- lookup("aliases"); if (uri!=myself) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:88[0-9]{8}") { # Offline VoIP Network user sent to * for voicemail route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("408")) { if (avp_check("s:fwdnoanswer","eq/voicemail/I")) { avp_delete("s:fwdnoanswer"); resetflag(26); revert_uri(); rewritehostport("x.x.x.x:5060");#goto voicemail server append_branch(); t_relay_to_udp("x.x.x.x", "5060"); break; } else { avp_pushto("$ruri", "s:fwdnoanswer"); avp_delete("s:fwdnoanswer"); resetflag(26); route(6); break; }; }; end_media_session(); }

2 1

[Serusers] Problem with fr_inv_timer_avp
by Michel Zenone 10 Aug '06

10 Aug '06

Hi! Does somebody know why 'fr_inv_timer_avp' does nothing when I use it as follow: ... modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("avpops","avp_url","mysql://root@localhost/ser") modparam("avpops","avp_table","avptimer") # 'avptimer' is like 'usr_preferences' ... avp_db_load("$ruri","inv_timeout/avptimer"); Actually I try to set a fr_inv_timer different from an user to another. 'avptimer' timer defines, for each users, their proper fr_inv_timer. If somebody can help... Thanks in advance, Michel

2 1

[Serusers] xmlrpc
by Victoria Cortez 10 Aug '06

10 Aug '06

Hello everybody, Can anybody tell me about xmlrpc module? I've found it in the CVS version. I could not find any docs about it yet. What is this module suppose to do, exactly? Thanks in advance, Victoria

2 1

[Serusers] Looking for Call termination service provider
by John Kim 10 Aug '06

10 Aug '06

Hi, I am looking for a VoIP wholesaler who will allow us to operate our own SIP server infrastructure while we can forward all PSTN destined calls to their gateway. I really didn't think that it would be hard to find one but I have not been successful. I have checked so far with iCallGlobe.com and Deltathree.com. Can any of you recommend us some VoIP wholesaler with quality service and competitive rate who allows this business setup? Below is logical business relation setup: Our subscriber <--->ATA setup to connect sip.ours.com<----->PSTN gateway of wholesaler.com<--->PSTN Basically we are looking for good "wholesaler.com". Any comments will be appreciated. John K.

1 0

Re: [Serusers] SS7 Signaling & Gateways
by Max Clark 09 Aug '06

09 Aug '06

Thanks Victor - unfortunately Cirpack does not support North American networks. On 8/9/06, Victor Stanescu <victor.stanescu(a)gtstelecom.ro> wrote: > I don't know what is reasonable for you, but I really advise you to take > a look at Cirpack (www.cirpack.com). It's the full option gateway/switch > (isdn, r2, ss7, v5, mgcp, sip, h.323, h.248), and if you take time to > study their hardware, you can start from very small and pay as you grow. > > I am using Cirpack's for 3 years and I was never more happier about > other telco equipment :-) > > Max Clark wrote: > > Hi all, > > > > I am looking for a _resonable_ cost SS7 solution to bundle with a TDM > > <-> SIP media gateway. I know the Cisco 5350 supports SS7 when bundled > > with an external solution - is there anything reasonably priced that > > will give me SS7 capability? > > > > Thanks in advance, > > Max > > > -- Max Clark http://www.clarksys.com

1 0

[Serusers] Trying to use a remote mysql....
by Roger Lewau 09 Aug '06

09 Aug '06

I have been running SER quite successfully for a long time now. So tonight I dared to replicate the database to a remote server and changed the ser.cfg as follows. This is the two lines I changed in my ser.cfg filen #fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_db_url="mysql://ser:heslo@192.168.12.140/ser" #modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@192.168.121.140/ser") Also updated serctl as follows: SQL_HOST=192.168.121.140 I then Stopped Ser, MySQL and restarted SER. It takes a long time then I get these line in the error log.. and ser fails to start. Aug 9 22:36:08 ns2 ser[43517]: new_connection(): Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) Aug 9 22:36:08 ns2 ser[43517]: db_init(): Could not create a connection Aug 9 22:36:08 ns2 ser[43517]: ERROR: group_db_ver: unable to open database connection Aug 9 22:36:08 ns2 ser[43517]: group:mod_init(): Error while querying table version Aug 9 22:36:08 ns2 ser[43517]: init_mod(): Error while initializing module group Then I start the old local mySQL server again and restart ser and it springs to life. Why is that? I am not pointing to the local mysql server anymore... Have I missed something in the configuration? Even more strange, it seems that all authentication and accounting gets queried in the new server as I see can see the data in the new server alse sees the client connection from SER. Any one who can help with an explanation on this behaviour? Kind regards Roger

2 2

[Users] I can`t add new client by using openserctl add. Please help...
by Ferianto siregar 09 Aug '06

09 Aug '06

Dear all, Now, I try to test my openser configuration by using xlite. I tried to call from client to test the openser configuration that I have built working sucessfully. So, I must create a new user by using "openserctl add" command. But when i tried to do that, I got an error message. The error message said that " no rows effected". I think that this message said that I can not add a new user. For example: #openserctl add aldi aldi aldi(a)pcr.ac.id What can i do for solve this problem? I do hope anybody can give me a sugesstion.Please help me...Please Note: I use mysql for the database, and it can work goodly. ( I can show all the table that there is in openser database) Thanks with cheers, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

2 1

[Users] Q:ABOUT:DELAY THE INVITE
by Alejandro Sanchez 09 Aug '06

09 Aug '06

MY scenario is the next. I have 2 video telephones that are conected to the PSTN and they has access trougth a dial up conection to the IP network and the problem is that the 2 video telephones shuld be register before one to other geretate the INVITE then i need make that when the first REGISTER arrive the sip server generate a delay for wait for the REGISTER of the other video telephone and both VT state register for the INVITE message.. Someone knows how can i do this? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Serusers] Call limiting ?
by Alexandre Snarskii 09 Aug '06

09 Aug '06

Hi! As far as I know, SER does not have functionality of limiting number of calls on per-user basis. It was discussed years ago, and the reason for that was "We don't have ability to know is call still alive" (http://lists.iptel.org/pipermail/serusers/2004-November/013035.html) But, in April 2005, RFC 4028 (Session Timers in SIP) appeared, and it (at least theoretically) can be utilised to achieve knowlege of active dialogs, and, as result - on number of active calls. And this RFC looks like supported on Cisco gateways: U X.X.X.X:59104 -> X.X.X.X:5060 INVITE sip:aaaaaaa@X.X.X.X:5060 SIP/2.0. Supported: 100rel,timer,replaces. ^^^^^ Min-SE: 1800. ^^^^^^^^^^^^^^ User-Agent: Cisco-SIPGateway/IOS-12.x. Question: is this RFC/functionality planned to be supported in SER ? If not - are there any good intro on "How to write SER module by yourself" ? :)

