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37471 discussions

[Kamailio-Users] how can I add .sar file as application to Wesip application server
by skumar453＠gmail.com 07 Jan '09

07 Jan '09

hi all I am workin on the "WeSIP" application server. I installed opensips as proxy with SEAS module and wesip as Application server. these two are successfully connected ...but here i have no clue on how to add application "click2dial" to WeSIP. 1. please let me know how can I add .sar file as application. 2. http://localhost:5075/manager/html? is showig a blank page. how can I get this page displayed? regards suresh

2 1

[Kamailio-Users] moving my kamilio1.2 server to opensips
by Asim Riaz 07 Jan '09

07 Jan '09

Hi, could anyone advice me, I like to move my Kamalio 1.2 server to Opensips ? is it possible to move all the exisitng configuration transparently? Thanks in Advance. -Asim

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[Kamailio-Users] Regarding 600 bussy every where when i Forward INVITe message from b2bua to scscf
by sureshkumar.jaiswal 07 Jan '09

07 Jan '09

Hi when i ran b2bua_simple.py like this python b2bua_simple.py -f -l 172.18.2.187 -p 99999 -n 172.18.2.187 b2bua_simple.py is running 172.18.2.187 and port for this is 99999 and scscf is also running 172.18.2.187:6060 and forward my INVITE message from scscf to b2bua_simple then b2bua got this INVITE message but when b2bua again forward to scscf then we getting 600 bussy every where and call has been terminated. folllowing message i have attache along this mail. can any one let me know what i have to do for successfull call using b2bua_simple.py. python b2bua_simple.py -f -l 172.18.2.187 -p 99999 -n 172.18.2.187 08 Jan 15:26:22/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:6060: INVITE sip:alice@info-spectrum.com SIP/2.0 Record-Route: <sip:mo@scscf.info-spectrum.com:6060;lr> Route: <sip:172.18.2.187:99999;lr>, <sip:iscmark@scscf.info-spectrum.com:6060;lr;s=1;h=0;d=0;a=7369703a626f6240696e666f2d737065637472756d2e636f6d> Record-Route: <sip:mo@pcscf.info-spectrum.com:4060;lr> Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 Via: SIP/2.0/UDP 172.18.2.187:4060;branch=z9hG4bK2ede.baaf9433.0 Via: SIP/2.0/UDP 172.18.2.74:2423;rport=2423;branch=z9hG4bK32209 From: <sip:bob@info-spectrum.com>;tag=9791 To: <sip:alice@info-spectrum.com> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 201 INVITE Max-Forwards: 15 Allow: INVITE,ACK,CANCEL,BYE,MESSAGE,OPTIONS,UPDATE,NOTIFY,PRACK Contact: <sip:bob@172.18.2.74:2423;transport=udp>;expires=1000 User-Agent: Mercuro IMS Client Beta (4.0.1011.0) Supported: sdp-anat Content-Type: application/sdp P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=00000000 Privacy: none Content-Length: 468 P-Asserted-Identity: <sip:bob@info-spectrum.com> P-Charging-Vector: icid-value="P-CSCFabcd4965cdc6000000ea";icid-generated-at=172.18.2.187;orig-ioi="info-spectrum.com" Timestamp: 1231408582 v=0 o=- 3344 3344 IN IP4 172.18.2.74 s=Mercuro IMS Client Session t=0 0 m=audio 30904 RTP/AVP 0 8 101 c=IN IP4 172.18.2.74 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 30080 RTP/AVP 111 34 31 c=IN IP4 172.18.2.74 a=rtpmap:111 H263-1998/90000 a=fmtp:111 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=2;QCIF=2/MaxBR=2400 08 Jan 15:26:22/GLOBAL/b2bua: SENDING message to 172.18.2.187:6060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 Via: SIP/2.0/UDP 172.18.2.187:4060;branch=z9hG4bK2ede.baaf9433.0 Via: SIP/2.0/UDP 172.18.2.74:2423;rport=2423;branch=z9hG4bK32209 Record-Route: <sip:mo@scscf.info-spectrum.com:6060;lr> Record-Route: <sip:mo@pcscf.info-spectrum.com:4060;lr> From: <sip:bob@info-spectrum.com>;tag=9791 To: <sip:alice@info-spectrum.com> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 201 INVITE Server: Sippy B2BUA (Simple) 08 Jan 15:26:22/GLOBAL/b2bua: SENDING message to 172.18.2.187:5060: INVITE sip:alice@172.18.2.187:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.2.187:99999;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18;rport Max-Forwards: 70 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 INVITE Contact: Anonymous <sip:bob@172.18.2.187:99999> Expires: 300 User-Agent: Sippy B2BUA (Simple) cisco-GUID: 288073477-2250398923-1693218158-3417804826 h323-conf-id: 288073477-2250398923-1693218158-3417804826 Content-Length: 468 Content-Type: application/sdp v=0 o=- 3344 3344 IN IP4 172.18.2.74 s=Mercuro IMS Client Session t=0 0 m=audio 30904 RTP/AVP 0 8 101 c=IN IP4 172.18.2.74 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 30080 RTP/AVP 111 34 31 c=IN IP4 172.18.2.74 a=rtpmap:111 H263-1998/90000 a=fmtp:111 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2;QCIF=2/MaxBR=2400 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=2;QCIF=2/MaxBR=2400 08 Jan 15:26:23/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.18.2.187:99999;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18;rport=34463 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187> Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 INVITE Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux)) Content-Length: 0 Timestamp: 1231408582 Warning: 392 172.18.2.187:5060 "Noisy feedback tells: pid=3661 req_src_ip=172.18.2.187 req_src_port=34463 in_uri=sip:alice@172.18.2.187:5060 out_uri=sip:alice@172.18.2.187:5060 via_cnt==1" 08 Jan 15:26:33/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:5060: SIP/2.0 600 Busy everywhere - Forwarding to S-CSCF failed Via: SIP/2.0/UDP 172.18.2.187:99999;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18;rport=34463 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187>;tag=0fb5d9176cb7d6a1cb577893687b8ec9-47d1 Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 INVITE Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux)) Content-Length: 0 Timestamp: 1231408593 Warning: 392 172.18.2.187:5060 "Noisy feedback tells: pid=3664 req_src_ip=172.18.2.187 req_src_port=5060 in_uri=sip:alice@172.18.2.187:5060 out_uri=sip:alice@172.18.2.187:5060 via_cnt==0" 08 Jan 15:26:33/GLOBAL/b2bua: SENDING message to 172.18.2.187:5060: ACK sip:alice@172.18.2.187:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.2.187:99999;rport;branch=z9hG4bK8801d0c21346d39cb48f86a2abbedc18 Max-Forwards: 70 From: <sip:bob@172.18.2.187>;tag=7cb0f980e37d8198e1c1dfeb4af2e531 To: <sip:alice@172.18.2.187>;tag=0fb5d9176cb7d6a1cb577893687b8ec9-47d1 Call-ID: 1231260517025(a)172.18.2.74 CSeq: 200 ACK Expires: 300 User-Agent: Sippy B2BUA (Simple) 08 Jan 15:26:33/GLOBAL/b2bua: SENDING message to 172.18.2.187:6060: SIP/2.0 600 Busy everywhere - Forwarding to S-CSCF failed Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 Via: SIP/2.0/UDP 172.18.2.187:4060;branch=z9hG4bK2ede.baaf9433.0 Via: SIP/2.0/UDP 172.18.2.74:2423;rport=2423;branch=z9hG4bK32209 Record-Route: <sip:mo@scscf.info-spectrum.com:6060;lr> Record-Route: <sip:mo@pcscf.info-spectrum.com:4060;lr> From: <sip:bob@info-spectrum.com>;tag=9791 To: <sip:alice@info-spectrum.com>;tag=2ac46b4f62d11be3189628535aa7b381 Call-ID: 1231260517025(a)172.18.2.74 CSeq: 201 INVITE Server: Sippy B2BUA (Simple) 08 Jan 15:26:33/GLOBAL/b2bua: RECEIVED message from 172.18.2.187:6060: ACK sip:alice@info-spectrum.com SIP/2.0 Via: SIP/2.0/UDP 172.18.2.187:6060;branch=z9hG4bK2ede.b679b107.0 From: <sip:bob@info-spectrum.com>;tag=9791 Call-ID: 1231260517025(a)172.18.2.74 To: <sip:alice@info-spectrum.com>;tag=2ac46b4f62d11be3189628535aa7b381 CSeq: 201 ACK Route: <sip:orig@scscf.info-spectrum.com:6060;lr> User-Agent: Sip EXpress router(2.1.0-dev1 OpenIMSCore (i386/linux)) Content-Length: 0

