sr-users December 2008

sr-users@lists.kamailio.org

73 participants
107 discussions

[Kamailio-Users] Is there a way to detect use_media_proxy called or not?
by Krunal Patel 18 Dec '08

18 Dec '08

Hi, Does anybody know how to detect whether use_media_proxy is called? What I want to do is: Do end_media_session for bye or cancel if & only if use_media_proxy has been called during invite or reply route. Is it possible? If it is then please let me know how to implement it? Thanks in advance -- Krunal Patel

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[Serusers] SER with FreeRADIUS
by Robert Borz 18 Dec '08

18 Dec '08

Hi guys, we're using ser 0.9.7 on Debian/Lenny together with FreeRADIUS. Authentication against the RADIUS already works basically for our sip clients. Now we want to make the contact information persisting restarts of SER. Is there a way to store the contact information for the registered sip clients in the radius database or must an additional database, e.g. mysql or postgres, used for that? Regards, Robert ____________________________________________________________________ Psssst! Schon vom neuen WEB.DE MultiMessenger gehört? Der kann`s mit allen: http://www.produkte.web.de/messenger/?did=3123

1 0

Re: [Kamailio-Users] openser and asterisk auth problem.
by BERGANZ François 18 Dec '08

18 Dec '08

I want to authenticate in openser some users which are in asterisk conf (to do load balancing with another asterisk). If I delete the user from the asterisk config and insecure=invite for the openser, asterisk accept the call, but, I want that asterisk see that it is that user which is doing the call. Cordialement, BERGANZ François http://www.acropolistelecom.net  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Andrew V.Statsenko [mailto:alter@siptalk.ru] Envoyé : jeudi 18 décembre 2008 11:00 À : BERGANZ François Objet : ***SPAM*** Re: [Kamailio-Users] openser and asterisk auth problem. В Чтв, 18/12/2008 в 10:31 +0100, BERGANZ François пишет: > -----I have that error: > <--- SIP read from UDP://192.168.1.156:5060 ---> > INVITE sip:33662971130@192.168.1.156 SIP/2.0 > Record-Route: <sip:192.168.1.156;lr=on;ftag=0c46027a> > Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0 > Via: SIP/2.0/UDP 192.168.1.60:43490;branch=z9hG4bK-d8754z-c15d2350d0742a11-1---d8754z-;rport=43490 > Max-Forwards: 70 > Contact: <sip:6833211245@192.168.1.60:43490> > To: "33662971130"<sip:33662971130@192.168.1.156> > From: "6833211245"<sip:6833211245@192.168.1.156>;tag=0c46027a > Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc. > CSeq: 2 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > Content-Type: application/sdp > User-Agent: X-Lite release 1100l stamp 47546 > Content-Length: 231 [..] > <-------------> > --- (14 headers 10 lines) --- > Sending to 192.168.1.156 : 5060 (no NAT) > Using INVITE request as basis request - YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc. > Found user '6833211245' for '6833211245' This is the key trouble in your configuration: Asterisk trying to authenticate this call as call from _user_ (INVITE->401->INVITE+auth), but not as _peer_. Will you describe us where you want to authenticate you users : - at the OpenSER frontend (BTW, I don't see auth headers in first INVITE) - or in the Asterisk backend ? Which service logic you are building ? -- WBR, Alter

3 4

Re: [Kamailio-Users] serMyAdmin database connection
by Flavio Goncalves 18 Dec '08

18 Dec '08

Message: 1 Date: Wed, 17 Dec 2008 16:29:56 +0200 From: TCB <tawandac(a)gmail.com> Subject: [Kamailio-Users] serMyAdmin error To: users <users(a)lists.kamailio.net> Message-ID: <f9db9e000812170629v34e6e1edr2a472912427c78eb(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" I'm trying to install serMyAdmin using a remote mysql DB. I keep getting the following error. Caused by: org.springframework.jdbc.CannotGetJdbcConnectionException: Could not get JDBC Connection; nested exception is com.mysql.jdbc.exceptions.jdbc4.CommunicationsException: Communications link failure I changed context.xml to point to the remote DB. Where else should i change DB the credential -- TC Please try to connect to your database using mysql command line, to check if your permissions are ok. I will verify the JDBC connection with a remote database. Regards, Flavio E. Goncalves