4 5

[Serusers] SS7 Signaling & Gateways
by Max Clark 09 Aug '06

09 Aug '06

Hi all, I am looking for a _resonable_ cost SS7 solution to bundle with a TDM <-> SIP media gateway. I know the Cisco 5350 supports SS7 when bundled with an external solution - is there anything reasonably priced that will give me SS7 capability? Thanks in advance, Max -- Max Clark http://www.clarksys.com

1 0

[Serusers] presence_b2b UA
by Curt Moore 09 Aug '06

09 Aug '06

Hello all. I have been experimenting with the ser-0.10.99-xxx branch(es) and have a question about the presence_b2b module. I realize that the module is not yet fully complete but according to the documentation, when finished, the module should be able to take an inbound SUBSCRIBE request and then create a new SUBSCRIBE request outbound to another SIP entity or multiple SIP entities, ie Asterisk servers, hence the back to back functionality. What is the status of this functionality? Is anyone currently working to complete this functionality? I would be interested in helping complete or sponsoring the completion of this functionality. My intent is to have a SER server act as the central watcher database server and have all SUBSCRIBE and NOTIFY requests channeled through this SER server. This SER server will then generate new SUBSCRIBE requests out to each server in a farm of Asterisk servers and the Asterisk servers will then send NOTIFIES back to the SER watcher server. The watcher server will then take care of sending the NOTIFY messages to the correct watcher(s), according to the data in its watcher database. Is anyone currently doing anything similar to this using a different method or is using the yet to be completed B2B UA the correct/only way to accomplish this? Thanks, -Curt

2 1

[Users] Serial forking and accounting
by Andreas Granig 09 Aug '06

09 Aug '06

Hi, In a scenario A -> OpenSER -> B with the request being INVITE, B sends back 404 to OpenSER (the flag "db_missed_flag" of the acc module is set). OpenSER forks serially and sends another request to C, which is answered with 200. Now I get two acc records, the 404 from B because of the missed-flag and the 200 from C. Now is there a way to selectively suppress the acc record for the 404 in the failure route? Concretely I'd like to decide in the failure route whether that transaction is being accounted. Thanks, Andy

2 7

[Users] Troubles parsing headers
by Lars Sundqvist 09 Aug '06

09 Aug '06

Hi everyone, We have been having a problem with our openser installation for a while, certain REGISTER requests generates errors. The openser version we are running is 1.0.0 from cvs. Does anyone have a clue why the following register generates the error below? Register: xxx.xxx.xx.xx:5061 -> yyy.yyy.yyy.yyy:5060 REGISTER sip:sip.xxxxx.xxx SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xx.xx:5061;branch=z9hG4bK-39f541a2..From: aaaa.bbbbb <sip: username123(a)sip.xxxxx.xxx>;tag=ea 8cef4b108a99bco1..To: aaaa.bbbbb <sip: username123(a)sip.xxxxx.xxx>..Call-ID: b96fead3-f03493d4@xxx.xxx.xx.xx..CSeq: 2508 REGISTER..Max-Forwards: 70..Conta ct: aaaa.bbbbb <sip: username123@xxx.xxx.xx.xx:5061>;expires=3600..User-Agent: Linksys/RT31P2-2.0.10(LIc)..Content-Length: 0..Allow: ACK, BYE, CANCEL, IN FO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura.... Error: /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR:parse_from_header: bad from header /sbin/openser[18198]: xl_get_from: ERROR cannot parse FROM header /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: xl_get_to: ERROR cannot parse TO header /sbin/openser[18198]: [REGISTER] from [<null>] to [<null>] /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! /sbin/openser[18198]: ERROR:parse_from_header: bad from header /sbin/openser[18198]: insert_RR(): From parsing failed /sbin/openser[18198]: record_route(): Error while inserting Record-Route line /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: find_first_route: Error while parsing headers /sbin/openser[18198]: analyzing REGISTER request /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: find_credentials(): Error while parsing headers /sbin/openser[18198]: pre_auth(): Error while looking for credentials /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR: get_hdr_field: bad to header /sbin/openser[18198]: ERROR: bad header field /sbin/openser[18198]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed /sbin/openser[18198]: pre_auth(): Error while sending 400 reply Many thanks, Lars Sundqvist

1 0

[Serusers] Gizmo Project API
by Voipers Portugal 09 Aug '06

09 Aug '06

Hi, Does anyone here used Gizmo Project API to build a softphone that can be used with other SIP servers like SER? My idea was to develop something like their softphone (Gizmo) but make it work with my own VoIP network. Any suggestions? Jose Simoes

1 0

RE: [Serusers] Call Forward Billing
by Rosa De Santis 09 Aug '06

09 Aug '06

Hi again, Jiri. I had to leave this for a while, but I'm back. I'm trying to do the original idea: A calls B, B forwards to C. The call is A-C and the billing is B-C. In SER I added B in the Diversion header of the SIP message and this is forwarded to *b2bua. In *b2bua I change the Radius Message to charge B and change the SIP message to make the call from A to C, just like you told me. Everything seems to be OK: C phone rings, but....the call can't be established, the call hangs up when C answers. It seems that something is missing after the call is answered... Can you please give me some help? Best regards Rosa. ---------------------------------------- > Date: Tue, 11 Jul 2006 16:05:59 +0200 > To: rosadesantis(a)hotmail.com; serusers(a)lists.iptel.org > From: jiri(a)iptel.org > Subject: RE: Re: [Serusers] Call Forward Billing > > At 15:45 11/07/2006, Rosa De Santis wrote: > >Thanks Jiri and Greger for your answers. > > > >I just wanted to make it simple, because I'm using a third party billing system. > >I'm using SER + Asterisk b2bua. My original idea was to add a Diversion header to the sip message in SER, and to modify the Radius message in Asterisk > b2bua when i find that diversion header, to give the Radius message ready and modified to the billing system, so they don't have to make any additional > process for the call forward case. > > If that's all you need then it is as easy as putting the proper identity in AVP, printing the AVP > in diversion header, and integrating all pieces together. > > -jiri > > > >Please, is this possible? any ideas? > >Thanks again. > > > >Rosa. > > > >----------------------------------------------- > >> Date: Tue, 11 Jul 2006 08:18:59 +0200 > >> From: greger(a)teigre.com > >> To: jiri(a)iptel.org > >> CC: rosadesantis(a)hotmail.com; serusers(a)lists.iptel.org > >> Subject: Re: [Serusers] Call Forward Billing > >> > >> Just to expand on "special-casing": Getting user actions into $ can be very tricky and is very dependent on your billing system. I.e. the capabilities of the type of invoice records that can be created (and manipulated; rating, VAT, no VAT, customer discounts, volume discounts, time of day etc etc). You normally have to start there and trace back: What should the normalized CDRs look like for the billing system to import and rate them correctly. Then you step backwards again: Your accounting records (start/stop), what should they look like. The tool taking start/stop records must be capable of interpreting the records and create normalized CDRs. If you gather accounting records from multiple sources (PSTN GWs, routers etc), it may make sense to consolidate the record collection and normalization (i.e. adapt your accounting records to be similar across sources so you simplify the normalization)- > >> Puh! This leads us to Jiri's comment: Once you know what you need, you can use additional AVPs/flags in acc table. > >> g-) > >> PS! In the PSTN world, the forwarding party/C-number is often added in a third column in the records and one set of start/stop will exist for each call leg. If present, the C-number will be used for charging. > >> Jiri Kuthan wrote: > >> It is a configuration thing. It appears a misconception to me to build special-casing in SER. acc module can store additional AVPs or flags in acc table. These can indicate reverse charging at your convenience, to be processed by your favorite CDR postprocesssing tool. Moreover, if done well, all the authentication-authorization-accounting process can be, if configured properly, brought to the same least common denominator, which is AVP -- you authenticate someone, check if his ACLs expressed as an AVP permit a call (e.g., 'international'), account it (possibly with additional AVPs/flags providing additional hints for how to interpret the CDR). -jiri At 19:49 10/07/2006, Rosa De Santis wrote: > >> Thanks for your answer. And this functionality is not present in SER ? Rosa. ----------------------------------------------- > >> Date: Sat, 8 Jul 2006 00:20:44 -0700 From: t_maron_lists(a)yahoo.com Subject: Re: [Serusers] Call Forward Billing To: rosadesantis(a)hotmail.com If you are locked by this, you can import a patch from openser, they have it for some time, see: http://openser.org/docs/modules/1.1.x/acc.html#MULTI-CALL-LEGS T. On 07/07/06 18:46, Rosa De Santis wrote: Hello all. Please, can somebody help about changing the billing when making call forward ? I'm trying to change the billed user when A calls B, B forwards to C (C is PSTN). I want to charge the call to B. I'm using Ser+Asterisk B2BUA. Any idea, please? Thanks Rosa. _________________________________________________________________ Be one of the first to try Windows Live Mail. http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49… Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ----------------------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. > >> _________________________________________________________________ Be one of the first to try Windows Live Mail. http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49… Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > >> -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > > >_________________________________________________________________ > >Be one of the first to try Windows Live Mail. > >http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-49… > >Serusers mailing list > >Serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > _________________________________________________________________ Descubre Live.com - tu mundo en línea reunido: noticias, deportes, el tiempo, y mucho más. http://www.live.com/getstarted