2 1

[Serusers] Large MTU Size of SIP Invites
by Jai Rangi 06 Jan '09

06 Jan '09

All, Greeting, I am using ser as a proxy to forward calls to Asterisk servers. When I forward the calls the total packet size is anywhere between 1505 to 1520. So the packets are frangmented. Some asterisks behind the firewall are not able to handle the franented sip packets of size 1500+. Is there a way to reduce the packets size to less than 1500. Any hint will be be appreciated. Thank you, Regards, -jai

2 1

[Kamailio-Users] dialplan and carrierroute
by victorpy 05 Jan '09

05 Jan '09

-- View this message in context: http://www.nabble.com/dialplan-and-carrierroute-tp21299057p21299057.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] extending carrierroute to load dynamic carriers
by victorpy 05 Jan '09

05 Jan '09

Hello, I want to extend the module to load a carrier tree based on $si(source ip) of a request. I want know if im going the right way doing this. The reason why im doing this is to avoid a hardcoded script. I have read other post that talk about this subject and the solution with domains and host doesn't meet my needs. Hope someone could help me. thanks for the answers greetings -- View this message in context: http://www.nabble.com/extending-carrierroute-to-load-dynamic-carriers-tp212… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 5

[Kamailio-Users] Setting answer timeout
by mayamatakeshi 05 Jan '09

05 Jan '09

Hello, I know I can customize the timeout for answer using fr_inv_timer: modparam("tm", "fr_inv_timer", 20) But is it possible to set the answer timeout on a transaction basis? (so that it can be customized by subscriber for example). regards, mayama

2 2

[Kamailio-Users] Owner of rtpproxy.sock
by Franz Edler 05 Jan '09

05 Jan '09

Hello experts! I have a problem with the owner of rtpproxy.sock. When I use the -u parameter at start of rtpproxy like "rtpproxy -s unix:/var/run/rtpproxy.sock -u sip:sip" the owner and group of the socket is still "root root". Is this correct? Should not it be sip sip? I want to avoid kamailio running as root and therefore I have to modify the permissions of rtpproxy.sock after start of rtpproxy accordingly. My question now is: Maybe I have a misunderstanding in the purpose of the -u parameter. What is the behaviour of the -u parameter? Can anyone shed some light on this? Regards Franz

2 4

[Kamailio-Users] Bring India together
by Sunkara RaviPrakash 04 Jan '09

04 Jan '09

1 0

[Kamailio-Users] Bring India together
by Sunkara RaviPrakash 04 Jan '09

04 Jan '09

1 0

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