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RE: [Serusers] problem in using acc.so and radius_acc.so and auth.so
by Benny Ho 25 Feb '03

25 Feb '03

Dear Jiri, I have successfully compile the acc.so with SQL support. And I run a number of testing on it. It seems works fine but I encounter the following error which crach the ser. =========================================================== [root@md benny]# 0(3545) mod_init(): Database connection opened successfuly Maxfwd module- initializing rr - initializing acc - initializing stateless - initializing mysql - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 9(3565) INFO: fifo process starting: 3565 9(3565) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 10(3574) ACC: transaction answered: method=INVITE, i-uri=sip:1000@md.com;user=ph one, o-uri=sip:1000@192.168.10.18:5000;user=phone, call_id= 426a69eafe94e7923666 1278d86d5024(a)192.168.10.18, from= UserAgent-1<sip:1000@192.168.10.18:5000;user=p hone>, code=408 10(3574) use_table(): Invalid parameter value 0(3545) child process 3574 exited by a signal 11 0(3545) core was not generated 0(3545) INFO: terminating due to SIGCHLD ========================================================================= It seems to be a crash on use_table() when code=408, I tried to unanswer to call or call to itself, and this problem generated. Benny -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Monday, February 24, 2003 6:30 PM To: Benny Ho Subject: RE: [Serusers] problem in using acc.so and radius_acc.so and auth.so First of all, note well: http://lists.iptel.org/pipermail/serusers/2003-February/000493.html At 04:36 AM 2/24/2003, Benny Ho wrote: >Could you tell me how to enable SQL_ACC def? and setting teh db_url >parameter? uncomment the "#DEFS+=-DSQL_ACC" line in modules/acc/Makefile >I cannot find any doc for mysql modules. mysql module is not used from ser scripts directly -- modules which rely on mysql use it internally. acc w/mysql has not been released yet, it is work in progress on CVS and it has not been documented. -jiri

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Re: [Serusers] Redundant forwarding
by lgfausak＠august.net 24 Feb '03

24 Feb '03

I'm currently working with SNOM to get this issue fixed. I'll report when I know something. It might not hurt if someone else reported it to them! ---greg Greg Fausak > >At 09:34 PM 2/24/2003, Jamin W. Collins wrote: >>On Mon, Feb 24, 2003 at 08:54:00PM +0100, Jiri Kuthan wrote: >>> What is CIC? >> >>A PBX made by I3 (www.inin.com) >> >>> Send us network dumps please -- that will make it easier to understand >>> for us. You can for example use "ngrep port 5060". >> >>Network dump from the SER server is attached. > >snom phone is a common suspect again. > >The first signaling transaction looks ok. The only unusual thing is that >for some reason, the ININ server advertises 192.168.0.9 as media receiver >whereas its signaling lives at 192.168.1.16. Is media decomposed? > >The really suspicious is the second transaction. The snom phone sitting >at .36 advertises contacts with .34 in its replies. That makes the >caller (ININ) to send an ACK to .34. Snom does not received it and >keeps soliciting it by retransmitting 200s. > >-Jiri > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Some thoughts and questions
by Tomas Björklund 24 Feb '03

24 Feb '03

Hi, I've been looking at ser for a couple of days now and there some things I can't find in the documentation. .1. Logging funktionality I'm used to being able to log more detail than what I can get out ser. I'm problably missing something in the documentation, but what I want is something like the following timestamp ser[pid] Incoming packet from ip:port [dump of packet (only the sip header/body)] timestamp ser[pid] Invite from [] to [] timestamp ser[pid] After rewrite: from [] to [] timestamp ser[pid] Relaying packet to [pstn-gw/other sip server] / timestamp ser[pid] Handling request for local user [] Can you send arguments to the log() function in the configuration file for example log(1,"Call from %s", uri:from); ?? .2. Rewrite rules Is it possible to rewrite the from uri? For example when I [sip:tomas@krixor.xy.org] want to call a pstn number I must rewrite the from address to [sip:myphonenumber@krixor.xy.org] before I send the request to the pstn gateway And when a incoming call comes from the pstn gateway I want to rewrite the from address from [<sip:number@ip-of-pstn-gateway>] to {"PSTN Luser" <sip:number@pstn-gw.mydomain.com>] I haven't looked at the source in the CVS yet, but I aim to do so soon, I saw in the list archive that you have lots of nice new features planned for the next release. ENUM support is one of the better ones :) Anyway, looking forward to trying to help with the development of this software. Best regards, Thomas Björklund

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[Serusers] Rewriting URI in the Contact field
by Maxim Sobolev 24 Feb '03

24 Feb '03

Folks, I need an advise on how to better implement one feature, which isn't currently present in SER. We need to allow UAs behind NAT properly register with the registrar - by "properly" I mean that host:port portion of URI in Contact field should not be used, but host:port the request came from should be used instead. By definition we know that those UAs will support symmetric SIP signalling, so that this scheme will work just fine. In my opinion there are two ways to do it: either add new rewritecontact* family of functions similar to rewritehost ones. or add a new flag for the save() function. This is where I need your help - which implementation looks better for you (or maybe you have even some better idea), since we are really interested in inclusion of our changes into the mainline to reduce our local hacks. Regards, Maxim

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[Serusers] Some patches for integration
by Maxim Sobolev 24 Feb '03