2 1

[Users] Problem to compile Module unixodbc
by Jose Gil Navarrete 08 Aug '06

08 Aug '06

Hi!!!!! I have the next problem: When I try to compile, I obtain the next error: Where and How I can obtain these files (sql.h, sqlext.h) that I need to compile this module? Thanks!!!!! make[1]: Entering directory `/root/OPENSER/openser-1.1.0-notls/modules/unixodbc' Compiling dbase.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.0"' -D__CPU_i386 -D__OS_linux -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c dbase.c -o dbase.o In file included from dbase.c:44: val.h:36:17: error: sql.h: No such file or directory val.h:37:20: error: sqlext.h: No such file or directory In file included from dbase.c:44: val.h:50: error: expected â)â before â*â token In file included from dbase.c:45: my_con.h:57: error: expected specifier-qualifier-list before âSQLHENVâ my_con.h:87: error: expected declaration specifiers or â...â before âSQLHANDLEâ my_con.h:87: error: expected declaration specifiers or â...â before âSQLSMALLINTâ dbase.c: In function âreconnectâ: dbase.c:60: error: âSQLCHARâ undeclared (first use in this function) dbase.c:60: error: (Each undeclared identifier is reported only once dbase.c:60: error: for each function it appears in.) dbase.c:60: error: expected â;â before âoutstrâ dbase.c:61: error: âSQLSMALLINTâ undeclared (first use in this function) dbase.c:61: error: expected â;â before âoutstrlenâ dbase.c:67: warning: implicit declaration of function âSQLDisconnectâ dbase.c:67: error: âstruct my_conâ has no member named âdbcâ dbase.c:76: warning: implicit declaration of function âSQLDriverConnectâ dbase.c:76: error: âstruct my_conâ has no member named âdbcâ dbase.c:77: error: expected expression before â)â token dbase.c:79: warning: implicit declaration of function âSQL_SUCCEEDEDâ dbase.c:81: error: âstruct my_conâ has no member named âdbcâ dbase.c:82: error: âSQL_HANDLE_DBCâ undeclared (first use in this function) dbase.c:82: error: too many arguments to function âextract_errorâ dbase.c:86: warning: implicit declaration of function âSQLAllocHandleâ dbase.c:86: error: âSQL_HANDLE_STMTâ undeclared (first use in this function) dbase.c:86: error: âstruct my_conâ has no member named âdbcâ dbase.c:87: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:89: error: âstruct my_conâ has no member named âdbcâ dbase.c:91: error: âstruct my_conâ has no member named âdbcâ dbase.c:91: error: too many arguments to function âextract_errorâ dbase.c: In function âsubmit_queryâ: dbase.c:104: error: âSQLCHARâ undeclared (first use in this function) dbase.c:104: error: expected â;â before âsqlstateâ dbase.c:107: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:109: warning: implicit declaration of function âSQLCloseCursorâ dbase.c:109: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:110: warning: implicit declaration of function âSQLFreeHandleâ dbase.c:110: error: âSQL_HANDLE_STMTâ undeclared (first use in this function) dbase.c:110: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:113: error: âstruct my_conâ has no member named âdbcâ dbase.c:113: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:116: error: âstruct my_conâ has no member named âdbcâ dbase.c:118: error: âstruct my_conâ has no member named âdbcâ dbase.c:118: error: âSQL_HANDLE_DBCâ undeclared (first use in this function) dbase.c:119: error: âsqlstateâ undeclared (first use in this function) dbase.c:119: error: too many arguments to function âextract_errorâ dbase.c:131: warning: implicit declaration of function âSQLExecDirectâ dbase.c:131: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:131: error: expected expression before â)â token dbase.c:134: error: expected â;â before âsqlstateâ dbase.c:136: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:137: error: too many arguments to function âextract_errorâ dbase.c:147: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:147: error: expected expression before â)â token dbase.c:151: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:152: error: too many arguments to function âextract_errorâ dbase.c:154: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:155: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:162: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:163: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c: At top level: dbase.c:209: error: expected â)â before â*â token dbase.c:240: error: expected â)â before â*â token dbase.c:286: error: expected â)â before â*â token dbase.c: In function âdb_free_resultâ: dbase.c:451: error: âSQL_HANDLE_STMTâ undeclared (first use in this function) dbase.c:451: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c:452: error: âstruct my_conâ has no member named âstmt_handleâ dbase.c: In function âdb_queryâ: dbase.c:505: warning: implicit declaration of function âprint_whereâ dbase.c:505: error: âstruct my_conâ has no member named âdbcâ dbase.c: In function âdb_insertâ: dbase.c:581: warning: implicit declaration of function âprint_valuesâ dbase.c:581: error: âstruct my_conâ has no member named âdbcâ dbase.c: In function âdb_deleteâ: dbase.c:628: error: âstruct my_conâ has no member named âdbcâ dbase.c: In function âdb_updateâ: dbase.c:672: warning: implicit declaration of function âprint_setâ dbase.c:672: error: âstruct my_conâ has no member named âdbcâ dbase.c:683: error: âstruct my_conâ has no member named âdbcâ dbase.c: In function âdb_replaceâ: dbase.c:727: error: âstruct my_conâ has no member named âdbcâ make[1]: *** [dbase.o] Error 1 make[1]: Leaving directory `/root/OPENSER/openser-1.1.0-notls/modules/unixodbc' --------------------------------- Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Serusers] CIDName to Alias
by Ali Zaidi 08 Aug '06

08 Aug '06

Hi, Is this possible to change CIDName to Alias, for example my uri is 5001231234(a)myser.com and alias 2121231234. When person call to other user, it shows "Jon Doe" 500-123-1234 but i want this to shown as "Jon Doe" 2121231234 <= which is publicly known PSTN number. Please help.... Regards, Ali....

