sr-users February 2012

sr-users@lists.kamailio.org

79 participants
139 discussions

by Fabian Borot

Hello We need to implement this scenario: client -> proxy --> provider The provider sends 183 with SDP and we need to drop it and send 180 instead. I managed to do it like this: 1- on_reply block, I catch the 183 if it comes, call a route block named "DROP_180" then drop it: if ($rs == "183") { route(DROP_180); drop(); } 2- on the new route block "DROP_180" I use the command send_reply(180, "Ringing") route[DROP_180] { xlog("L_INFO","mylog: On Route DROP_180."); send_reply(180,"Ringing"); } I had to do it on a separate block because the send_reply command could not be used on the same on_reply block, also I had to send the 180 first before the 183 is dropped because the drop() apparently stops the script execution and the 180 was not sent The idea works fine, BUT the To tag on the locally generated 180 is not the same as the one on the original 183 message. After the progress (183 message), it comes the 200 OK, which we forward downstream unaltered. Now, the client seems to ignore the 200 OK, (there is no ACK), we think that since the To tag on the original 183 and the 200 OK does not match the one on the 180 that is why the customer is processing the 200 OK. Any ideas here? txs a lot

12 years, 10 months

XMLOPS xpath issues

by Olle E. Johansson

Friends, I'm having issues with the xpath support. If I run Daniel's example in the XMLOPS, xpath works as documented. But if I take the body of a PUBLISH from the and run xpath, I don't get expected results. Now, I'm no XPATH guru so I may be totally off the markup here... Here's the test script: $xml(pub=>doc) = $rb; $var(contact)=$xml(pub=>xpath:registration/contact@event); xlog("--- Publish contact $var(contact) \n"); xlog("--- Publish aor $xml(pub=>xpath:registration@aor) \n"); xlog("--- Publish aor: $xml(pub=>xpath://@aor) \n"); xlog("--- Publish reg state: $xml(pub=>xpath://reginfo@state) \n"); xlog("--- Publish uri: $xml(pub=>xpath:contact/uri/text()) \n"); I've tested with various paths, orginating from /reginfo or just picking an attribute. Here's the XML from Carsten's PUA_REGINFO module: <?xml version="1.0"?> <reginfo xmlns="urn:ietf:params:xml:ns:reginfo" version="0" state="full"> <registration aor="sip:oej@testnamn.se" id="0x80590a590" state="terminated"/> </reginfo> I wonder if it's something with using the request body that doesn't parse properly in the XMLOPS module? Error messages are a series of "XPath error : Invalid expression" /O

12 years, 10 months

Using Voicemail strange behavior - git head 3.3

by David

I was testing the head version in GIT; I ran across a strange issue. The new config includes a "WITH_VOICEMAIL" definition I am not sure exactly how it is designed to work, but by default if the "usrloc" fails it creates a branch and goes to voicemail That part work correctly. It seems however if a phone is registered and the you get a busy, the failure route creates a branch but also retries the original RURI but I dont see a fork being created for the new branch to the voicemail server. Seems to just retry the original RURI. I am using the default config from the kamailio. repo. Any ideas. Thanks.

12 years, 10 months

change DBURL without restarting kamailio?

by Uri Shacked

Hi, Is there a way to set a DBURL as a parametes that i can change in run time without restarting kamailio? It is usfull if i have a backup data base and i dont want to use mysql clustering... Cheers, Uri