24 Feb '03

Folks, Attached please find several patches for ser 0.8.10. They do the following: 1. Removes hardcoded `/usr/local' from Makefiles and replaces it with LOCALBASE variable setable in the top Makefile.defs. This should make life of packagers easier. 2. Allows to override PREFIX variable with something else from environment. 3. Adds support for FreeBSD doc/man directories layout. 4. Adds some missed -I's into radius modules. 5. Fixes radius modules to use the same symbolic avp names as in dictionary. 6. Adds support for recording source port in msg structure and adds it in "Noisy feedback". This should be extremely useful for debugging various SIP&NAT scenarios and later could be re-used to implement `rport' support described in my previous message. It would be nice to have those patches integrated into the next release. Thanks! -Maxim

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[Serusers] Record-route:/Contact: conflict?
by Greg Fausak 24 Feb '03

24 Feb '03

I've got a packet that looks like: # U 2003/02/24 07:56:52.503535 216.87.144.203:5060 -> 216.87.145.22:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-ng5tokyx448r. From: "snom man" <sip:4695461245@augustvoice.net>;tag=8u6ju8wxuc. To: <sip:2143357976@augustvoice.net;user=phone>;tag=3CBB0360-532. Date: Mon, 24 Feb 2003 13:56:43 GMT. Call-ID: 3c267202b6a8-lgseu8olovlp(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Session-Expires: 7200;refresher=uac. Require: timer. Allow-Events: telephone-event. Contact: <sip:92143357976@216.87.144.196:5060;user=phone>. Record-Route: <sip:2143357976@216.87.144.203;ftag=8u6ju8wxuc;lr>. Content-Type: application/sdp. Content-Length: 209. . v=0. o=CiscoSystemsSIP-GW-UserAgent 7543 5694 IN IP4 216.87.144.196. s=SIP Call. c=IN IP4 216.87.144.196. t=0 0. m=audio 16632 RTP/AVP 0 100. a=rtpmap:0 PCMU/8000. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. The main thing to note is the packet is sent to a SNOM phone from my PROXY server. Curious, when the SNOM phone responds it is responding to 216.87.144.196, which is my GATEWAY. My GATEWAY is configured to only listen to the PROXY server, so it drops the packet. In the packet above, the Record-route line indicates that 216.87.144.203 should be responded to, which is my PROXY. Also in the packet above, there is a 'Contact:' line that indicates the GATEWAY at 216.87.144.196 should be contacted. Obviously the SNOM phone is getting the address from the Contact: line (or maybe the SDP) rather than the Record-route: line. Is it doing so incorrectly, or do I have a bug in my GATEWAY, or does the PROXY need to strip out the Contact: line?

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[Serusers] SER parser benchmarks
by Sebastien Pierre 24 Feb '03

24 Feb '03

Hi! I was just wondering if you had some benchmark for the SER parser... as SER does not seem to offer a libser, it makes it more difficult to write benchmarking programs :/ TIA, -- Sebastien

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Re: [Serusers] FCP support in SER: Modifying SDP
by jaime.gill＠orange.co.uk 24 Feb '03

24 Feb '03

Thanks Jiri, I'll have a look at the CVS version. I'm sure there will be more hacking to do only to make it compile.... :( However, got the replacement of the "c=.." part working. I'll test this as soon as I can. Jaime Jiri Kuthan <jiri(a)iptel.org> on 24/02/2003 13:20:00 To: Jaime GILL/EN/HTLUK@HTLUK Nils Ohlmeier <nils(a)ohlmeier.de> cc: Jan Janak <J.Janak(a)sh.cvut.cz> serusers(a)lists.iptel.org Subject: Re: [Serusers] FCP support in SER: Modifying SDP At 12:50 PM 2/24/2003, jaime.gill(a)orange.co.uk wrote: [...] >It think the problem is in the replacement of the SDP information. The first >occurrence of the IP address in "v= " and the port in "m= " in the SDP get >replaced, but the second IP in "c=" is not. nit: it's not the "v=" line, but "o=" ('owner') line which you are replacing. However, you are not probably worried so much about this one -- it maintains primarily a (not widely utilizied) identification purpose. All "c=" occurences do matter. (In addition to port numbers in "m=" lines.) [...] >I have been trying to understand how the proxy builds the forwarded message from >the old one, and realised that for the Via replacement (or adding of more >params), I need to be using a string called add_to_branch_s and >add_to_branch_len (so ignore the replace_via implementation in the current >tar.gz). I suggest you used the mhomed option (available only on CVS). The issue is you need to print the correct IP address in Via on multihomed host. With mhomed enabled, IP routing is utilized to determine the right IP address. Let me know if you need something more for getting Via right. >But for the SDP, whenever I work with get_body, it does not modify it >appropriately. So currently, I'm using msg->orig to get to the initial message, >search for certain IP4 and audio strings and replace them with the information >provided by the fcp server. That means, in the case of the SDP, 2 IP address >replacements (in v=.. and c=..) and 1 port replacement (in m=..). As I mentioned >before, I only manged to change the v=.. and m=... Whenever I try to replace >more than one appearance, strange things happen, like strings in non expected >places, like Via, and cannot work out why. So my question is an open one:): what >is the best way to change the SDP part? I suggest here too -- use the CVS version. It has departed from the use of the buffers (orig and buf) -- we have now just one buffer (buf) without any zero termination. Previously, the two buffers and 0-termination caused lot of issues, some of them possibly annoying you right now. Look at the textops/replace_all action (only on CVS too) to see how to replace multiple occurences of a string in SIP messages. (Caution: you will eventually need to calculate new SDP body size and change content-length too.) >The other of my questions is whether all this mess with NAT's will get solved >when the proxy supports TCP, The major problem is media, which will keep using UDP. >and whether this is the best approach to solve the >SIP through NAT/FW problem. As all NAT traversal methods -- none of them is perfect, each has cons and pros. The benefit of FCP is that once fcpd works, maintenance of the SIP code is easier in user space. Also, you can better couple your pinhole policy with SER's SIP-layer policy. >For example, how about a nathelper module for >netfilter/iptables that gets this working, in the same manner as IRC or ftp >currently? Does anybody know about any work progressing this for linux/FreeBSD? I'm not aware of such. There is Billy Biggs masquerading module, but it is pretty old and no longer maintained. -Jiri _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