1 0

[Serusers] CIDName to Alias
by Ali Zaidi 08 Aug '06

08 Aug '06

Hi, Is this possible to change CIDName to Alias, for example my uri is 5001231234(a)myser.com and alias 2121231234. When person call to other user, it shows "Jon Doe" 500-123-1234 but i want this to shown as "Jon Doe" 2121231234 <= which is publicly known PSTN number. Please help.... Regards, Ali....

1 0

[Users] mysql problems with openser
by Ever Zalazar 08 Aug '06

08 Aug '06

Hi ! I have a question: When I try to create a user openserctl add 202020 12345 202020(a)hotmail.com Appear a message: Warning: no database engine found - tried ' ' This command requires a database engine - none was loaded. Thanks Ever

2 1

[Users] reading info from custom db table
by zm23 08 Aug '06

08 Aug '06

Hi, I want to store some information in a separate table in mysql database. Think of it as a second user preference table but I want to keep that information separate. What will be the best way to achieve that? So that I am able to lookup values from this additional table and use it to make routing decisions. Also, does openser provide a way to do custom sql queries. Thanks in advance for your help. -- Zahid Mehmood CUIT Network Systems

2 2

[Serusers] SER on best OS
by ram 08 Aug '06

08 Aug '06

Hi all What is the best OS to install SER on. and its very stable for Prepaid and Post paid Iam trying to install on FC5 or RHEL 4.0 Expert coments will be apprciated Ram

3 3

Re: [Users] Can not register to OPENSER
by Kristian Kielhofner 08 Aug '06

08 Aug '06

Bill Zhang wrote: > Bogdan, > > Thanks for the reminder, this is a question that I don't know the answer, as > I'm struggling to understand the whole configuration. I will try to find > out why it is called:). > > BTW, I always want to see if I can make SER a powerful server with following > configuration: > > 1. Check if SIP end point(base on the SIP message, like Register, Invite) > is behind the NAT, if it is with private IP, then use SER NAThelper for SIP > Message and RTP Proxy for RTP. > 2. If it is with public IP, check if its IP and Source port is as claimed > in the SIP Message(Invite, Register, etc), if yes, then proceed to treat > this SIP end point with public IP. > 3. If 2 is not true, then this end point may use STUN with wrong detection > result, then we treat it as in 1. > > I wonder if there is any sample configurations for reference? > > Best Regards, > Bill Zhang > Tel: 626-956-0260 ext 601 > Cell: 626-354-0967 > web: www.grandstream.com > email: wzhang(a)grandstream.com > > Bill, nathelper.so has all of that functionality: http://www.openser.org/docs/modules/1.1.x/nathelper.html 1) nat_uac_test & fix_nated_* can do everything you are looking for. Make sure to look at the flags for nat_uac_test. They should probably be >= 19. 2) nat_uac_test only returns true if it detects nat based on your flags. Therefore, you can use logic like this: #--------REGISTER section---------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("23")) { setflag(6); fix_nated_register(); force_rport(); }; #-----INVITE SECTION-------- #(in INVITE message handler) if (nat_uac_test("23")) { setflag(6); } #--------Extra Routes-------------- route[4] { # NAT Traversal Section if (isflagset(6)) { xlog("Fixing NAT in section 4\n"); force_rport(); fix_nated_contact(); force_rtp_proxy(); } } Now you can just call route(4) BEFORE you call your other route(s). 3) I'm not sure that I really follow you on this one... P.S. - You'll also want to run RTPProxy to proxy the media streams for clients that are behind nats. -- Kristian Kielhofner

1 0

[Users] Open SER on Best OS
by ram 08 Aug '06

08 Aug '06

Hi all What is the best OS to install Open SER on. and its very stable for Prepaid and Post paid Iam trying to install on FC5 or RHEL 4.0 Expert coments will be apprciated Ram

6 6

[Serusers] SER & RTPProxy
by Sebastian Gabris 08 Aug '06

08 Aug '06

Hello everybody, does anybody have a sample config - file for this scenario : (no authentication, mysql or whatever needed. SER should just relay the call) UA(privateIP) <-> (privateIP)SER(publicIP) <-> UA-(VoIP - Provider, pubicIP) I already tried the onsip configfiles and many other files :) but I still see no way to get this SER - Router working. I already tested this enviroment with siproxd and all worked well. So the problem must be with my config file. Any help would be very appreciated :-) ______________________________________________________________________ XXL-Speicher, PC-Virenschutz, Spartarife & mehr: Nur im WEB.DE Club! Jetzt gratis testen! http://freemail.web.de/home/landingpad/?mc=021130

