I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound. My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly. The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338> Snippets of the log that looks suspicious to a freeswitch noob: 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me). Thanks, -Terry
Check to see if your IP addressing settings are set correctly within the dingaling profile. Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
Thanks, -Terry
__________________________________________________________ _______________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}. <include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> Thanks for any pointers. -Terry On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
Thanks, -Terry
__________________________________________________________ _______________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi FS gurus, Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings? Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ... Thanks for any help, -Terry On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
I stopped fs, did a fresh git clone git://git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z) I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output: 2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq> 2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</redirect> <ses:redirect xmlns:ses="http://www.google.com/session">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</ses:redirect> </error> </iq> -Terry On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Since I'm sure everyone is anxiously awaiting the next installment of my google voice monologue... ;) So for google inbound calls I somehow overlooked that call screening needed to be enabled ("execute_on_answer=send_dtmf 1") so I added it and now inbound google voice calls work properly. Does this imply that NAT is being successfully traversed and this is not the reason for the google outbound problem? Looking further at the dingaling debug output for google outbound, there's an empty decline tag and also a "Caller disconnected." As an inexperienced dingaling(!) user I'm not sure how to interpret this--the caller *has been* disconnected? Meaning google terminated the call? Or the caller *has* disconnected? Meaning freeswitch terminated the call? Any ideas why this is failing? 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3597 candidate 173.194.79.127:19305 PASS ACL wan.auto 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 173.194.79.127:19305 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-09-01 17:44:27.625074 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" from="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE502F"> <query></query> </iq> 2013-09-01 17:44:28.084920 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" to="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE5032" type="set"> <ses:session type="terminate" id="6914264867" initiator="USER@gmail.com/talkEAB90707" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:decline></ses:decline> <ses:text>Caller disconnected</ses:text> </ses:reason> <pho:call-ended xmlns:pho="http://www.google.com/session/phone"></pho:call-ended> </ses:session> </iq> 2013-09-01 17:44:28.084920 [CRIT] libdingaling.c:391 Message for Session 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4114 using Existing session for 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+760XXXXXXX@voice.google.com Thanks, -Terry On Sep 1, 2013, at 10:45 AM, Terry Barnum wrote:
I stopped fs, did a fresh git clone git://git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z)
I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output:
2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq>
2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</redirect> <ses:redirect xmlns:ses="http://www.google.com/session">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</ses:redirect> </error> </iq>
-Terry
On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
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Terry, Unable to contribute to this discussion, just wanted to thank you for taking time to share your learning. I'm pretty new to FS, but what I've been enjoying about FS is that, usually Wiki's contain (almost) everything needed to get something going, or searching this list (thanks to threads like this one) or the JIRA comments helps a lot. Of course, this self-learning the hard-way takes time, but for me helps remember and understand things better. I've to admit that I'm flirting a bit on the sidelines with Asterisks, especially as I keep getting blocked and feel that I am not making much progress with FS, but keep coming back. Not sure, but probably the list is a bit silent on features that are niche, and not used by mainstream users. That's precisely what I've found. Pretty usual for any FOSS project. Please do keep sharing your learning here. Jay. On Mon, Sep 2, 2013 at 6:55 AM, Terry Barnum <terry@digital-outpost.com>wrote:
Since I'm sure everyone is anxiously awaiting the next installment of my google voice monologue... ;)
So for google inbound calls I somehow overlooked that call screening needed to be enabled ("execute_on_answer=send_dtmf 1") so I added it and now inbound google voice calls work properly. Does this imply that NAT is being successfully traversed and this is not the reason for the google outbound problem?
Looking further at the dingaling debug output for google outbound, there's an empty decline tag and also a "Caller disconnected." As an inexperienced dingaling(!) user I'm not sure how to interpret this--the caller *has been* disconnected? Meaning google terminated the call? Or the caller *has* disconnected? Meaning freeswitch terminated the call? Any ideas why this is failing?