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Re: [Serusers] FCP support in SER: Modifying SDP
by jaime.gill＠orange.co.uk 24 Feb '03

24 Feb '03

Hi Nils, Once again I'm writing to ask about more things and to give you an update on what is happening with the FCP tests... We have set up a simple scenario with 2 proxies separated by a firewall/NAT, and 2 UA, one within (UA1) and another outside the firewall (UA2). One UA registers with the natted proxy (UA1), the other with the "public" proxy (UA2). At the moment, SIP messages go forwards and backwards without problems, but media is not flowing across the firewall. It think the problem is in the replacement of the SDP information. The first occurrence of the IP address in "v= " and the port in "m= " in the SDP get replaced, but the second IP in "c=" is not. I have been trying all sorts of things, but no joy :( . Here are the INVITE messages in more detail. I also include the latest fcp module for you to play with it. ---------- UA1 to proxy message ------------------ U 172.21.68.78:1129 -> 192.168.6.153:5060 INVITE sip:jaime@asereje.orange.co.uk SIP/2.0 Call-ID: 5812832001907970791(a)172.21.68.78 Content-Length: 121 Content-Type: application/sdp To: sip:jaime@asereje.orange.co.uk From: sip:pepe@asereje.orange.co.uk;tag=-779729009 Contact: sip:pepe@172.21.68.78:5061 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.21.68.78:5061;branch=AC15444E13C5000000F38F819DE5-2*0 v=0..o=- 1046084768435 1046084768465 IN IP4 172.21.68.78 s=- c=IN IP4 172.21.68.78 t=0 0 m=audio 5006 RTP/AVP 8 3 0 ---------- End of UA1 to proxy message ----------------- -------- Proxy to UA2 message --------------------- U 192.168.6.153:5060 -> 172.21.68.78:15592 INVITE sip:172.21.68.78:15592 SIP/2.0 Call-ID: 5812832001907970791(a)172.21.68.78 Content-Length: 121 Content-Type: application/sdp To: sip:jaime@asereje.orange.co.uk From: sip:pepe@asereje.orange.co.uk;tag=-779729009 Contact:<sip:192.168.0.1:33240> CSeq: 1 INVITE..Via: SIP/2.0/UDP 192.168.6.153;branch=z9hG4bK1019.21c52996.0 Via: SIP/2.0/UDP 172.21.68.78:5061;branch=AC15444E13C5000000F38F819DE5-2*0 v=0..o=- 1046084768435 1046084768465 IN IP4 192.168.0.1 s=- c=IN IP4 172.21.68.78 <--- Need to change this as well!!! t=0 0 m=audio 33240 RTP/AVP 8 3 0 ------------ End of Proxy to UA2 message ------------------- (See attached file: fcp-module210203.tar.gz) I have been trying to understand how the proxy builds the forwarded message from the old one, and realised that for the Via replacement (or adding of more params), I need to be using a string called add_to_branch_s and add_to_branch_len (so ignore the replace_via implementation in the current tar.gz). But for the SDP, whenever I work with get_body, it does not modify it appropriately. So currently, I'm using msg->orig to get to the initial message, search for certain IP4 and audio strings and replace them with the information provided by the fcp server. That means, in the case of the SDP, 2 IP address replacements (in v=.. and c=..) and 1 port replacement (in m=..). As I mentioned before, I only manged to change the v=.. and m=... Whenever I try to replace more than one appearance, strange things happen, like strings in non expected places, like Via, and cannot work out why. So my question is an open one:): what is the best way to change the SDP part? The other of my questions is whether all this mess with NAT's will get solved when the proxy supports TCP, and whether this is the best approach to solve the SIP through NAT/FW problem. For example, how about a nathelper module for netfilter/iptables that gets this working, in the same manner as IRC or ftp currently? Does anybody know about any work progressing this for linux/FreeBSD? Greetings, Jaime Nils Ohlmeier <nils(a)ohlmeier.de> on 18/02/2003 02:58:47 To: Jaime GILL/EN/HTLUK@HTLUK cc: Jan Janak <J.Janak(a)sh.cvut.cz> Jiri Kuthan <jiri(a)iptel.org> Subject: Re: [Serusers] FCP support in SER: Modifying SDP Hi Jaime, debugging without the code is really hard :-) But maybe your problme with SDP is correlated to a bug in the Via header which i marked below. Greetings Nils On Monday 17 February 2003 11:58, jaime.gill(a)orange.co.uk wrote: > U 192.168.6.153:5060 -> 172.21.68.78:5061 > INVITE sip:pepe@172.21.68.78:5061 SIP/2.0..Via: SIP/2.0/UDP > 192.168.6.153;b ranch=z9hG4bKb848.8a014f84.0..Via: SIP/2.0/UDP > 192.168.0.1192.168.0.1:9439. .From: "jaime" ^^^^^^^^^^^^^^^^^^^^^^ Here you inserted the external IP twice. Maybe your SDP replacer did this? > <sip:jaime@asereje.orange.co.uk>;tag=8c20540f-4259-11d7-9cc5 > -00065b4c11cb..To: <sip:pepe@asereje.orange.co.uk>..Call-ID: 8c205410-4259- > 11d7-9cc5-00065b4c11cb@172.21.68.78..CSeq: 1 INVITE..Contact:<sip:192.168.0 > .1:33186>.User-Agent: Windows RTC/1.0..Content-Type: application/sdp..Conte > nt-Length: 211....v=0..o=gill_j 0 0 IN IP4 172.21.68.78..s=session..c=IN IP > 4 172.21.68.78..b=CT:1000..t=0 0..m=audio 33186 RTP/AVP 97 0 8 4..a=rtpmap: > 97 red/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:4 G723/80 > 00.. -- gpg-key: http://www.ohlmeier.org/public_key.asc ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