1 0

[Users] Proxy authentication message to wrong port
by M D 08 Aug '06

08 Aug '06

Hi I have an OpenSER 1.1 box on a public IP running a config taken more-or-less verbatim from the iptel.org getting started examples. I have a UA behind a PIX which is translating port 5060 on the phone to port 8907 on the firewall. OpenSER is ignoring this and sending replies to INVITEs to port 5060 on the firewall. If it's likely to make any difference, the PATed IP and the IP of the OpenSER box are on the same network. 31 61.574505 193.x.x.15 -> 193.x.x.5 SIP/SDP Request: INVITE sip:5551212@193.x.x.5;user=phone, with session description 32 61.575998 193.x.x.5 -> 193.x.x.15 SIP Status: 407 Proxy Authentication Required Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: SIP Request: Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: method: <INVITE> Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: uri: < sip:5551212@193.x.x.5;user=phone> Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: version: <SIP/2.0> Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: flags=2 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: Found param type 232, <branch> = <z9hG4bK4ae31c203ab6ceb>; state=16 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: end of header reached, state=5 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: Via found, flags=2 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: this is the first via Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: After parse_msg... Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: preparing to run routing scripts... Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: flags=100 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG:parse_to:end of header reached, state=10 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DBUG:parse_to: display={}, ruri={sip:5551212@193.x.x.5;user=phone} Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG: get_hdr_field: <To> [39]; uri=[ sip:5551212@193.x.x.5;user=phone] Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG: to body [< sip:5551212@193.x.x.5;user=phone>^M ] Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG: get_hdr_body : content_length=284 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: found end of header Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG: is_maxfwd_present: max_forwards header not found! Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG: add_param: tag=3783260355 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG:parse_to:end of header reached, state=29 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DBUG:parse_to: display={}, ruri={sip:84410001@193.x.x.5;user=phone} Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: flags=200 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: find_first_route: No Route headers found Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: loose_route: There is no Route HF Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: grep_sock_info - checking if host==us: 12==12 && [ 193.x.x.5] == [193.x.x.5] Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: grep_sock_info - checking if port 5060 matches port 5060 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: XXX INVITE handler: start Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: flags=10000 Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: pre_auth(): Credentials with given realm not found Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: XXX INVITE handler: proxy_authorize failed Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: build_auth_hf(): 'Proxy-Authenticate: Digest realm=" 193.x.x.5", nonce="44d3636e40c00e3f51456a587f994d0f285325af"^M ' Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: parse_headers: flags=ffffffffffffffff Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: check_via_address( 193.x.x.15, 10.200.100.46, 0) Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: DEBUG:destroy_avp_list: destroying list (nil) Aug 4 16:05:38 sip3 /usr/sbin/openser[22195]: receive_msg: cleaning up How can I force proxy_challenge() to send its challenge to port 8907? Cheers, Mark Config: debug=8 fork=yes log_stderror=no listen=193.82.139.5 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openserro:openserro@localhost/openser" loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", " mysql://openserro:openserro@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { log("XXX default handler: start"); # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { log("XXX REGISTER handler: start"); # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { log("XXX REGISTER handler: valid contact and nat_uac_test(19) true"); setflag(6); fix_nated_register(); force_rport(); }; log("XXX REGISTER handler: 100 trying"); sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { log("XXX REGISTER handler: www_authorize failed"); www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; log("XXX REGISTER handler: location saved"); } route[3] { log("XXX INVITE handler: start"); # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { log("XXX INVITE handler: proxy_authorize failed"); proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { log("XXX NAT traversal: start"); # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } onreply_route[1] { log("XXX onreply_route: start"); if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { log("XXX onreply_route: nat_uac_test(1) true"); fix_nated_contact(); }; }

2 3

[Users] auth_radius
by Pavel D. Kuzin 08 Aug '06

08 Aug '06

trying to setup radius and set to work together. sems have a strange problem.... ser recives " 0(5447) ERROR:auth_radius:radius_authorize_sterman: rc_auth failed" but freeradius reply is "Sending Access-Accept of id 172 to 127.0.0.1 port 56137 Reply-Message = "Authenticated" Sip-RPId = "88123321139" Finished request 0 " Please tell me what is a problem? -- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk(a)nodex.ru http://nodex.ru

3 5

[Users] Need help on PHP and Radius
by raviprakash sunkara 08 Aug '06

08 Aug '06

Hi Guys... I need to integrate php and raadius... Actually i want to create the own user interface on OPENSER..... That on PHP 5 is having the radis integration . So i need it... Now , I'm using the Radius in openser for " AAA " purpose only... Now I need to develop web interface for my application with radius... plz help me.... on this............. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

3 2

Re: [Users] Can not register to OPENSER
by Bogdan-Andrei Iancu 08 Aug '06

08 Aug '06

Hi Bill, you should you call force_rtp_proxy() for a REGISTER? regards, bogdan Bill Zhang wrote: >The issue is, this is a SIP Registration message, it certainly does NOT >contain SDP, in fact there are other SIP messages that does NOT contain SDP >message but still need nathelper to recognize it behind a NAT so that >OPENSER can send response back to the client correctly. > >In summary, nathelper should be able to base on the SIP message to handle it >correctly. > >Best Regards, >Bill Zhang >-----Original Message----- >From: Marc Haisenko [mailto:haisenko@comdasys.com] >Sent: Thursday, August 03, 2006 12:09 AM >To: users(a)openser.org >Cc: Bill Zhang >Subject: Re: [Users] Can not register to OPENSER > >On Thursday 03 August 2006 00:46, Bill Zhang wrote: > > >>Anyone knows why I keep getting following error message? >> >> >> >>Aug 2 13:10:26 localhost openser[30695]: ERROR: extract_body: message >> >> >body > > >>has length zero >> >>Aug 2 13:10:26 localhost openser[30695]: ERROR: force_rtp_proxy2: can't >>extract body from the message >> >> > >The message says it, and your SIP message proves it: the SIP message has no >body. > >nathelper operates on the SDP body, it mangles them (together with rtpproxy) > >so that the SDP body sent to the receiver contains the address of your >machine running OpenSER instead of the real phone. This is so that the >machine can act as RTP proxy on network edges. > >Since your SIP message has no SDP body there is nothing nathelper can do :-) >C'ya, > Marc > > > >

2 1

[Users] record route
by abhishek kaushik 07 Aug '06

07 Aug '06

Hi all, This is abhishek Kaushik from delhi , i am not using record route in my proxy , but still the ACk/Bye passes through my proxy ,can somebody please explain, thanks and regards abhishek kaushik ____________________________________________________________ FREE ONLINE PHOTOSHARING - Share your photos online with your friends and family! Visit http://www.inbox.com/photosharing to find out more!

2 1

[Users] The lcr module has incorrect syntax for the current mysql release
by Brandon Price 07 Aug '06

07 Aug '06

Of coarse I found this out after upgrading my test box from mysql 5.0.19 to 5.0.22. The lcr module runs the following query... SELECT gw.ip_addr, gw.port, gw.uri_scheme, gw.transport, gw.strip, gw.prefix FROM gw, lcr WHERE ''<sip:12345678910@eastern.mynoc.net>'' LIKE lcr.from_uri AND '15184782222' LIKE CONCAT(lcr.prefix, '%') AND lcr.grp_id = gw.grp_id ORDER BY CHAR_LENGTH(lcr.prefix), lcr.priority DESC, RAND() which returns ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near ':12345678910@eastern.mynoc.net>'' LIKE lcr.from_uri AND '15184782222' LIKE CONCA' at line 1 the correct query would be ... SELECT gw.ip_addr, gw.port, gw.uri_scheme, gw.transport, gw.strip, gw.prefix FROM gw, lcr WHERE '<sip:12345678910@eastern.mynoc.net>' LIKE lcr.from_uri AND '15184782222' LIKE CONCAT(lcr.prefix, '%') AND lcr.grp_id = gw.grp_id ORDER BY CHAR_LENGTH(lcr.prefix), lcr.priority DESC, RAND(); Notice that I removed the double single quotes around <sip: 12345678910(a)eastern.mynoc.net> which is the from uri. This appears to be broken in current cvs as well.

