mod_dingaling & inbound audio
Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio. FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem In and out calling on SIP to provider works fine in both cases. So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users? Thank you for you help. "Resistance is Fruitful"
use pastebin.freeswitch.org and pastebin your configs and a debug output of the failed call. be sure that you have debug level output (which is default if you use fs_cli) -MC On Tue, Feb 8, 2011 at 1:07 PM, Locutis of Borg <locutis@sect001.net> wrote:
Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio.
FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem
In and out calling on SIP to provider works fine in both cases.
So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users?
Thank you for you help.
"Resistance is Fruitful"
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As instructed, I have posted the jingle config and a fs_cli debug log from starting at the dialplan down through hangup. Thank you for your help on this. Pastebin log and config under the name Locutis :) -Borg On Tue, Feb 8, 2011 at 4:15 PM, Michael Collins <msc@freeswitch.org> wrote:
use pastebin.freeswitch.org and pastebin your configs and a debug output of the failed call. be sure that you have debug level output (which is default if you use fs_cli) -MC
On Tue, Feb 8, 2011 at 1:07 PM, Locutis of Borg <locutis@sect001.net>wrote:
Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio.
FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem
In and out calling on SIP to provider works fine in both cases.
So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users?
Thank you for you help.
"Resistance is Fruitful"
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On previous git version I have working mod dingaling, but until today my dingaling is never working. If anybody has freeswitch source with dingaling works, please share the url. I will download Thx Budi -----Original Message----- From: Locutis of Borg <locutis@sect001.net> Sender: freeswitch-users-bounces@lists.freeswitch.org Date: Wed, 9 Feb 2011 14:33:02 To: FreeSWITCH Users Help<freeswitch-users@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] mod_dingaling & inbound audio _______________________________________________ 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
On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo@gmail.com> wrote:
On previous git version I have working mod dingaling, but until today my dingaling is never working. If anybody has freeswitch source with dingaling works, please share the url. I will download
What does the console log show? Also, turn on dingaling debugging w/ the "dl_debug on" command. -MC
hi michael this is my output screen, i consider on BREAK and CHANNEL KILL here, i use xlite and acrobits with different ip provider to test. and both silent with dingaling thx budi 2011-02-13 08:26:06.523030 [INFO] libdingaling.c:1366 SecRECV: ------------------------------------------------------------------------------- <iq to="bwibowo@gmail.com/gtalkE0340285" from="+ 14085264000@voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" id="306" type="result"></iq> 2011-02-13 08:26:06.523030 [DEBUG] libdingaling.c:968 Cancel packet 306 2011-02-13 08:26:06.617029 [DEBUG] libdingaling.c:1450 Processing 1 packets in retry queue 2011-02-13 08:26:06.617029 [DEBUG] libdingaling.c:1475 Discarding packet 306 2011-02-13 08:26:06.721027 [INFO] libdingaling.c:1366 SecRECV: ------------------------------------------------------------------------------- <iq from="+14085264000@voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" to="bwibowo@gmail.com/gtalkE0340285" id="jingle:10.13.118.26-26031162:1:03888679" type="set"> <ses:session type="transport-info" id="9537230161" initiator=" bwibowo@gmail.com/gtalkE0340285" xmlns:ses="http://www.google.com/session"> <transport xmlns="http://www.google.com/transport/p2p"> <candidate name="rtp" address="74.125.153.126" port="19295" username="f1ipq5kmX1kt1UlH" preference="1.0" protocol="udp" network="mediaproxy" generation="0"></candidate> <candidate name="rtp" address="74.125.153.126" port="19294" username="f1ipq5kmX1kt1UlH" preference="0.6" protocol="tcp" network="mediaproxy" generation="0"></candidate> <candidate name="rtp" address="74.125.153.126" port="443" username="f1ipq5kmX1kt1UlH" preference="0.5" protocol="ssltcp" network="mediaproxy" generation="0"></candidate> </transport> </ses:session> </iq> 2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:383 Message for Session 9537230161 2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:528 New Candidate 1 name=rtp type=stun protocol=udp username=f1ipq5kmX1kt1UlH password=(null) address=74.125.153.126 port=19295 pref=1.00 2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:528 New Candidate 2 name=rtp type=stun protocol=tcp username=f1ipq5kmX1kt1UlH password=(null) address=74.125.153.126 port=19294 pref=0.60 2011-02-13 08:26:06.721027 [DEBUG] libdingaling.c:528 New Candidate 3 name=rtp type=stun protocol=ssltcp username=f1ipq5kmX1kt1UlH password=(null) address=74.125.153.126 port=443 pref=0.50 2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:2941 using Existing session for 9537230161 2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:3279 3 candidates 2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:3299 candidate 