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[Serusers] Cannot add "alias" mysql DB (using default passwords)
by Lenny Tropiano 24 Feb '03

24 Feb '03

I have turned on mysql authentication, and was able to add users, but aliases complain that "table" doesn't exist: $ serctl add lenny abcdef lenny(a)voiping.com MySql Password: new user added $ serctl alias add 1234 sip:lenny@voiping.com sip:lenny@voiping.com 400 Table 'aliases' Not Found $ mysql -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 243 to server version: 3.23.53 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql> connect ser; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Connection id: 244 Current database: ser mysql> show tables; +-----------------+ | Tables_in_ser | +-----------------+ | acc | | active_sessions | | aliases | | config | | event | | grp | | location | | missed_calls | | pending | | phonebook | | reserved | | silo | | subscriber | | version | +-----------------+ 14 rows in set (0.00 sec) mysql> --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

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[Serusers] On CVS, Radius and Generally SER Status (2003/II)
by Jiri Kuthan 23 Feb '03

23 Feb '03

I'm observing increasing number of downloads from CVS and would like to issue few disclaimers to prevent any disappointments. The CVS code is work in progress -- it has not been integrated, tested, it is not complete and we cannot provide help on incomplete software. Also, folks should know that compilation of one of the most frequently asked features, Radius support, is disabled on purpose. It is an external contribution which deserves a clean-up, we are working on it but we are not there yet. My previous forecast for the upcoming release is turning out to be too optimistic. We decided to carry out some more clean-ups and the work takes time. Anyway, we think it is better to clean up now than later -- interest in ser is growing, and we need to keep the codebase sane despite all the new features. -Jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] problem in using acc.so and radius_acc.so and auth.so
by Benny Ho 23 Feb '03

23 Feb '03

Dear all, after days of trial, I finally load radius_acc.so in my ser. But when I testing the modules with the new routing config. Nothing happen. I follow the sample config in the CVS as follow. I check the radius server with radius -x, no packet seem to send to radius server. I also check the Mysql database, no entry is wrote to acc table. Why? Can anyone tell me what's wrong? ================== [benny@md benny]$ more /etc/ser/ser.cfg # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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/lib/ser/modules/mysql.so" [benny@md benny]$ cat /etc/ser/ser.cfg # # $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) 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/lib/ser/modules/mysql.so" # Uncomment this to use radius for accounting loadmodule "/usr/lib/ser/modules/radius_acc.so" loadmodule "//usr/lib/ser/modules/acc.so" #--- radius_ acc.so params --- modparam("radius_acc","log_level",1) modparam("radius_acc","acc_flag",1) modparam("radius_acc","report_ack",1) modparam("radius_acc","early_media",0) modparam("radius_acc","failed_transactions",0) #--- acc.so params --- modparam("acc","log_level",1) modparam("acc","acc_flag",1) modparam("acc","report_ack",1) modparam("acc","early_media",1) modparam("acc","failed_transactions",1) loadmodule "//usr/lib/ser/modules/sl.so" loadmodule "//usr/lib/ser/modules/tm.so" loadmodule "//usr/lib/ser/modules/rr.so" loadmodule "//usr/lib/ser/modules/maxfwd.so" loadmodule "//usr/lib/ser/modules/usrloc.so" loadmodule "//usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "//usr/lib/ser/modules/auth.so" # ----------------- setting module-specific parameters --------------- # -- 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", "secret", "alsdkhglaksdhfkloiwr") #modparam("auth", "calculate_ha1", yes) #modparam("auth", "calculate_ha1", no) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present # rewriteFromRoute(); #if (method=="INVITE") { addRecordRoute(); #} else { rewriteFromRoute(); #}; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #labeled all transaction for accouting setflag(1); #record-route INVITES to make sure BYEs will visit our server too #if (method=="INVITE") addRecordRoute(); # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } [benny@md benny]$ ==================================

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[Serusers] NAT Traversal Technique Survey
by Jiri Kuthan 23 Feb '03