2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3597 candidate 173.194.79.127:19305 PASS ACL wan.auto 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 173.194.79.127:19305 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-09-01 17:44:27.625074 [NOTICE] libdingaling.c:1749 SecSEND:
------------------------------------------------------------------------------- <iq type="result" to="+ 760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" from="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE502F"> <query></query> </iq>
2013-09-01 17:44:28.084920 [INFO] libdingaling.c:1747 SecRECV:
------------------------------------------------------------------------------- <iq from="+ 760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" to=" USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE5032" type="set"> <ses:session type="terminate" id="6914264867" initiator=" USER@gmail.com/talkEAB90707" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:decline></ses:decline> <ses:text>Caller disconnected</ses:text> </ses:reason> <pho:call-ended xmlns:pho="http://www.google.com/session/phone "></pho:call-ended> </ses:session> </iq>
2013-09-01 17:44:28.084920 [CRIT] libdingaling.c:391 Message for Session 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4114 using Existing session for 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+760XXXXXXX@voice.google.com
Thanks, -Terry
On Sep 1, 2013, at 10:45 AM, Terry Barnum wrote:
I stopped fs, did a fresh git clone git:// git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z)
I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output:
2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND:
-------------------------------------------------------------------------------
<iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+ 760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone "></pho:payload-type> </pho:description> </ses:session> </iq>
2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV:
-------------------------------------------------------------------------------
<iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator=" USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"> xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ== </redirect> <ses:redirect xmlns:ses="http://www.google.com/session"> xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ== </ses:redirect> </error> </iq>
-Terry
On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/ 102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/ 102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/ 102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/ 102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/ 102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/ 102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+ 7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/ 102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/ 102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com[200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks for the encouraging words Jay. I also spent a little time deciding which telephony package to try although it didn't take too long to decide on FS. I have been scouring the wiki, JIRA and the Freeswitch book but it's slow going because I'm not exactly sure what the underlying issue is with my broken outbound google voice. It will likely turn out to be a stupid configuration error but deciphering what's relevant/important in debug output has been a slow process since everything is so unfamiliar. If it ends up being a doofus syntax error I'll have to get this T-shirt I saw recently: Let's eat grandma! Let's eat, grandma! Punctuation save lives! -Terry On Sep 1, 2013, at 11:00 PM, Jayanth Acharya wrote:
Terry,
Unable to contribute to this discussion, just wanted to thank you for taking time to share your learning. I'm pretty new to FS, but what I've been enjoying about FS is that, usually Wiki's contain (almost) everything needed to get something going, or searching this list (thanks to threads like this one) or the JIRA comments helps a lot. Of course, this self-learning the hard-way takes time, but for me helps remember and understand things better.
I've to admit that I'm flirting a bit on the sidelines with Asterisks, especially as I keep getting blocked and feel that I am not making much progress with FS, but keep coming back.
Not sure, but probably the list is a bit silent on features that are niche, and not used by mainstream users. That's precisely what I've found. Pretty usual for any FOSS project. Please do keep sharing your learning here.
Jay.
On Mon, Sep 2, 2013 at 6:55 AM, Terry Barnum <terry@digital-outpost.com> wrote: Since I'm sure everyone is anxiously awaiting the next installment of my google voice monologue... ;)
So for google inbound calls I somehow overlooked that call screening needed to be enabled ("execute_on_answer=send_dtmf 1") so I added it and now inbound google voice calls work properly. Does this imply that NAT is being successfully traversed and this is not the reason for the google outbound problem?
Looking further at the dingaling debug output for google outbound, there's an empty decline tag and also a "Caller disconnected." As an inexperienced dingaling(!) user I'm not sure how to interpret this--the caller *has been* disconnected? Meaning google terminated the call? Or the caller *has* disconnected? Meaning freeswitch terminated the call? Any ideas why this is failing?