3 6

Re: [Serusers] Cisco like translation rules in SER.
by Jiri Kuthan 07 Aug '06

07 Aug '06

There is a buch of commands you can use for uri maniputlation: strip, prefix, subst_uri. -jiri At 15:52 25/07/2006, Robert Zorop wrote: >Hi guys. > I'd like to hear about different methods of implementing cisco like transtation rules in SER. I was thinking about AVPs, but still couldn't find the ight way. > Is someone is doing it inside or outside ser, please shed a light on this!. What i need is to stip some numbers, and then add some other (variables) before proxying the calls. > > >Thanks in advance. > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

3 4

Re: [Users] OpenSER Administrator Release
by Mike Williams 07 Aug '06

07 Aug '06

It's done on SVN. Thanks for the tip. Let me know what you think of the release. Mike On Monday 07 August 2006 02:53, Louis Alexander wrote: > Gem install mysql worked great! > > You might want to add this to your INSTALL document as it is not in there. > Thanks! > > Louis > > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > Behalf Of Mike Williams > Sent: Monday, August 07, 2006 2:40 PM > To: users(a)openser.org > Subject: Re: [Users] OpenSER Administrator Release > > Hmm, > > Did you run 'gem install mysql'? According to some websites, this is > needed. > > Also, what OS and version of MySQL are you running? > > Mike > > On Monday 07 August 2006 02:03, Louis Alexander wrote: > > Unfortunately (fortunately?) I've allowed that user under the machines > > domain name, 127.0.0.1, and localhost. Yet the issue still persists. > > Ill keep trying to figure it out. > > > > Thanks > > > > Louis > > > > -----Original Message----- > > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > > Behalf Of Mike Williams > > Sent: Monday, August 07, 2006 1:19 PM > > To: users(a)openser.org > > Subject: Re: [Users] OpenSER Administrator Release > > > > > > Google seems to suggest that it might be a permissions problem: remember > > that > > to MySQL, localhost and 127.0.0.1 are different. You may want to look at > > this > > link: http://www.adamsdesk.com/be/?p=177 > > > > Mike > > > > On Monday 07 August 2006 01:02, Louis Alexander wrote: > > > Mike, > > > I'm having trouble logging into the OpenSER Administrator. I get > > > the following error in my browser: > > > > > > Got any tips? > > > > > > > > > ActiveRecord::StatementInvalid in LoginController#login > > > > > > Mysql::Error: Lost connection to MySQL server during query: SELECT * > > FROM > > > > users WHERE (login = 'openser' AND password = > > > '0dd62bae0b6b4d8e0388c620134decb8405b6976') LIMIT 1 > > > > > > RAILS_ROOT: script/../config/.. > > > Application Trace | Framework Trace | Full Trace > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/abstract_adapter.rb:120:in `log' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:184:in `execute' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:336:in `select' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:175:in `select_all' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:390:in `find_by_sql' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:924:in `find_every' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:918:in `find_initial' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:380:in `find' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/dep > > > >r ecated_finders.rb:22:in `find_first' > > > #{RAILS_ROOT}/app/models/user.rb:7:in `authenticate' > > > app/controllers/login_controller.rb:11:in `login' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/abstract_adapter.rb:120:in `log' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:184:in `execute' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:336:in `select' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:175:in `select_all' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:390:in `find_by_sql' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:924:in `find_every' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:918:in `find_initial' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:380:in `find' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/dep > > > >r ecated_finders.rb:22:in `find_first' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >a se.rb:910:in `perform_action_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > > >i lters.rb:368:in `perform_action_without_benchmark' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > > /usr/local/lib/ruby/1.8/benchmark.rb:293:in `measure' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/r > > > >e scue.rb:82:in `perform_action' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >a se.rb:381:in `process_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > > >i lters.rb:377:in `process_without_session_management_support' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/s > > > >e ssion_management.rb:117:in `process' > > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/dispatcher.rb:38:in > > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:115:i > > > >n `handle_dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:81:in > > > > `service' > > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:104:in `service' > > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:65:in `run' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:173:in `start_thread' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start_thread' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:95:in `start' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `start' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:23:in `start' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:82:in `start' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:67:in > > > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/servers/webrick. > > > >r b:59 > > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > > >p endencies.rb:147:in `require' > > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/server.rb:30 > > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > > >p endencies.rb:147:in `require' > > > script/server:3 > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/abstract_adapter.rb:120:in `log' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:184:in `execute' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:336:in `select' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > > >n ection_adapters/mysql_adapter.rb:175:in `select_all' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:390:in `find_by_sql' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:924:in `find_every' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:918:in `find_initial' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > > >e .rb:380:in `find' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/dep > > > >r ecated_finders.rb:22:in `find_first' > > > #{RAILS_ROOT}/app/models/user.rb:7:in `authenticate' > > > app/controllers/login_controller.rb:11:in `login' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >a se.rb:910:in `perform_action_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > > >i lters.rb:368:in `perform_action_without_benchmark' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > > /usr/local/lib/ruby/1.8/benchmark.rb:293:in `measure' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/r > > > >e scue.rb:82:in `perform_action' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > > >a se.rb:381:in `process_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > > >i lters.rb:377:in `process_without_session_management_support' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/s > > > >e ssion_management.rb:117:in `process' > > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/dispatcher.rb:38:in > > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:115:i > > > >n `handle_dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:81:in > > > > `service' > > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:104:in `service' > > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:65:in `run' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:173:in `start_thread' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start_thread' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:95:in `start' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `start' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:23:in `start' > > > /usr/local/lib/ruby/1.8/webrick/server.rb:82:in `start' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:67:in > > > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/servers/webrick. > > > >r b:59 > > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > > >p endencies.rb:147:in `require' > > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/server.rb:30 > > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > > >p endencies.rb:147:in `require' > > > script/server:3 > > > > > > Request > > > > > > Parameters: {"user_login"=>"openser", "user_password"=>"openserrw", > > > "login"=>"Login \302\273"} > > > > > > Show session dump > > > > > > --- > > > return-to: /login > > > flash: !map:ActionController::Flash::FlashHash {} > > > > > > > > > Response > > > Headers: {"cookie"=>[], "Cache-Control"=>"no-cache"} > > > > > > -----Original Message----- > > > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > > > Behalf Of Mike Williams > > > Sent: Friday, August 04, 2006 10:38 AM > > > To: users(a)openser.org > > > Cc: devel(a)openser.org > > > Subject: [Users] OpenSER Administrator Release > > > > > > All, > > > > > > I've just finished my first release of OpenSER Administrator, a web > > > interface > > > for managing OpenSER. It's GPL'd and written in Ruby on Ralis, and can > > be > > > > downloaded from openseradmin.sourceforge.net > > > > > > Please let me know what you think, and let me know of any features you > > > want, > > > > > > bugs you find, etc. > > > > > > Thanks, > > > > > > Mike Williams > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > This e-mail and any attachment is for authorised use by the intended > > > recipient(s) only. It may contain proprietary material, confidential > > > information and/or be subject to legal privilege. It should not be > > > copied, disclosed to, retained or used by, any other party. If you are > > > not an intended recipient then please promptly delete this e-mail and > > any > > > > attachment and all copies and inform the sender. Thank you. > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

