Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow the article located here:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
I've tried to add all the sections marked with WITH_FREESWITCH in the sample config in the article into my own kamailio-advanced.cfg file. Here's what my cfg file looks like: http://pastebin.com/KsvrYVN7
I've restarted kamailio after making these changes. Then I tried to dial 41 to listen to vmail or 44999 to leave a message for user 999 but both return a busy tone.
Any suggestions would be appreciated. Thanks.
Two updates:
1. the link i had was incorrect. this is the correct link to the "how to" that I'm following:
http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc 2. I've narrowed down the issue a little bit. Here's what I've found, along with some more background information:
a) i have two polycom phones on my network, ext 888 and 999. b) they both register fine to the sip proxy. (192.168.1.101)
c) when i try to call ext 888 from ext 999, via a tcpdump, i can see that the call makes it to the freeswitch server (192.169.1.111) Since ext 888 is online, it should have just directed the call to the phone (vs. going to vmail) but instead, I get the following error message:
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK9db9.11128f7.0 Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKe920d3d0328D8D59 Max-Forwards: 14 From: "999" sip:999@192.168.1.101;tag=11B3C4E2-A9A9E183 To: sip:888@192.168.1.101;user=phone;tag=vmNpFtt7t5t1D Call-ID: ff78246e-9eeda51f-a54bce3c@192.168.1.102 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140320T233219Z~dd242f3ba6~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "kb-888" <kb-888>;party=calling;privacy=off;screen=no
In debugging on freeswitch I can see that it tries to match a dialplan for kb-888 and then ends up attempting to do a enum look up on "kb-888". Then it says that it has completed the dialplan.
I don't have an extension kb-888 registered. I can see where in the configs that I am prefixing "kb" to the calls on the kamailio side. And on the freeswitch side of the house, I see where I a regular expression looking for this prefix. But I don't know how i can get freeswitch to send the call to 888@192.168.1.101
Here's some of the debug data from freeswitch:
2014-03-26 11:04:07.345480 [DEBUG] switch_ivr.c:1830 (sofia/external/999@192.168.1.101) State Change CS_EXECUTE -> CS_ROUTING 2014-03-26 11:04:07.345480 [DEBUG] switch_core_session.c:1385 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.345480 [DEBUG] switch_core_session.c:905 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.345480 [NOTICE] switch_ivr.c:1837 Transfer sofia/external/999@192.168.1.101 to enum[kb-888@default] 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:530 (sofia/external/999@192.168.1.101) State EXECUTE going to sleep 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:467 (sofia/external/999@192.168.1.101) Running State Change CS_ROUTING 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:523 (sofia/external/999@192.168.1.101) State ROUTING 2014-03-26 11:04:07.345480 [DEBUG] mod_sofia.c:123 sofia/external/999@192.168.1.101 SOFIA ROUTING 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:164 sofia/external/999@192.168.1.101 Standard ROUTING 2014-03-26 11:04:07.345480 [DEBUG] mod_enum.c:642 ENUM Lookup on kb-888 2014-03-26 11:04:07.345480 [DEBUG] mod_enum.c:494 No Nameservers specified, using host default 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:214 (sofia/external/999@192.168.1.101) State Change CS_ROUTING -> CS_EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] switch_core_session.c:1385 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:523 (sofia/external/999@192.168.1.101) State ROUTING going to sleep 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:467 (sofia/external/999@192.168.1.101) Running State Change CS_EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:530 (sofia/external/999@192.168.1.101) State EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] mod_sofia.c:178 sofia/external/999@192.168.1.101 SOFIA EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:256 sofia/external/999@192.168.1.101 Standard EXECUTE 2014-03-26 11:04:07.405488 [NOTICE] switch_core_state_machine.c:313 sofia/external/999@192.168.1.101 has executed the last dialplan instruction, hanging up. 2014-03-26 11:04:07.405488 [NOTICE] switch_core_state_machine.c:315 Hangup sofia/external/999@192.168.1.101 [CS_EXECUTE] [NORMAL_CLEARING] 2014-03-26 11:04:07.405488 [DEBUG] switch_channel.c:3215 Send signal sofia/external/999@192.168.1.101 [KILL] 2014-03-26 11:04:07.405488 [DEBUG] switch_core_session.c:1385 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:530 (sofia/external/999@192.168.1.101) State EXECUTE going to sleep 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:467 (sofia/external/999@192.168.1.101) Running State Change CS_HANGUP 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:730 (sofia/external/999@192.168.1.101) Callstate Change RINGING -> HANGUP 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:732 (sofia/external/999@192.168.1.101) State HANGUP 2014-03-26 11:04:07.405488 [DEBUG] mod_sofia.c:413 Channel sofia/external/999@192.168.1.101 hanging up, cause: NORMAL_CLEARING 2014-03-26 11:04:07.405488 [DEBUG] mod_sofia.c:547 Responding to INVITE with: 480 send 777 bytes to udp/[192.168.1.101]:5060 at 11:04:07.412221: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK9db9.11128f7.0 Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKe920d3d0328D8D59 Max-Forwards: 14 From: "999" sip:999@192.168.1.101;tag=11B3C4E2-A9A9E183 To: sip:888@192.168.1.101;user=phone;tag=vmNpFtt7t5t1D Call-ID: ff78246e-9eeda51f-a54bce3c@192.168.1.102 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140320T233219Z~dd242f3ba6~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "kb-888" <kb-888>;party=calling;privacy=off;screen=no Unfortunately, i'm new to sip, kamailio and freeswitch so I apologize in advanced if I've missed something basic. but I've been over the article and have tried to ensure that I did every step. The good news is that conference calls work! But i can't call between extensions or get voicemail working.