2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3597 candidate 173.194.79.127:19305 PASS ACL wan.auto 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 173.194.79.127:19305 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-09-01 17:44:27.625074 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" from="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE502F"> <query></query> </iq>
2013-09-01 17:44:28.084920 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" to="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE5032" type="set"> <ses:session type="terminate" id="6914264867" initiator="USER@gmail.com/talkEAB90707" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:decline></ses:decline> <ses:text>Caller disconnected</ses:text> </ses:reason> <pho:call-ended xmlns:pho="http://www.google.com/session/phone"></pho:call-ended> </ses:session> </iq>
2013-09-01 17:44:28.084920 [CRIT] libdingaling.c:391 Message for Session 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4114 using Existing session for 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+760XXXXXXX@voice.google.com
Thanks, -Terry
On Sep 1, 2013, at 10:45 AM, Terry Barnum wrote:
I stopped fs, did a fresh git clone git://git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z)
I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output:
2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq>
2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</redirect> <ses:redirect xmlns:ses="http://www.google.com/session">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</ses:redirect> </error> </iq>
-Terry
On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Jay and terry, I have been a asterisk production user for a 5 years and while asterisk easier to get going 5 years ago. I can tell you this making the switch from asterisk to FS was super easy. Irc the wiki are major helpers. The syntax and dial plans are much cleaner and easier to maintain then asterisk. Sent from my iPhone 5 please excuse typos
On Sep 2, 2013, at 2:29 AM, Terry Barnum <terry@digital-outpost.com> wrote:
Thanks for the encouraging words Jay. I also spent a little time deciding which telephony package to try although it didn't take too long to decide on FS. I have been scouring the wiki, JIRA and the Freeswitch book but it's slow going because I'm not exactly sure what the underlying issue is with my broken outbound google voice. It will likely turn out to be a stupid configuration error but deciphering what's relevant/important in debug output has been a slow process since everything is so unfamiliar.
If it ends up being a doofus syntax error I'll have to get this T-shirt I saw recently:
Let's eat grandma! Let's eat, grandma!
Punctuation save lives!
-Terry
On Sep 1, 2013, at 11:00 PM, Jayanth Acharya wrote:
Terry,
Unable to contribute to this discussion, just wanted to thank you for taking time to share your learning. I'm pretty new to FS, but what I've been enjoying about FS is that, usually Wiki's contain (almost) everything needed to get something going, or searching this list (thanks to threads like this one) or the JIRA comments helps a lot. Of course, this self-learning the hard-way takes time, but for me helps remember and understand things better.
I've to admit that I'm flirting a bit on the sidelines with Asterisks, especially as I keep getting blocked and feel that I am not making much progress with FS, but keep coming back.
Not sure, but probably the list is a bit silent on features that are niche, and not used by mainstream users. That's precisely what I've found. Pretty usual for any FOSS project. Please do keep sharing your learning here.
Jay.
On Mon, Sep 2, 2013 at 6:55 AM, Terry Barnum <terry@digital-outpost.com> wrote: Since I'm sure everyone is anxiously awaiting the next installment of my google voice monologue... ;)
So for google inbound calls I somehow overlooked that call screening needed to be enabled ("execute_on_answer=send_dtmf 1") so I added it and now inbound google voice calls work properly. Does this imply that NAT is being successfully traversed and this is not the reason for the google outbound problem?
Looking further at the dingaling debug output for google outbound, there's an empty decline tag and also a "Caller disconnected." As an inexperienced dingaling(!) user I'm not sure how to interpret this--the caller *has been* disconnected? Meaning google terminated the call? Or the caller *has* disconnected? Meaning freeswitch terminated the call? Any ideas why this is failing?