23 Feb '03

Folks, please help me -- share with me techniques for NAT traversal you use and have hands-on experience with. People repeatedly ask about it, and I'd like to create an FAQ that reflects deployment experience and as wide user feed-back as possible. Just tell me the technique you use, its requirements, limitations, the devices it is known (not) to work with, why you prefer one method over the other, etc. I'll then try to compile it in an FAQ. So please send me an e-mail, an example is attached. I will appreciate any practical details. Thank you, -Jiri ---------------------------------------------------------------- technique: using symmetric communication requirements: phone devices that support symmetric communication; existing species: Cisco's ATA configuration practice: ATAs need to be configured to advertise public address in signaling, or learn it from REGISTER replies; alternatively, one can rewrite signaling using ser's nethelper module; one needs to rewrite SIP anyway because ATAs don't advertise their symmetricity; see www.foo.bar for info on configuring ATA... limitations: non-symmetric devices, like Messenger don't work; misc: ATA has no display, that's why I am anxiously waiting for more vendors to support symmetric signaling ---------------------------------------------------------------- technique: UPnP requirements: NATs and phones with UPnP support; Messenger and snom are known to support UPnP; there is linux support for it configuration practice: of course, upnp requires by definition no configuration ;-) (I'm not serious -- anyone actually tried it?) ---------------------------------------------------------------- technique: geek tweaks: set-up port forwading manually configuration practice: you need to configure NATs to split its public-side port numbers accross your private-side phones, and configure the phones (if they allows so) to use these port numbers; also, phones need to be configured to use publicly reachable address in their payloads requriements: configurable NATs (many residental NATs are configurable) and configurable phones (ATAs do that, I heard pingtel did it too) ---------------------------------------------------------------- technique: ALG requirements: SIP-capable NAT (like Intertex or Cisco/PIX) issues: intertex freezes my ssh connections after some time on-line and elderly models don't like all Ethernet devices; when things don't work, the red-button off-on helps sometimes ---------------------------------------------------------------- technique: STUN requirements: STUN-enabled phone (like k-phone, snom) limitations: doesn't work over symmetric NATs (words-of-mouth propaganda has been telling me that many residential NATs are fortunately not symmetric, but I don't know how objective this information really is) ---------------------------------------------------------------- -- Jiri Kuthan http://iptel.org/~jiri/

4 7

[Serusers] Windows- Sms gateway
by ant buckland 21 Feb '03

21 Feb '03

Hi- I am a very Junior designer, and as part of my efforts to impress my boss, I am trying to create a simple email to sms application. Sadly, here in South Africa, the cellular service providers do not provide an sms gateway to the public, and so I am researching the iptel.org SIP Express Router. I need to know the following: after reading the iptel.org website, I would like to know whether it is possible to install the application on a Windows 2000 server, and run the gateway. From what I read, apparently only the core application can be installed... and this does not appear to include the gateway? Obviously, I will pass the project on to a programmer who knows more, so you will not be plagued by too-simple questions from an inexperienced designer... but I would appreciate any assistance/info that will allow me to convince my boss. Thank you! ph (021) 685 6499 anthony_buckland(a)yahoo.co.uk 70 Strubens Rd, Observatory, Cape Town --------------------------------- With Yahoo! Mail you can get a bigger mailbox -- choose a size that fits your needs

2 1

[Serusers] Absorbing provisional responses
by Sebastien Pierre 21 Feb '03

21 Feb '03

Hi all, This message is somewhat related to a preceding one (Simple SER Use Case), because I realised I did not ask the right question. Here it is: I want to make a proxy that will try to locate the callee by sequentially trying n different UA, by sending an INVITE and waiting for either OK or timeout to occur. On timeout, the proxy will try the next UA. The detail here is that the proxy sends a "100 Trying" to the caller *before* contacting the various UA, and that *no more* "100 Trying" are forwarded by the proxy, which absorbs those responses from contacted UA. Jiri indicated that the configuration presented in <http://www.iptel.org/ser/doc/seruser-html/x510.html#AEN616> realised the scenario I was mentioning... but it seems to me that the proxy does not send and then absorb "100" responses, but rather relays this responses to the caller. This would lead to the caller receiving as many "100" responses as contacted UAs. Is it possible to do this with SER ? TIA, -- Sebastien