I've attached a tcpdump on port 5060 from my kam server.
I'm not expecting hand holding but even if you could just tell me which module i should look into or additional steps on how to troubleshoot, that'd be great. So far, I've turned on debugging using the freeswitch cli, and I'm using tcpdump for the kam side of things.
thanks.
On Monday, March 24, 2014 11:43:05 AM, mark li limark67@yahoo.com wrote:
Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow the article located here:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
I've tried to add all the sections marked with WITH_FREESWITCH in the sample config in the article into my own kamailio-advanced.cfg file. Here's what my cfg file looks like: http://pastebin.com/KsvrYVN7
I've restarted kamailio after making these changes. Then I tried to dial 41 to listen to vmail or 44999 to leave a message for user 999 but both return a busy tone.
Any suggestions would be appreciated. Thanks. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Ok so here's the latest status:
I still get a busy signal when I try to call from one phone to the other. But I found one problem that was contributing to the issue.
In my kamailio.cfg, I was adding a 'kb-' prefix before I route the call to freeswitch. And on the freeswitch side, I was looking for that prefix in my dialplan and then stripping off the prefix before I tried to send the call back kamailio. The INVITE that Kamailio was creating had the 'kb-' in the SIP Address. But I don't have any extensions called kb-888 or kb-999. They are 888 and 999. So now, my INVITE requests that Kamailio creates look correct.http://pastebin.com/tuGGpCn7
The call still doesn't complete but i think i'm one step closer.
what i'm wondering is are there any settings in Kamailio that i need to "accept" sip calls from freeswitch?
In freeswitch you have to specify specific domains in the acl.conf.xml file... just wondering if kamailio has something similar?
Thanks.
________________________________ From: mark li limark67@yahoo.com To: mark li limark67@yahoo.com; "sr-users@lists.sip-router.org" sr-users@lists.sip-router.org; Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Sent: Wednesday, March 26, 2014 2:27:05 PM Subject: Re: [SR-Users] Integrating Kamailio and Freeswitch
Two updates:
1. the link i had was incorrect. this is the correct link to the "how to" that I'm following:
http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc 2. I've narrowed down the issue a little bit. Here's what I've found, along with some more background information:
a) i have two polycom phones on my network, ext 888 and 999. b) they both register fine to the sip proxy. (192.168.1.101)
c) when i try to call ext 888 from ext 999, via a tcpdump, i can see that the call makes it to the freeswitch server (192.169.1.111) Since ext 888 is online, it should have just directed the call to the phone (vs. going to vmail) but instead, I get the following error message:
SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK9db9.11128f7.0 Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKe920d3d0328D8D59 Max-Forwards: 14 From: "999" sip:999@192.168.1.101;tag=11B3C4E2-A9A9E183 To: sip:888@192.168.1.101;user=phone;tag=vmNpFtt7t5t1D Call-ID: ff78246e-9eeda51f-a54bce3c@192.168.1.102 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140320T233219Z~dd242f3ba6~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "kb-888" <kb-888>;party=calling;privacy=off;screen=no
In debugging on freeswitch I can see that it tries to match a dialplan for kb-888 and then ends up attempting to do a enum look up on "kb-888". Then it says that it has completed the dialplan.