2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3597 candidate 173.194.79.127:19305 PASS ACL wan.auto 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 173.194.79.127:19305 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-09-01 17:44:27.625074 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" from="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE502F"> <query></query> </iq>
2013-09-01 17:44:28.084920 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" to="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE5032" type="set"> <ses:session type="terminate" id="6914264867" initiator="USER@gmail.com/talkEAB90707" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:decline></ses:decline> <ses:text>Caller disconnected</ses:text> </ses:reason> <pho:call-ended xmlns:pho="http://www.google.com/session/phone"></pho:call-ended> </ses:session> </iq>
2013-09-01 17:44:28.084920 [CRIT] libdingaling.c:391 Message for Session 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4114 using Existing session for 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+760XXXXXXX@voice.google.com
Thanks, -Terry
On Sep 1, 2013, at 10:45 AM, Terry Barnum wrote:
I stopped fs, did a fresh git clone git://git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z)
I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output:
2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq>
2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</redirect> <ses:redirect xmlns:ses="http://www.google.com/session">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</ses:redirect> </error> </iq>
-Terry
On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
> -----Original Message----- > From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- > users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum > Sent: Monday, August 26, 2013 9:33 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Google voice call fails > > I've been reading the freeswitch book (great read!) and playing with my first > freeswitch install on a Mac mini host with freeswitch installed via git into a > Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and > a Yealink T32G, internally between extensions, using a free DID to test > incoming calls and a free but limited iptel account to test outbound. > > My next step was to try and play with real outbound calls so I followed these > instructions for Google voice: > <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and > are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just > PCMU but no change. xmpp appears to be connecting and authenticating to > google correctly. > > The log of an attempted call (edited for privacy) is at > <http://pastebin.freeswitch.org/21338> > > Snippets of the log that looks suspicious to a freeswitch noob: > > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 > rtp candidates. > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 > rtcp candidates. > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 > video_rtp candidates > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 > video_rctp candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing > session for 2155444888 > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate > 74.125.141.127:19305 PASS ACL wan.auto > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp > Candidate 74.125.141.127:19305 > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp > candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp > candidates > 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing > session for 2155444888 > 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup > dingaling/gtalk/+7609876543@voice.google.com > > Where should I be looking to try and fix? Freeswitch is behind the firewall > with a private IP but STUN says it's working (or at least it looks that way to > me).
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
This is a reject from google, I think its saying the remote side rejected your call. If your calling to your own account do you get any indication? What if your calling to another freeswitch? On Sep 1, 2013, at 9:25 PM, Terry Barnum <terry@digital-outpost.com> wrote:
Since I'm sure everyone is anxiously awaiting the next installment of my google voice monologue... ;)
So for google inbound calls I somehow overlooked that call screening needed to be enabled ("execute_on_answer=send_dtmf 1") so I added it and now inbound google voice calls work properly. Does this imply that NAT is being successfully traversed and this is not the reason for the google outbound problem?
Looking further at the dingaling debug output for google outbound, there's an empty decline tag and also a "Caller disconnected." As an inexperienced dingaling(!) user I'm not sure how to interpret this--the caller *has been* disconnected? Meaning google terminated the call? Or the caller *has* disconnected? Meaning freeswitch terminated the call? Any ideas why this is failing?
2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3597 candidate 173.194.79.127:19305 PASS ACL wan.auto 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 173.194.79.127:19305 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-09-01 17:44:27.625074 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" from="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE502F"> <query></query> </iq>
2013-09-01 17:44:28.084920 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" to="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE5032" type="set"> <ses:session type="terminate" id="6914264867" initiator="USER@gmail.com/talkEAB90707" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:decline></ses:decline> <ses:text>Caller disconnected</ses:text> </ses:reason> <pho:call-ended xmlns:pho="http://www.google.com/session/phone"></pho:call-ended> </ses:session> </iq>
2013-09-01 17:44:28.084920 [CRIT] libdingaling.c:391 Message for Session 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4114 using Existing session for 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+760XXXXXXX@voice.google.com
Thanks, -Terry
On Sep 1, 2013, at 10:45 AM, Terry Barnum wrote:
I stopped fs, did a fresh git clone git://git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z)
I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output:
2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq>
2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</redirect> <ses:redirect xmlns:ses="http://www.google.com/session">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</ses:redirect> </error> </iq>
-Terry
On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum Sent: Monday, August 26, 2013 9:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Google voice call fails
I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.
My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.
The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>
Snippets of the log that looks suspicious to a freeswitch noob:
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com
Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me).