3 4

[Serusers] Crash in SER
by Maxim Sobolev 21 Feb '03

21 Feb '03

Hi, We've observed the following crash in ser. Do you have any ideas what the cause could be? I'm keeping corefile and binary compiled with debug symbols, so that don't hesitate to contact me if any additional information is necessary. Thank you in advance! -Maxim Script started on Fri Feb 21 05:43:09 2003 bash-2.05a$ gdb ~/sip/ser/work/ser-0.8.10/ser ser.core GNU gdb 4.18 (FreeBSD) Copyright 1998 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-unknown-freebsd"...Deprecated bfd_read called at /usr/src/gnu/usr.bin/binutils/gdb/../../../../contrib/gdb/gdb/dbxread.c line 2627 in elfstab_build_psymtabs Deprecated bfd_read called at /usr/src/gnu/usr.bin/binutils/gdb/../../../../contrib/gdb/gdb/dbxread.c line 933 in fill_symbuf Core was generated by `ser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /usr/lib/libc.so.4...done. Reading symbols from /usr/local/lib/ser/modules/sl.so...done. Reading symbols from /usr/local/lib/ser/modules/tm.so...done. Reading symbols from /usr/local/lib/ser/modules/rr.so...done. Reading symbols from /usr/local/lib/ser/modules/maxfwd.so...done. Reading symbols from /usr/local/lib/ser/modules/usrloc.so...done. Reading symbols from /usr/local/lib/ser/modules/registrar.so...done. Reading symbols from /usr/local/lib/ser/modules/nathelper.so...done. Reading symbols from /usr/local/lib/ser/modules/textops.so...done. Reading symbols from /usr/local/lib/ser/modules/radius_auth.so...done. Reading symbols from /usr/local/lib/libradiusclient.so.0...done. Reading symbols from /usr/lib/libcrypt.so.2...done. Reading symbols from /usr/libexec/ld-elf.so.1...done. #0 0x807bdd3 in qm_status (qm=0x2816d000) at mem/q_malloc.c:447 447 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++); (gdb) bt #0 0x807bdd3 in qm_status (qm=0x2816d000) at mem/q_malloc.c:447 #1 0x8079a70 in qm_debug_frag (qm=0x2816d000, f=0x282fd0a4) at mem/q_malloc.c:109 #2 0x807ab37 in qm_free (qm=0x2816d000, p=0x282fd0bc, file=0x2a197a2d "t_reply.c", func=0x2a197e28 "t_on_reply", line=826) at mem/q_malloc.c:359 #3 0x2a188000 in t_on_reply (p_msg=0x80dfe30) at t_reply.c:826 #4 0x805933a in forward_reply (msg=0x80dfe30) at forward.c:346 #5 0x806af1c in receive_msg (buf=0x80c3340 "SIP/2.0", len=360, src_su=0x81dafa0) at receive.c:161 #6 0x8077ff0 in udp_rcv_loop () at udp_server.c:336 #7 0x805eaa5 in main_loop () at main.c:618 #8 0x806197d in main (argc=1, argv=0xbfbffc04) at main.c:1383 #9 0x804c5bd in _start () (gdb) up #1 0x8079a70 in qm_debug_frag (qm=0x2816d000, f=0x282fd0a4) at mem/q_malloc.c:109 109 qm_status(qm); (gdb) l 104 }; 105 if ((FRAG_END(f)->check1!=END_CHECK_PATTERN1)|| 106 (FRAG_END(f)->check2!=END_CHECK_PATTERN2)){ 107 LOG(L_CRIT, "BUG: qm_*: fragm. %p end overwritten(%x, %x)!\n", 108 f, FRAG_END(f)->check1, FRAG_END(f)->check2); 109 qm_status(qm); 110 abort(); 111 } 112 if ((f>qm->first_frag)&& 113 ((PREV_FRAG_END(f)->check1!=END_CHECK_PATTERN1) || (gdb) up #2 0x807ab37 in qm_free (qm=0x2816d000, p=0x282fd0bc, file=0x2a197a2d "t_reply.c", func=0x2a197e28 "t_on_reply", line=826) at mem/q_malloc.c:359 359 qm_debug_frag(qm, f); (gdb) l 354 return; 355 } 356 prev=next=0; 357 f=(struct qm_frag*) ((char*)p-sizeof(struct qm_frag)); 358 #ifdef DBG_QM_MALLOC 359 qm_debug_frag(qm, f); 360 if (f->u.is_free){ 361 LOG(L_CRIT, "BUG: qm_free: freeing already freed pointer," 362 " first free: %s: %s(%d) - aborting\n", 363 f->file, f->func, f->line); (gdb) up #3 0x2a188000 in t_on_reply (p_msg=0x80dfe30) at t_reply.c:826 826 shm_free(ack); (gdb) l 821 /* acknowledge negative INVITE replies */ 822 if (t->is_invite && (msg_status>=300 || (t->local && msg_status>=200))) { 823 ack = build_ack( p_msg, t, branch , &ack_len); 824 if (ack) { 825 SEND_PR_BUFFER( &uac->request, ack, ack_len ); 826 shm_free(ack); 827 } 828 } /* ack-ing negative INVITE replies */ 829 830 /* clean-up the transaction when transaction completed */ (gdb) up #4 0x805933a in forward_reply (msg=0x80dfe30) at forward.c:346 346 if (mod->exports->response_f(msg)==0) goto skip; (gdb) l 341 /* quick hack, slower for mutliple modules*/ 342 for (mod=modules;mod;mod=mod->next){ 343 if ((mod->exports) && (mod->exports->response_f)){ 344 DBG("forward_reply: found module %s, passing reply to it\n", 345 mod->exports->name); 346 if (mod->exports->response_f(msg)==0) goto skip; 347 } 348 } 349 350 /* we have to forward the reply stateless, so we need second via -bogdan*/ (gdb) print msg $1 = (struct sip_msg *) 0x80dfe30 (gdb) print msg [1@* $2 = {id = 1104, first_line = {type = 2, u = {request = {method = {s = 0x80c3340 "SIP/2.0", len = 7}, uri = {s = 0x80c3348 "488", len = 3}, version = { s = 0x80c334c "Not Acceptable Here", len = 19}, method_value = 488}, reply = {version = { s = 0x80c3340 "SIP/2.0", len = 7}, status = {s = 0x80c3348 "488", len = 3}, reason = { s = 0x80c334c "Not Acceptable Here", len = 19}, statuscode = 488}}}, via1 = 0x80e27fc, via2 = 0x80e2d1c, headers = 0x80e620c, last_header = 0x80e0b08, parsed_flag = 63, h_via1 = 0x80e620c, h_via2 = 0x80e0b98, callid = 0x80e0008, to = 0x80e10ac, cseq = 0x80e0b08, from = 0x80e0f2c, contact = 0x0, maxforwards = 0x0, route = 0x0, record_route = 0x0, content_type = 0x0, content_length = 0x0, authorization = 0x0, expires = 0x0, proxy_auth = 0x0, www_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, eoh = 0x0, unparsed = 0x80c3493 "Content-Length: 0\r\n\r\n", src_ip = { af = 2, len = 4, u = {addr32 = {1214690368, 0, 0, 0}, addr16 = {46144, 18534, 0, 0, 0, 0, 0, 0}, addr = "@╢fH", '\000' <repeats 11 times>}}, src_port_no = 5061, dst_ip = {af = 2, len = 4, u = {addr32 = {1214690368, 0, 0, 0}, addr16 = {46144, 18534, 0, 0, 0, 0, 0, 0}, addr = "@╢fH", '\000' <repeats 11 times>}}, orig = 0x80e2a48 "SIP/2.0 488 Not Acceptable Here\r\nVia: SIP/2.0/UDP 64.180.102.72:5060;branch=z9hG4bKfa9f.17ad3774445fef6b70f18b74de26e0b2.0\r\nVia: SIP/2.0/UDP 172.17.1.127:5060;rport=5060\r\nTo: <sip:151@64.180.102.72;us"..., buf = 0x80c3340 "SIP/2.0", len = 360, new_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0}, add_rm = 0x0, repl_add_rm = 0x80e0514, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 54 times>, add_to_branch_len = 0, hash_index = 0, flags = 0} (gdb) up #5 0x806af1c in receive_msg (buf=0x80c3340 "SIP/2.0", len=360, src_su=0x81dafa0) at receive.c:161 161 forward_reply(msg); (gdb) l 156 gettimeofday( & tvb, &tz ); 157 STATS_RX_RESPONSE ( msg->first_line.u.reply.statuscode / 100 ); 158 #endif 159 160 /* send the msg */ 161 forward_reply(msg); 162 163 #ifdef STATS 164 gettimeofday( & tve, &tz ); 165 diff = (tve.tv_sec-tvb.tv_sec)*1000000+(tve.tv_usec-tvb.tv_usec); (gdb) q bash-2.05a$ exit Script done on Fri Feb 21 05:44:36 2003