12 years, 10 months

CSEQ problem with SPA504G

by David

Hello, I am trying to connect to a kamailio server using a Cisco SPA504G phone using TCP instead of TCP. When in UDP, it works fine, but in TCP I get errors relating to CSEQ. I included the trace taking with ngrep after the email. I don't understand the CSeq error. I had a look at RFC3261, it says that the CSEQ number has to increase and that's what it has done. Can anyone tell me where the errorlies ? I tried changeing the check sanity to not check that CSeq is valid, but it screws up authentication as it mis detects the method type. Please advise, David interface: eth0 (SERVERIP/255.255.255.128) filter: (ip or ip6) and ( port 5060 and host 206.248.130.233 #### T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-b03863b5;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. Max-Forwards: 70. Contact: "104 Zone T" <sip:testzone.104@192.168.1.225:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 192.168.1.225:5075;received=206.248.130.233;branch=z9hG4bK-b03863b5;rport=5075. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=a889ade7c63805944f01d11bb71a916b.6a29. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. WWW-Authenticate: Digest realm="SERVERNAME.TLD", nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7". Server: OmniVigil 5.0a. Content-Length: 0. . ## T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Max-Forwards: 70. Authorization: Digest username="testzone.104",realm="SERVERNAME.TLD",nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7",uri="sip:SERVERNAME.TLD",algorithm=MD5,response="77ddf2a1baafd33ed0c675fdc074bc03". Contact: "104 Zone T" <sip:testzone.104@206.248.130.233:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 400 CSeq method does not match request method. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport=5075;received=206.248.130.233. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=1feca66ab82b1f32a901c4cd8a4b7a05.e8e6. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Server: kamailio (3.1.5 (x86_64/linux)). Content-Length: 0. .

12 years, 10 months

How to do accounting for Kamailio-generated BYEs?

by Yufei Tao

Hi I've got a mediaproxy that detects no media, and sends dlg_end_dlg to Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling parties. It all works fine. But my question is, how to do accounting for these Kamailio-generated BYEs? They don't appear in the acc table. I'm using Kamailio 3.1.5. Thank you very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

12 years, 10 months

CallControl and MediaProxy

by Reda Aouad

Hi, When I use the function call_control( ) of the call_control module, it automatically engages mediaproxy if it finds the mediaproxy module loaded. If the mediaproxy module is not loaded, call_control doesn't even try to engage it. I need mediaproxy for NAT traversal in some cases, but don't want it to be engaged on every call. How can I disable this behavior? Thanks Reda

12 years, 10 months

Social networking event, London, March 6, 2012

by Daniel-Constantin Mierla

Hello, we are already several folks planing to go out after the first day of UC Expo and continue discussions about latest news of Kamailio/SER project and VoIP/Telecomunications. If you want to join, drop me a message to send you the details of the location. Looking forward to meeting many of you next week, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

12 years, 10 months

Re: [SR-Users] How to do request URI rewrites using database tables

by Javier Gallart

Hello Ben dialplan module should help you: http://kamailio.org/docs/modules/3.2.x/modules/dialplan.html Regards Javi > ------------------------------ > > Message: 5 > Date: Fri, 17 Feb 2012 12:49:04 +1300 > From: Ben WIlliams <benwilliams(a)joobworld.com> > Subject: Re: [SR-Users] How to do request URI rewrites using database > tables? > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANfYGQkeK3i=vBbdox+ekGeeKmjWhhgarUJcAo3oV=rrO=Wwig(a)mail.gmail.com > > > Content-Type: text/plain; charset=ISO-8859-1 > > I've managed to get dbaliases and lcr to do most of this. But the lcr > prefix does not allow regular expressions. Is there any other module > that allows you to store in the database a regular expression rewrite > rule? > > On Thu, Feb 16, 2012 at 1:21 PM, Ben WIlliams <benwilliams(a)joobworld.com> > wrote: > > Hi, can someone please recommend the most appropriate modules to > > rewrite R-URIs based on a database lookup table? I've read the > > documentation for lcr and carrierroute but not sure if they can do > > this. > > > > In most cases it will be a simple R-URI rewrite but I also need to > > rewrite based on From user. > > > > ie > > > > R-URI match ? ? ? ? ? ? From match ? ? ? ? ? ? ? ? ?new R-URI > > =========== ? ? ? ? ? ? ========== ? ? ? ? ? ? ? ? ?========= > > *97(a)example.com ? ? ? ? 6[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.1 > > *97(a)example.com ? ? ? ? 7[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.2 > > 6[0-9](a)example.com ? ? ?match any ? ? ? ? ? ? ? ? ? rewrite domain to > 10.0.0.1 > > abc(a)example.com ? ? ? ? match any ? ? ? ? ? ? ? ? ? def(a)example.com > > > > Thanks > > Ben > > > >

12 years, 11 months

Have-my-domain! - host your own SIP domain at iptel.org

by SuperCel.mx.tc

Sirs: We cannot get our domain to configure to the iptel SIP Service (Have-my-domain! - host your own SIP domain at iptel.org). I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank, CTO SuperCel.tel

12 years, 11 months

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sr-users February 2012