I don't have an extension kb-888 registered. I can see where in the configs that I am prefixing "kb" to the calls on the kamailio side. And on the freeswitch side of the house, I see where I a regular expression looking for this prefix. But I don't know how i can get freeswitch to send the call to 888@192.168.1.101
Here's some of the debug data from freeswitch:
2014-03-26 11:04:07.345480 [DEBUG] switch_ivr.c:1830 (sofia/external/999@192.168.1.101) State Change CS_EXECUTE -> CS_ROUTING 2014-03-26 11:04:07.345480 [DEBUG] switch_core_session.c:1385 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.345480 [DEBUG] switch_core_session.c:905 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.345480 [NOTICE] switch_ivr.c:1837 Transfer sofia/external/999@192.168.1.101 to enum[kb-888@default] 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:530 (sofia/external/999@192.168.1.101) State EXECUTE going to sleep 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:467 (sofia/external/999@192.168.1.101) Running State Change CS_ROUTING 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:523 (sofia/external/999@192.168.1.101) State ROUTING 2014-03-26 11:04:07.345480 [DEBUG] mod_sofia.c:123 sofia/external/999@192.168.1.101 SOFIA ROUTING 2014-03-26 11:04:07.345480 [DEBUG] switch_core_state_machine.c:164 sofia/external/999@192.168.1.101 Standard ROUTING 2014-03-26 11:04:07.345480 [DEBUG] mod_enum.c:642 ENUM Lookup on kb-888 2014-03-26 11:04:07.345480 [DEBUG] mod_enum.c:494 No Nameservers specified, using host default 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:214 (sofia/external/999@192.168.1.101) State Change CS_ROUTING -> CS_EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] switch_core_session.c:1385 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:523 (sofia/external/999@192.168.1.101) State ROUTING going to sleep 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:467 (sofia/external/999@192.168.1.101) Running State Change CS_EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:530 (sofia/external/999@192.168.1.101) State EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] mod_sofia.c:178 sofia/external/999@192.168.1.101 SOFIA EXECUTE 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:256 sofia/external/999@192.168.1.101 Standard EXECUTE 2014-03-26 11:04:07.405488 [NOTICE] switch_core_state_machine.c:313 sofia/external/999@192.168.1.101 has executed the last dialplan instruction, hanging up. 2014-03-26 11:04:07.405488 [NOTICE] switch_core_state_machine.c:315 Hangup sofia/external/999@192.168.1.101 [CS_EXECUTE] [NORMAL_CLEARING] 2014-03-26 11:04:07.405488 [DEBUG] switch_channel.c:3215 Send signal sofia/external/999@192.168.1.101 [KILL] 2014-03-26 11:04:07.405488 [DEBUG] switch_core_session.c:1385 Send signal sofia/external/999@192.168.1.101 [BREAK] 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:530 (sofia/external/999@192.168.1.101) State EXECUTE going to sleep 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:467 (sofia/external/999@192.168.1.101) Running State Change CS_HANGUP 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:730 (sofia/external/999@192.168.1.101) Callstate Change RINGING -> HANGUP 2014-03-26 11:04:07.405488 [DEBUG] switch_core_state_machine.c:732 (sofia/external/999@192.168.1.101) State HANGUP 2014-03-26 11:04:07.405488 [DEBUG] mod_sofia.c:413 Channel sofia/external/999@192.168.1.101 hanging up, cause: NORMAL_CLEARING 2014-03-26 11:04:07.405488 [DEBUG] mod_sofia.c:547 Responding to INVITE with: 480 send 777 bytes to udp/[192.168.1.101]:5060 at 11:04:07.412221: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK9db9.11128f7.0 Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKe920d3d0328D8D59 Max-Forwards: 14 From: "999" sip:999@192.168.1.101;tag=11B3C4E2-A9A9E183 To: sip:888@192.168.1.101;user=phone;tag=vmNpFtt7t5t1D Call-ID: ff78246e-9eeda51f-a54bce3c@192.168.1.102 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.12b+git~20140320T233219Z~dd242f3ba6~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "kb-888" <kb-888>;party=calling;privacy=off;screen=no Unfortunately, i'm new to sip, kamailio and freeswitch so I apologize in advanced if I've missed something basic. but I've been over the article and have tried to ensure that I did every step. The good news is that conference calls work! But i can't call between extensions or get voicemail working.
I've attached a tcpdump on port 5060 from my kam server.
I'm not expecting hand holding but even if you could just tell me which module i should look into or additional steps on how to troubleshoot, that'd be great. So far, I've turned on debugging using the freeswitch cli, and I'm using tcpdump for the kam side of things.
thanks.
On Monday, March 24, 2014 11:43:05 AM, mark li limark67@yahoo.com wrote:
Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow the article located here:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
I've tried to add all the sections marked with WITH_FREESWITCH in the sample config in the article into my own kamailio-advanced.cfg file. Here's what my cfg file looks like: http://pastebin.com/KsvrYVN7
I've restarted kamailio after making these changes. Then I tried to dial 41 to listen to vmail or 44999 to leave a message for user 999 but both return a busy tone.
Any suggestions would be appreciated. Thanks. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
mark li