2 1

[Serusers] problem with loading compiled radius modules
by Benny Ho 21 Feb '03

21 Feb '03

I had the same error msgs when I load the radius_acc modules the following is the error msgs [root@md benny]# /usr/sbin/ser restart 0(3699) ERROR: load_module: could not open module </usr/lib/ser/modules/radius_ acc.so>: libradiusclient.so.0: cannot open shared object file: No such file or d irectory 0(3699) parse error (25,13-48): failed to load module ERROR: bad config file (1 errors) [root@md benny]# and I check that the radius_acc.so is in /usr/lib/ser/modules and the libradiusclient.so.0 is also in /usr/local/lib as it is stated in the Makefile. Where is the error came from? Have any one successfully implement it?

1 3

[Serusers] RE: RE: problem with loading compiled radius modules
by Yang Xiang 21 Feb '03

21 Feb '03

Hi Benny, I had the same problem as you. After I statically linked the radiusclient library in the both radius modules this problem seems to be resolved. http://lists.iptel.org/pipermail/serusers/2003-February/000419.html However radius authentication still doesn't work and the developer said: "the path is not staticaly linked, it should be resolved by you dynamic linker, I don't know where is the problem, it seems to me that there is some problem with radiusclient libraries on your system, " I use Solaris 8. Now I am trying to recompile and reinstall radiusclient library. If I find out something, I'll tell you. regards, yang

1 0

[Serusers] CVS version current, gives this error
by Lenny Tropiano 20 Feb '03

20 Feb '03

Default config (ser.cfg installed from gmake install), gives this error (I bumped up the debug level and had it log to stderr): p2:[root@enterprise.icus.com] 31 $ ser 0(62901) loading module /usr/local/lib/ser/modules/sl.so 0(62901) loading module /usr/local/lib/ser/modules/tm.so 0(62901) loading module /usr/local/lib/ser/modules/rr.so 0(62901) loading module /usr/local/lib/ser/modules/maxfwd.so 0(62901) loading module /usr/local/lib/ser/modules/usrloc.so 0(62901) loading module /usr/local/lib/ser/modules/registrar.so 0(62901) find_param_export: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(62901) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: <rewriteFromRoute> not found 0(62901) parse error (80,19-20): unknown command, missing loadmodule? 0(62901) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(62901) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(62901) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(62901) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(62901) INFO: signal 15 received 0(62901) Memory status (pkg): 0(62901) qm_status (0x80a1780): 0(62901) heap size= 1046128 0(62901) used= 0, used+overhead=2448, free=1043680 0(62901) max used (+overhead)= 2448 0(62901) dumping all allocked. fragments: 0(62901) dumping free list stats : 0(62901) hash= 137. fragments no.: 1 0(62901) ----------------------------- --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

2 2

[Serusers] Some compile time errors for FreeBSD of CVS Tree source (sip_router)
by Lenny Tropiano 20 Feb '03

20 Feb '03

pass_fd.c: added #include <sys/uio.h> tcp_main.c: moved #include <sys/select.h> underneath #include <sys/types.h.> Still getting this: $ gmake Makefile.defs:331: Old gcc detected (2.9x), use gcc >= 3.1 for better results gcc -g -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.11pre6-tcp5-tm"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 2.95"' -D__CPU_i386 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DVOICE_MAIL -D_TOTAG -DUSE_TCP -DFAST_LOCK -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -c tcp_main.c -o tcp_main.o tcp_main.c: In function `tcp_send': tcp_main.c:351: `MSG_NOSIGNAL' undeclared (first use in this function) tcp_main.c:351: (Each undeclared identifier is reported only once tcp_main.c:351: for each function it appears in.) gmake: *** [tcp_main.o] Error 1 Cannot find proper #define for MSG_NOSIGNAL The flags, according to "send()" man page is: The flags parameter may include one or more of the following: #define MSG_OOB 0x1 /* process out-of-band data */ #define MSG_PEEK 0x2 /* peek at incoming message */ #define MSG_DONTROUTE 0x4 /* bypass routing, use direct interface */ #define MSG_EOR 0x8 /* data completes record */ #define MSG_EOF 0x100 /* data completes transaction */ The flag MSG_OOB is used to send ``out-of-band'' data on sockets that support this notion (e.g. SOCK_STREAM); the underlying protocol must also support ``out-of-band'' data. MSG_EOR is used to indicate a record mark for protocols which support the concept. MSG_EOF requests that the sender side of a socket be shut down, and that an appropriate indication be sent at the end of the specified data; this flag is only implemented for SOCK_STREAM sockets in the PF_INET protocol family, and is used to implement Transaction TCP (see ttcp(4)). MSG_DONTROUTE is usually used only by diagnostic or routing programs. --- Lenny Tropiano E-mail: lenny(a)voiping.com Partner, Networking Specialist Pager: pager-lenny(a)voiping.com VoIPing, LLC URL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]