1 0

Re: [Users] OpenSER Administrator Release
by Mike Williams 07 Aug '06

07 Aug '06

Hmm, Did you run 'gem install mysql'? According to some websites, this is needed. Also, what OS and version of MySQL are you running? Mike On Monday 07 August 2006 02:03, Louis Alexander wrote: > Unfortunately (fortunately?) I've allowed that user under the machines > domain name, 127.0.0.1, and localhost. Yet the issue still persists. Ill > keep trying to figure it out. > > Thanks > > Louis > > -----Original Message----- > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > Behalf Of Mike Williams > Sent: Monday, August 07, 2006 1:19 PM > To: users(a)openser.org > Subject: Re: [Users] OpenSER Administrator Release > > > Google seems to suggest that it might be a permissions problem: remember > that > to MySQL, localhost and 127.0.0.1 are different. You may want to look at > this > link: http://www.adamsdesk.com/be/?p=177 > > Mike > > On Monday 07 August 2006 01:02, Louis Alexander wrote: > > Mike, > > I'm having trouble logging into the OpenSER Administrator. I get > > the following error in my browser: > > > > Got any tips? > > > > > > ActiveRecord::StatementInvalid in LoginController#login > > > > Mysql::Error: Lost connection to MySQL server during query: SELECT * FROM > > users WHERE (login = 'openser' AND password = > > '0dd62bae0b6b4d8e0388c620134decb8405b6976') LIMIT 1 > > > > RAILS_ROOT: script/../config/.. > > Application Trace | Framework Trace | Full Trace > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/abstract_adapter.rb:120:in `log' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:184:in `execute' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:336:in `select' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:175:in `select_all' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:390:in `find_by_sql' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:924:in `find_every' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:918:in `find_initial' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:380:in `find' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/dep > > >r ecated_finders.rb:22:in `find_first' > > #{RAILS_ROOT}/app/models/user.rb:7:in `authenticate' > > app/controllers/login_controller.rb:11:in `login' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/abstract_adapter.rb:120:in `log' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:184:in `execute' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:336:in `select' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:175:in `select_all' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:390:in `find_by_sql' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:924:in `find_every' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:918:in `find_initial' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:380:in `find' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/dep > > >r ecated_finders.rb:22:in `find_first' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >a se.rb:910:in `perform_action_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > >i lters.rb:368:in `perform_action_without_benchmark' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > /usr/local/lib/ruby/1.8/benchmark.rb:293:in `measure' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/r > > >e scue.rb:82:in `perform_action' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >a se.rb:381:in `process_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > >i lters.rb:377:in `process_without_session_management_support' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/s > > >e ssion_management.rb:117:in `process' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/dispatcher.rb:38:in > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:115:i > >n `handle_dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:81:in > > `service' > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:104:in `service' > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:65:in `run' > > /usr/local/lib/ruby/1.8/webrick/server.rb:173:in `start_thread' > > /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start_thread' > > /usr/local/lib/ruby/1.8/webrick/server.rb:95:in `start' > > /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `start' > > /usr/local/lib/ruby/1.8/webrick/server.rb:23:in `start' > > /usr/local/lib/ruby/1.8/webrick/server.rb:82:in `start' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:67:in > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/servers/webrick. > > >r b:59 > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > >p endencies.rb:147:in `require' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/server.rb:30 > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > >p endencies.rb:147:in `require' > > script/server:3 > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/abstract_adapter.rb:120:in `log' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:184:in `execute' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:336:in `select' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/con > > >n ection_adapters/mysql_adapter.rb:175:in `select_all' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:390:in `find_by_sql' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:924:in `find_every' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:918:in `find_initial' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/bas > > >e .rb:380:in `find' > > /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.3/lib/active_record/dep > > >r ecated_finders.rb:22:in `find_first' > > #{RAILS_ROOT}/app/models/user.rb:7:in `authenticate' > > app/controllers/login_controller.rb:11:in `login' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >a se.rb:910:in `perform_action_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > >i lters.rb:368:in `perform_action_without_benchmark' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > /usr/local/lib/ruby/1.8/benchmark.rb:293:in `measure' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >e nchmarking.rb:69:in `perform_action_without_rescue' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/r > > >e scue.rb:82:in `perform_action' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/b > > >a se.rb:381:in `process_without_filters' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/f > > >i lters.rb:377:in `process_without_session_management_support' > > /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.12.3/lib/action_controller/s > > >e ssion_management.rb:117:in `process' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/dispatcher.rb:38:in > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:115:i > >n `handle_dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:81:in > > `service' > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:104:in `service' > > /usr/local/lib/ruby/1.8/webrick/httpserver.rb:65:in `run' > > /usr/local/lib/ruby/1.8/webrick/server.rb:173:in `start_thread' > > /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start_thread' > > /usr/local/lib/ruby/1.8/webrick/server.rb:95:in `start' > > /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `start' > > /usr/local/lib/ruby/1.8/webrick/server.rb:23:in `start' > > /usr/local/lib/ruby/1.8/webrick/server.rb:82:in `start' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/webrick_server.rb:67:in > > `dispatch' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/servers/webrick. > > >r b:59 > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > >p endencies.rb:147:in `require' > > /usr/local/lib/ruby/gems/1.8/gems/rails-1.1.4/lib/commands/server.rb:30 > > /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in > > `require' > > /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.3.1/lib/active_support/de > > >p endencies.rb:147:in `require' > > script/server:3 > > > > Request > > > > Parameters: {"user_login"=>"openser", "user_password"=>"openserrw", > > "login"=>"Login \302\273"} > > > > Show session dump > > > > --- > > return-to: /login > > flash: !map:ActionController::Flash::FlashHash {} > > > > > > Response > > Headers: {"cookie"=>[], "Cache-Control"=>"no-cache"} > > > > -----Original Message----- > > From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On > > Behalf Of Mike Williams > > Sent: Friday, August 04, 2006 10:38 AM > > To: users(a)openser.org > > Cc: devel(a)openser.org > > Subject: [Users] OpenSER Administrator Release > > > > All, > > > > I've just finished my first release of OpenSER Administrator, a web > > interface > > for managing OpenSER. It's GPL'd and written in Ruby on Ralis, and can be > > downloaded from openseradmin.sourceforge.net > > > > Please let me know what you think, and let me know of any features you > > want, > > > > bugs you find, etc. > > > > Thanks, > > > > Mike Williams > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > This e-mail and any attachment is for authorised use by the intended > > recipient(s) only. It may contain proprietary material, confidential > > information and/or be subject to legal privilege. It should not be > > copied, disclosed to, retained or used by, any other party. If you are > > not an intended recipient then please promptly delete this e-mail and any > > attachment and all copies and inform the sender. Thank you. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Confused on routing...
by Marvin Ottley 07 Aug '06

07 Aug '06

Hello all, I'm trying to implement routing in general and I'm also trying to use the lcr module. I'm a little confused as to where in the picture does xml fit in. While reading through documentation I came across the acronym RPID which seems to be an implementation of routing information in an xml document, that's passed to lcr module as a parameter. What are the mechanics behind this? Thanks in advance for your help, Marvin

2 1

[Users] OpenSER Administrator Release
by Mike Williams 07 Aug '06

07 Aug '06

All, I've just finished my first release of OpenSER Administrator, a web interface for managing OpenSER. It's GPL'd and written in Ruby on Ralis, and can be downloaded from openseradmin.sourceforge.net Please let me know what you think, and let me know of any features you want, bugs you find, etc. Thanks, Mike Williams

2 2

[Serusers] SER + Asterisk PSTN calls don't hung up
by Ricardo Carvalho 07 Aug '06

07 Aug '06

Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users database and sip routing, and Asterisk is used for voicemail and PSTN gateway. The system is already able to make and receive calls from the PSTN, although, only after the call has been established it can be hung up with success; when it is still ringing, if any side hungs up the call, it still keeps ringing on the other side. Observing with Ethereal, we concluded that in this erroneous cases, the CANCEL SIP request isn't transmitted from the SER to Asterisk (if cancelled from the VoIP side) being transmitted a "404 User Not Found" message from SER to Sip Phone. If hung from the PSTN side, the sip phone keeps calling after that, and ends calling by time-out being observed a "486 Busy Here" status message from Asterisk to SER and then from SER to sip phone. Any help, please? Regards, Ricardo.

