sr-users

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14 participants
37465 discussions

[Users] Call a stored procedure
by Miroslaw Nawrot 24 Aug '06

24 Aug '06

Hi, Does someone have an axample of using avp_db_query() with a mysql stored procedure? I keep getting an error "submit_query: PROCEDURE openser.sp_getrouteinfo can't return a result set in the given context" whenever I pass a stored procedure call to that functions. Thanks. Mirek N Globalive Communications Corp. The Future of Voice T:416-640-1088 F:416-640-1089 60 Adelaide St E, 6th Floor Toronto, ON, M5C 3E4 www.globalive.com

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[Users] openser admin
by Mike Williams 24 Aug '06

24 Aug '06

All, OpenSER Admin version 0.3 has been released. This fixes the bug where compatibility with openserctl was broken, and adds the ability to add other management accounts to the interface. Let me know what you think. Thanks, Mike Williams

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[Users] How to change domain when REGISTERing UA?
by Dmitry Lyubimkov 24 Aug '06

24 Aug '06

Hi! Description of the problem: The some UA can send only IP addresses as domains. Some subscribers incorrectly configure their UA and as a result as the domain IP of SIP Proxy comes. In result in the table location the part of subscribers has the symbolical domain (voapp.ru) and part - IP address of SIP Proxy (62.33.22.14) It leads to to the certain complexities in switching incming calls. The problem disappears if to tell: modparam("usrloc", "use_domain", 0) modparam("registrar", "use_domain", 0) And use only names without domains. But in our case it does not approach since we have some independent domains. I tried to rewrite fields From and To route(8); save("location"); route[8] { #correct From and To in REGISTER messages with IP address instead of domain name if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) { subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr om:\1sip:\2@voapp.ru\7/i"); subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\ 1sip:\2@voapp.ru\7/i"); xlog("L_ERR", "REGISTER From To done\n"); return(-1); }; return(1); } Fields varied but function save took their not changed values from initial message. After that I have tried to send REGISTER message once again to myself for processing: if (!route(8)) { t_relay(); } else { save("location"); }; In this case in the location table there was a record with the corrected name of the domain. But in Contact field there was an address of SIP Proxy server instead of address of UA. Can anybody prompt idea how to to change domain in From To fields at reception of REGISTER message? Tnx. How to change domain when REGISTERing UA? Hi! Description of the problem: The some UA can send only IP addresses as domains. Some subscribers incorrectly configure their UA and as a result as the domain IP of SIP Proxy comes. In result in the table location the part of subscribers has the symbolical domain (voapp.ru) and part - IP address of SIP Proxy (62.33.22.14) It leads to to the certain complexities in switching incming calls. The problem disappears if to tell: modparam("usrloc", "use_domain", 0) modparam("registrar", "use_domain", 0) And use only names without domains. But in our case it does not approach since we have some independent domains. I tried to rewrite fields From and To route(8); save("location"); route[8] { #correct From and To in REGISTER messages with IP address instead of domain name if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) { subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr om:\1sip:\2@voapp.ru\7/i"); subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\ 1sip:\2@voapp.ru\7/i"); xlog("L_ERR", "REGISTER From To done\n"); return(-1); }; return(1); } Fields varied but function save took their not changed values from initial message. After that I have tried to send REGISTER message once again to myself for processing: if (!route(8)) { t_relay(); } else { save("location"); }; In this case in the location table there was a record with the corrected name of the domain. But in Contact field there was an address of SIP Proxy server instead of address of UA. Can anybody prompt idea how to to change domain in From To fields at reception of REGISTER message? Tnx. Dmitry

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

24 Aug '06

Thanks...found it 2 seconds after sending the email! :PDate: Thu, 24 Aug 2006 21:21:10 +0200From: greger(a)teigre.comTo: steve_helfen(a)hotmail.comCC: serusers(a)iptel.orgSubject: Re: [Serusers] Alternate port for rewritehost Use rewritehostport("domain:5060") g-) Stephen Helfen wrote: 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! Try it now! _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Check the weather nationwide with MSN Search: Try it now! http://search.msn.com/results.aspx?q=weather&FORM=WLMTAG

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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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