2 1

[Serusers] Via tags question...
by lgfausak＠august.net 20 Feb '03

20 Feb '03

This is a protocol question, I don't know if it is directly related to ser. Concerning the Via tags, should any proxy/gateway modify tags that don't belong to it, or should they simply be copied verbatum? For example, I have a situation where I send an INVITE message, and the TRYING response I receive modifies one of the Via: tags. I don't think this is right. Here are the two packets in question: # U 2003/02/19 18:36:20.276844 164.9.42.25:5060 -> 184.294.324.2:5060 INVITE sip:12143357976@184.294.324.2:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:12143357976@164.9.42.25;branch=0>. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. CSeq: 2 INVITE. Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Contact: <sip:8178863582@216.87.145.22:4060>. User-Agent: FXS_GW (sip1028a.bin). Content-Type: application/SDP. Content-Length:291. P-hint: LD GATEWAY. . v=0. o=FXS_GW 12367 0 IN IP4 216.87.145.22. s=Audio Session. i=Audio Session. c=IN IP4 216.87.145.22. t=0 0. m=audio 16384 RTP/AVP 0 18 4 8 96. a=fmtp:96 0-11. a=rtpmap:0 PCMU/8000/1. a=rtpmap:18 G729/8000/1. a=rtpmap:4 G723/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:96 telephone-event/8000. And your response is: # U 2003/02/19 18:36:20.902858 184.294.324.2:5060 -> 164.9.42.25:5060 SIP/2.0 100 Trying . Via: SIP/2.0/UDP 164.9.42.25;branch=z9hG4bKcc26.d3c125d.0. Via: SIP/2.0/UDP 216.87.145.22:4060. From: <sip:8178863582@216.87.145.22> ;tag=d8579116-fdc-3e542303-23d34-4a84. To: <sip:12143357976@augustvoice.net> ;user=phone;tag=27DC1DCC-F61. Date: Thu, 20 Feb 2003 00:36:20 GMT. Call-ID: 59fd88-d8579116-fdc-3e542303-23d2f-3897(a)216.87.145.22. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Content-Length: 0. . See how the : Via: SIP/2.0/UDP 216.87.145.22:4060 ;branch=z9hG4bK-3e542304-23dfc-ea2. Has been changed to: Via: SIP/2.0/UDP 216.87.145.22:4060. Does anyone have any perl script to take a ngrep protocol dump and print it out in a pretty way? ---greg Greg Fausak August.Net Services, LLC

2 1

[Serusers] Add a user to subscriber table.
by Ivan Vignola 20 Feb '03

20 Feb '03

Hi, I would add a user to subsriber table by sql command in my application (not using serctl script). What text strings I must use to calculate HA1 (=phplib_id) and ha1b values if USER_ID=ivan,PASSWORD=ivanpswd and REALM=192.168.11.237 ? Thanks in advance. Regards, Ivan Vignola. _________________________________________________________________ Vinci la nuova Nissan Micra con MSN Messenger! http://www.msn.it/messenger/

3 2

[Serusers] serctl - tail problems on solaris8?
by Kamil Ince 20 Feb '03

20 Feb '03

bet i missed reading something, again :-/ but wasting too much time on this.. i get on solaris 8 the following: # echo $SIP_DOMAIN myser.my.net # echo $PW heslo # serctl add user1 hispw user1(a)my.net usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] new user added # serctl passwd user1 newpw usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] non-existent user of course the user has been added to the database... anyone came accross that before? tia -k

3 5

[Serusers] [bug] TM module config
by Sebastien Pierre 19 Feb '03

19 Feb '03

Hi again, Here is how to get a segfault: <snip> #Load modules required by a stateful User Agent Service loadmodule "modules/tm.so" #SER timer parameters modparam("tm", "fr_inv_timer", "10"); modparam("tm", "fr_timer", "10"); </snip> This happened after quoting the numbers in the modparam. Tested with ser-0.8.10 on Linux RH 8.0 -- Sebastien

2 1

[Serusers] Logging with SER
by Sebastien Pierre 19 Feb '03

19 Feb '03

Hi all, I finally found solutions to my problem: 1) It is easier when setting "fork=no" because then ser does not daemonise and stderr is then attached to the terminal in which I started ser 2) The configuration file should set a debugging level ("set debug=1"), and use a number equal or lower to the debug level in the log, like 'log(0, "Hello World")' So my HelloWorld.cfg is as follows: <snip> debug=0 log_stderror=yes fork=no route[0] { log(0, "Hello World!"); } </snip> And it works with "ser -f HelloWorld.cfg" Cheers, -- Sebastien

1 0

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