1 1

Re: [Serusers] mysql change tcp port
by Andrey Kouprianov 07 Aug '06

07 Aug '06

Wow! Thanx!!! On 8/7/06, sip <sip(a)arcdiv.com> wrote: > Usually, that's done in the /etc/my.cnf file. Is it not working there? > > My my.cnf file has a client ([client]) and server ([mysqld]) section. > > My server section (the non-commented part) looks like: > > [mysqld] > port = 3306 > socket = /var/lib/mysql/mysql.sock > > The client section looks the same (since I like the defaults to work the same > when connecting via the command-line client): > > [client] > [mysqld] > port = 3306 > socket = /var/lib/mysql/mysql.sock > > > There were a few packages I had that didn't come with an /etc/my.cnf file but > still read from one if it was there. > > N. > > On Mon, 7 Aug 2006 19:47:54 +0700, Andrey Kouprianov wrote > > Hi, > > > > I know it's the wrong list and I dont even expect anyone to answer. > > Anyway... Does anyone know how to specify different listening TCP > > port and socket name when installing mysql server from FreeBSD ports? > > > > Andrey. > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] mysql change tcp port
by Andrey Kouprianov 07 Aug '06

07 Aug '06

Hi, I know it's the wrong list and I dont even expect anyone to answer. Anyway... Does anyone know how to specify different listening TCP port and socket name when installing mysql server from FreeBSD ports? Andrey.

1 0

[Users] routes configurations
by Lens Oreste 07 Aug '06

07 Aug '06

Hi all I am new to openser and would like to configure my routes. I would like to have some samples. Please point me on the right direction. Regards Lens Oreste

2 1

[Users] rfc support
by abhishek kaushik 07 Aug '06

07 Aug '06

hi all , can anybody tell me if OpenSER supports rfc 4028? thnks abhishek

2 1

[Serusers] Simple Request/Response Forwarding
by Sebastian Gabris 07 Aug '06

07 Aug '06

Hello everybody, i'm new to SER and try to write my first config file. Since my english is not that good and i dont have much time, i want to minimize my reading of SER tutorials. This is my VoIP - test enviroment: UA(private/local IP) <-> (private/local IP)SER(public IP) <-> (public IP) VoIP - Provider I want SER to simply relay/forward all incoming and outgoing calls. I dont want that SER should have a SIP -domain, no authentication or registration. RTP - Streams should also be forwarded over the Proxy. No my questions: What is the simplest way to implement this? As far as i know i need the rtpproxy module. Do i need the nathelper or any other important modules? I also figured out to use the mhomed option. Has someone a sample config file for such a scenario? Any Documentation in German for the config file? Thanks in advance and any Help & Tipps would be very appreciated. _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

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[Users] OpenSER v1.1.0 notls in Debian unstable official repository
by Daniel-Constantin Mierla 07 Aug '06

07 Aug '06

Hello, thanks to the work of Julien BLACHE and pkg-voip-maintainers group from Debian, OpenSER v1.1.0 is available via official Debian APT repositories with the unstable distribution. http://packages.debian.org/unstable/net/openser People liking to use default repositories can remove the openser.org site from sources.list. Beware that the version included in Debian does not include the TLS support -- integration of GPL and OpenSSL license to satisfy Debian policies being in progress. If you want to use the TLS version, stay with openser.org's repository for a while. Cheers, Daniel

1 0

[Users] Rewriting Contact HF port
by Justin Pearce 07 Aug '06

07 Aug '06

I am wondering if there is a good way to rewrite the port in the contact header in OpenSER 1.0.1 before it is stored in the location table?(I'm using the MySQL database backend). Thus far, I am using "subst(/^(Contact.+):[0-9]+(.+)$/\1:5060\2/ig);" just before I do "save("location");", but that does not seem to be working. Suggestions? Justin Pearce Information Technology/Programming Price Video Productions JustinP(a)PriceVideo.com 361-572-3810 800-733-3810 Fax: 361-572-3894 www.PriceVideo.com

2 1

FW: [Serusers] using SER for SIP conversion from H323
by Richard C. Thompson 06 Aug '06

06 Aug '06

I know this may sound like heresy but a Quintum call relay does a great job doing this. We use a number of them with our SER and asterisk servers. Rick _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Torres, Javier Sent: Wednesday, August 02, 2006 12:19 PM To: serusers(a)lists.iptel.org Subject: [Serusers] using SER for SIP conversion from H323 Hello, I was wondering if anyone was using this in conjunction with Cisco Call manager to get out to a voip network. Call manager is horrible at SIP so I was hoping to have a h323 connection from Call manager to SER, the SER box would take that connection and forward it with SIP out to a voip peering fabric we are testing out. I know asterisk can be used in this fashion but had some voice quality issues and figured this piece of software might be better suited. Just hoping someone has done this and can confirm and maybe point me in the right direction for a how-to? Thanks, Javier

2 1

Re: [Serusers] registered UA's
by Jiri Kuthan 06 Aug '06

06 Aug '06

Well, I think there is nothing else better than troubleshooting and looking at what it is message by message. I agree NAT issues may be worth looking at. Other problem we experience frequently is that we drive registration period low, indicate that using Expires header-field, and client is silly not to honor it. -jiri At 17:14 03/08/2006, Weiter Leiter wrote: >My number one suspect would be the nat. >WL. > >On 8/3/06, Richard C. Thompson <<mailto:rthompson@vir2com.com>rthompson(a)vir2com.com> wrote: > >Hi All > > > >I hope someone can help me figure this out. We've had a ser server running for quite some time now on a Dell 2850 with dual cpu's. The load is very low due to the use of multiple mediaproxy servers to handle the RTP streams. We have about 2300 UID in the mysql database but at any given time only about 1300 show as registered in the location table and we have customers complaining about not receiving calls. When tech support has a customer power cycle their ATA, it registers just fine and they start receiving calls. What could cause this? From the beginning, I noticed this was happening but didn't give it much thought as the number of UIDs was low. But now it is a growing problem. SER version is 0.9.0 > > > >Any hints or help would be appreciated. > > > >Rick > > > > > > > >_______________________________________________ >Serusers mailing list ><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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