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thank you very much for the reply Michael. On the google voice history page I can see all my attempted calls at 0 minutes long, so they are making it to a certain point in the process. There's no detail though. I tried calling my google voice account from a softphone with the same result. Calls in the other direction work correctly: landline->GV number->FS->Yealink rings I'm sorry but I don't know how to call another freeswitch. Are there open test servers? Thanks, -Terry On Sep 3, 2013, at 5:17 AM, Michael Jerris wrote:
This is a reject from google, I think its saying the remote side rejected your call. If your calling to your own account do you get any indication? What if your calling to another freeswitch?
On Sep 1, 2013, at 9:25 PM, Terry Barnum <terry@digital-outpost.com> wrote:
Since I'm sure everyone is anxiously awaiting the next installment of my google voice monologue... ;)
So for google inbound calls I somehow overlooked that call screening needed to be enabled ("execute_on_answer=send_dtmf 1") so I added it and now inbound google voice calls work properly. Does this imply that NAT is being successfully traversed and this is not the reason for the google outbound problem?
Looking further at the dingaling debug output for google outbound, there's an empty decline tag and also a "Caller disconnected." As an inexperienced dingaling(!) user I'm not sure how to interpret this--the caller *has been* disconnected? Meaning google terminated the call? Or the caller *has* disconnected? Meaning freeswitch terminated the call? Any ideas why this is failing?
2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3597 candidate 173.194.79.127:19305 PASS ACL wan.auto 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 173.194.79.127:19305 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-09-01 17:44:27.625074 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-09-01 17:44:27.625074 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" from="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE502F"> <query></query> </iq>
2013-09-01 17:44:28.084920 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com/srvenc-y7H55UzKUf9oV+YD+bPou9s/IeoQd1fm" to="USER@gmail.com/talkEAB90707" id="jingle:10.76.86.101-7841406:1:C7EE5032" type="set"> <ses:session type="terminate" id="6914264867" initiator="USER@gmail.com/talkEAB90707" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:decline></ses:decline> <ses:text>Caller disconnected</ses:text> </ses:reason> <pho:call-ended xmlns:pho="http://www.google.com/session/phone"></pho:call-ended> </ses:session> </iq>
2013-09-01 17:44:28.084920 [CRIT] libdingaling.c:391 Message for Session 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4114 using Existing session for 6914264867 2013-09-01 17:44:28.084920 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+760XXXXXXX@voice.google.com
Thanks, -Terry
On Sep 1, 2013, at 10:45 AM, Terry Barnum wrote:
I stopped fs, did a fresh git clone git://git.freeswitch.org/freeswitch.git, edited modules.conf and did a make && make install but still no joy making a call via google voice. dingaling status shows AUTHORIZED. FreeSWITCH (Version 1.5.6b git dd6b8a8 2013-08-31 03:41:31Z)
I set dingaling_conf.xml to debug and it reports an error 302 when attempting a call. I see others have had this error in the past but I couldn't find a definitive answer. Can anyone shed light on this? Thanks for any pointers. Here's a snippet of the dingaling debug output:
2013-09-01 08:48:24.504839 [NOTICE] libdingaling.c:1749 SecSEND: ------------------------------------------------------------------------------- <iq xmlns="jabber:client" from="USER@gmail.com/talk5CB5BB1C" to="+760XXXXXXX@voice.google.com" type="set" id="300"> <ses:session xmlns:ses="http://www.google.com/session" type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq>
2013-09-01 08:48:24.604828 [INFO] libdingaling.c:1747 SecRECV: ------------------------------------------------------------------------------- <iq from="+760XXXXXXX@voice.google.com" to="USER@gmail.com/talk5CB5BB1C" type="error" id="300"> <ses:session type="initiate" id="2395886559" initiator="USER@gmail.com/talk5CB5BB1C" xmlns:ses="http://www.google.com/session"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"></pho:payload-type> </pho:description> </ses:session> <error code="302" type="modify"> <redirect xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</redirect> <ses:redirect xmlns:ses="http://www.google.com/session">xmpp:+760XXXXXXX@voice.google.com/srvenc-w8W24IH4ZBGhR6mQVmzatQ==</ses:redirect> </error> </iq>
-Terry
On Aug 29, 2013, at 8:43 PM, Terry Barnum wrote:
Hi FS gurus,
Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?
Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing. ... Dialplan: sofia/internal/102@172.16.1.199 parsing [default->gvoice_out] continue=false Dialplan: sofia/internal/102@172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false Dialplan: sofia/internal/102@172.16.1.199 Action set(call_direction=outbound) Dialplan: sofia/internal/102@172.16.1.199 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/102@172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/102@172.16.1.199 Action set(inherit_codec=true) Dialplan: sofia/internal/102@172.16.1.199 Action bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102@172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102@172.16.1.199 [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102@172.16.1.199) State ROUTING going to sleep 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102@172.16.1.199) Running State Change CS_EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102@172.16.1.199) State EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102@172.16.1.199 SOFIA EXECUTE 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102@172.16.1.199 Standard EXECUTE EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567) EXECUTE sofia/internal/102@172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391) EXECUTE sofia/internal/102@172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700] EXECUTE sofia/internal/102@172.16.1.199 set(call_direction=outbound) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [call_direction]=[outbound] EXECUTE sofia/internal/102@172.16.1.199 set(hangup_after_bridge=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_name=Terry) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_name]=[Terry] EXECUTE sofia/internal/102@172.16.1.199 set(effective_caller_id_number=760-999-9999) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [effective_caller_id_number]=[760-999-9999] EXECUTE sofia/internal/102@172.16.1.199 set(inherit_codec=true) 2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102@172.16.1.199 SET [inherit_codec]=[true] EXECUTE sofia/internal/102@172.16.1.199 bridge(dingaling/gtalk/+7609876543@voice.google.com) 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102@172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event 2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N] 2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543@voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543@voice.google.com) State Change CS_NEW -> CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543@voice.google.com [BREAK] 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543@voice.google.com CHANNEL KILL 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543@voice.google.com) Running State Change CS_INIT 2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543@voice.google.com) State INIT 2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543@voice.google.com! 2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543@voice.google.com) Callstate Change DOWN -> RINGING 2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec. 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU@8000] 2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116 2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo] 2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117 2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws] 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543@voice.google.com ...
Thanks for any help, -Terry
On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:
Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="USER@gmail.com/talk"/> <param name="password" value="MYPASS"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="102"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
Thanks for any pointers.
-Terry
On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
Check to see if your IP addressing settings are set correctly within the dingaling profile.
Most of the times you'll need to have the profile to listen internally but have dingaling to specify an external IP address when it talks to the outside world.
> -----Original Message----- > From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- > users-bounces@lists.freeswitch.org] On Behalf Of Terry Barnum > Sent: Monday, August 26, 2013 9:33 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Google voice call fails > > I've been reading the freeswitch book (great read!) and playing with my first > freeswitch install on a Mac mini host with freeswitch installed via git into a > Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and > a Yealink T32G, internally between extensions, using a free DID to test > incoming calls and a free but limited iptel account to test outbound. > > My next step was to try and play with real outbound calls so I followed these > instructions for Google voice: > <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and > are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just > PCMU but no change. xmpp appears to be connecting and authenticating to > google correctly. > > The log of an attempted call (edited for privacy) is at > <http://pastebin.freeswitch.org/21338> > > Snippets of the log that looks suspicious to a freeswitch noob: > > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 > rtp candidates. > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 > rtcp candidates. > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 > video_rtp candidates > 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 > video_rctp candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing > session for 2155444888 > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate > 74.125.141.127:19305 PASS ACL wan.auto > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp > Candidate 74.125.141.127:19305 > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp > candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates > 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp > candidates > 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing > session for 2155444888 > 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup > dingaling/gtalk/+7609876543@voice.google.com > > Where should I be looking to try and fix? Freeswitch is behind the firewall > with a private IP but STUN says it's working (or at least it looks that way to > me).
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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participants (5)
-
Jayanth Acharya
-
Jeff Leung
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Michael Jerris
-
Richard Genthner
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Terry Barnum