1 0

[Kamailio-Users] openser and asterisk auth problem.
by BERGANZ François 18 Dec '08

18 Dec '08

Hello, I have : ASTERISK-----OPENSER-----SOFTPHONE My softphone auth to openser (with username/password of asterisk sync with a database) Openser forward the INVITE to asterisk and asterisk return Unautorize! I tried to have the same realm to try de have the same auth for asterisk and openser But without success. In fact, I need to auth to openser(with asterisk config user in the database) and forward to asterisk the INVITE. Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 2

[Kamailio-Users] Use Carrierroute to rewrite from_uri based on prefix table
by rod 18 Dec '08

18 Dec '08

Hi all, I've been able to rewrite R_URI based on prefix matching in the carrierroute module, using: cr_route("default", "0", "$rU", "$rU", "call_id") I used this to normalize the ruri to E164: ex: in france, the country code is +33, if someone dials "01 10 10 10 10", I strip the first 0 and prefix with 33 so that I get 331 10 10 10 10 (space are there for convenience :p) What I'd like to do is to use the carrierroute module to do the same but for from uri: ex: I receive a call on my phone with a caller using e164 (from_uri: 331 20 20 20 20) and I'd like to present this number as 01 20 20 20 20 to the callee. I added a prefix 331 in the carrierroute tables. I tried this: - cr_route("default", "0", "$fU", "$fU", "call_id") - cr_route("default", "0", "$fU", "$fU", "from_uri") - cr_route("default", "0", "$fU", "$rU", "call_id") - cr_route("default", "0", "$fU", "$rU", "from_uri") but I've been unsuccessful with this. Is there a way to achieve from_uri translation using the carrierroute module. Regards, rod

2 2

[Kamailio-Users] nat problems
by Ricky Gutierrez 17 Dec '08

17 Dec '08

Hi list, I want to comment some problems of nat that I have, I have two connected remote UAC in a same location UAC 202 REMOTE A UAC 123 REMOTE A if the UAC 123 calls to UAC 202 there is not bidirectional audio UAC 204 REMOTE B UAC 122 REMOTE C if the UAC 122 calls to UAC 202 if there is bidirectional audio some help like I can solve this problem?, the remote clients in site A they are behind a nat symetrics I am using rtpproxy version 1.1 and openser 1.3.2 log sip U +2.975758 190.212.49.86:58944 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152:5060;branch=z9hG4bK74f3.13c6ec23.0 Via: SIP/2.0/UDP 192.168.10.37:5060;rport=63451;received=190.212.49.86;branch=z9hG4bK33571797d7d31cb0 Record-Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:202@192.168.10.40:5064;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=202 8000 8000 IN IP4 192.168.10.40 s=SIP Call c=IN IP4 192.168.10.40 t=0 0 m=audio 5004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000475 190.184.22.152:5060 -> 190.212.49.86:63451 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.37:5060;rport=63451;received=190.212.49.86;branch=z9hG4bK33571797d7d31cb0 Record-Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:202@190.212.49.86:58944;transport=udp;nat=yes> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 234 P-hint: onreply_route|force_rtp_proxy P-hint: Onreply-route - fixcontact v=0 o=202 8000 8000 IN IP4 192.168.10.40 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35036 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=nortpproxy:yes # U +0.241543 190.212.49.86:63451 -> 190.184.22.152:5060 ACK sip:202@190.212.49.86:58944;transport=udp;nat=yes SIP/2.0 Via: SIP/2.0/UDP 192.168.10.37:5060;branch=z9hG4bKa26522558c582a68 Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Contact: <sip:123@192.168.10.37:5060;transport=udp> Supported: path Proxy-Authorization: Digest username="123", realm="190.184.22.152", algorithm=MD5, uri="sip:202@190.184.22.152", nonce="49485f8958ba6dc4e39ee5893d15ea3e60ae19ad", response="afb5db07f3a90d4cfc6ea483fd120404" Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 ACK User-Agent: Grandstream BT200 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.000334 190.184.22.152:5060 -> 190.212.49.86:58944 ACK sip:202@190.212.49.86:58944;transport=udp SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=0966ab4346392f45> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK74f3.13c6ec23.2 Via: SIP/2.0/UDP 192.168.10.37:5060;rport=63451;received=190.212.49.86;branch=z9hG4bKa26522558c582a68 From: "ricardo-elim" <sip:123@190.184.22.152>;tag=0966ab4346392f45 To: <sip:202@190.184.22.152>;tag=15c74eb7e1a2ae92 Contact: <sip:123@190.212.49.86:63451;transport=udp;nat=yes> Supported: path Proxy-Authorization: Digest username="123", realm="190.184.22.152", algorithm=MD5, uri="sip:202@190.184.22.152", nonce="49485f8958ba6dc4e39ee5893d15ea3e60ae19ad", response="afb5db07f3a90d4cfc6ea483fd120404" Call-ID: 4dfa1b96e314c165(a)192.168.10.37 CSeq: 17345 ACK User-Agent: Grandstream BT200 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.383479 190.184.22.152:5060 -> 190.212.49.86:58944 OPTIONS sip:190.212.49.86:58944 SIP/2.0 Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=d7259e51 To: sip:190.212.49.86:58944 Call-ID: 1f9f65c1-fcccdef2-674a(a)190.184.22.152 CSeq: 1 OPTIONS Content-Length: 0 # U +0.173847 190.212.49.86:58944 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=d7259e51 To: sip:190.212.49.86:58944;tag=15c74eb7e1a2ae92 Call-ID: 1f9f65c1-fcccdef2-674a(a)190.184.22.152 CSeq: 1 OPTIONS User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:202@192.168.10.40:5064;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 # U +0.826048 190.184.22.152:5060 -> 190.212.35.160:50273 OPTIONS sip:190.212.35.160:50273 SIP/2.0 Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=e7259e51 To: sip:190.212.35.160:50273 Call-ID: 1f9f65c1-0dccdef2-774a(a)190.184.22.152 CSeq: 1 OPTIONS Content-Length: 0 # U +0.192726 190.212.35.160:50273 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152:5060;branch=0 From: sip:pinger@190.184.22.152;tag=e7259e51 To: sip:190.212.35.160:50273;tag=c2c330e1c7bae7ba Call-ID: 1f9f65c1-0dccdef2-774a(a)190.184.22.152 CSeq: 1 OPTIONS User-Agent: Grandstream GXP2000 1.1.6.16 Contact: <sip:203@192.168.0.16:5060;transport=udp;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 # U +1.807039 192.168.11.1:5060 -> 192.168.11.30:5060 OPTIONS sip:192.168.11.30:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.1:5060;branch=0 From: sip:pinger@190.184.22.152;tag=f7259e51 To: sip:192.168.11.30:5060 Call-ID: 1f9f65c1-1dccdef2-974a(a)192.168.11.1 CSeq: 1 OPTIONS Content-Length: 0 regards rickygm

2 1

[Kamailio-Users] simple call between 2phones
by BERGANZ François 17 Dec '08

17 Dec '08

Hello, I know kamailio, but, I dont know how to do : Call between 2phone which are registered ! Know you how to do ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 1

[Kamailio-Users] serMyAdmin error
by TCB 17 Dec '08

17 Dec '08

I'm trying to install serMyAdmin using a remote mysql DB. I keep getting the following error. Caused by: org.springframework.jdbc.CannotGetJdbcConnectionException: Could not get JDBC Connection; nested exception is com.mysql.jdbc.exceptions.jdbc4.CommunicationsException: Communications link failure I changed context.xml to point to the remote DB. Where else should i change DB the credential -- TC -- TC

1 0

[Kamailio-Users] failed to request mysql lock ()
by Brian Del Shasta 16 Dec '08

16 Dec '08

Is this an issue? Dec 16 14:47:45 fndipt03 cdrtool[10738]: Database error: failed to request mysql lock () Dec 16 14:47:45 fndipt03 cdrtool[10738]: Unlock ser_radius:radacct Dec 16 14:47:45 fndipt03 cdrtool[10738]: Error in unLockNormalization() The only problem I have is I am unable to get the pricing to work, but I think that may be unrelated... Thanks

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sr-users December 2008