74.125.153.126:19295 PASS ACL wan.auto 2011-02-13 08:26:06.721027 [DEBUG] mod_dingaling.c:3351 Acceptable Candidate 74.125.153.126:19295 2011-02-13 08:26:06.817031 [NOTICE] libdingaling.c:1368 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+ 14085264000@voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" from=" bwibowo@gmail.com/gtalkE0340285" id="jingle:10.13.118.26-26031162:1:03888679"> <query></query> </iq> 2011-02-13 08:26:08.841032 [INFO] libdingaling.c:1366 SecRECV: ------------------------------------------------------------------------------- <iq from="+14085264000@voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" to="bwibowo@gmail.com/gtalkE0340285" id="jingle:10.13.118.26-26031162:1:0388867F" type="set"> <ses:session type="accept" id="9537230161" initiator=" bwibowo@gmail.com/gtalkE0340285" 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"></pho:payload-type> <pho:payload-type id="101" name="telephone-event"></pho:payload-type> </pho:description> </ses:session> </iq> 2011-02-13 08:26:08.841957 [DEBUG] libdingaling.c:383 Message for Session 9537230161 2011-02-13 08:26:08.841957 [DEBUG] libdingaling.c:440 Add Payload [PCMU] id='0' 2011-02-13 08:26:08.841957 [DEBUG] libdingaling.c:440 Add Payload [telephone-event] id='101' 2011-02-13 08:26:08.841957 [DEBUG] mod_dingaling.c:2941 using Existing session for 9537230161 2011-02-13 08:26:08.841957 [DEBUG] mod_dingaling.c:3193 Already decided on a codec 2011-02-13 08:26:08.843032 [DEBUG] mod_dingaling.c:865 Set Read Codec to PCMU@8000 2011-02-13 08:26:08.843032 [DEBUG] mod_dingaling.c:880 Set Write Codec to PCMU@8000 2011-02-13 08:26:08.843032 [DEBUG] mod_dingaling.c:892 SETUP RTP 202.122.99.99:24240 -> 74.125.153.126:19295 2011-02-13 08:26:08.843032 [DEBUG] switch_rtp.c:1607 Starting timer [soft] 160 bytes per 20ms 2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:2788 (dingaling/gtalk/+ 14085264000@voice.google.com) Callstate Change DOWN -> ACTIVE 2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:2800 Send signal sofia/internal/1000@202.122.99.99 [BREAK] 2011-02-13 08:26:08.845035 [NOTICE] mod_dingaling.c:1202 Channel [dingaling/gtalk/+14085264000@voice.google.com] has been answered 2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1205 (dingaling/gtalk/+ 14085264000@voice.google.com) State Change CS_INIT -> CS_ROUTING 2011-02-13 08:26:08.845035 [DEBUG] switch_core_session.c:1116 Send signal dingaling/gtalk/+14085264000@voice.google.com [BREAK] 2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+ 14085264000@voice.google.com CHANNEL KILL 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:356 (dingaling/gtalk/+14085264000@voice.google.com) State INIT going to sleep 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:320 (dingaling/gtalk/+14085264000@voice.google.com) Running State Change CS_ROUTING 2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:1660 (dingaling/gtalk/+ 14085264000@voice.google.com) Callstate Change ACTIVE -> RINGING 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:359 (dingaling/gtalk/+14085264000@voice.google.com) State ROUTING 2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1219 dingaling/gtalk/+ 14085264000@voice.google.com CHANNEL ROUTING 2011-02-13 08:26:08.845035 [DEBUG] switch_ivr_originate.c:66 (dingaling/gtalk/+14085264000@voice.google.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-02-13 08:26:08.845035 [DEBUG] switch_core_session.c:1116 Send signal dingaling/gtalk/+14085264000@voice.google.com [BREAK] 2011-02-13 08:26:08.845035 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+ 14085264000@voice.google.com CHANNEL KILL 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:359 (dingaling/gtalk/+14085264000@voice.google.com) State ROUTING going to sleep 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:320 (dingaling/gtalk/+14085264000@voice.google.com) Running State Change CS_CONSUME_MEDIA 2011-02-13 08:26:08.845035 [DEBUG] switch_channel.c:1662 (dingaling/gtalk/+ 14085264000@voice.google.com) Callstate Change RINGING -> ACTIVE 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:378 (dingaling/gtalk/+14085264000@voice.google.com) State CONSUME_MEDIA 2011-02-13 08:26:08.845035 [DEBUG] switch_core_state_machine.c:378 (dingaling/gtalk/+14085264000@voice.google.com) State CONSUME_MEDIA going to sleep 2011-02-13 08:26:08.845035 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/1000@202.122.99.99! 2011-02-13 08:26:08.845035 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1000@202.122.99.99 [BREAK] 2011-02-13 08:26:08.845035 [NOTICE] switch_ivr_originate.c:479 Ring Ready sofia/internal/1000@202.122.99.99! 2011-02-13 08:26:08.845035 [DEBUG] sofia.c:4659 Channel sofia/internal/ 1000@202.122.99.99 entering state [early][180] 2011-02-13 08:26:08.866018 [DEBUG] sofia_glue.c:2987 AUDIO RTP [sofia/internal/1000@202.122.99.99] 202.122.99.99 port 20318 -> 202.43.188.15 port 4516 codec: 3 ms: 20 2011-02-13 08:26:08.866971 [DEBUG] switch_rtp.c:1607 Starting timer [soft] 160 bytes per 20ms 2011-02-13 08:26:08.868036 [DEBUG] sofia_glue.c:3228 Set 2833 dtmf send payload to 100 2011-02-13 08:26:08.868036 [DEBUG] sofia_glue.c:3233 Set 2833 dtmf receive payload to 100 2011-02-13 08:26:08.868036 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/ 1000@202.122.99.99: v=0 o=FreeSWITCH 1297536450 1297536451 IN IP4 202.122.99.99 s=FreeSWITCH c=IN IP4 202.122.99.99 t=0 0 m=audio 20318 RTP/AVP 3 100 a=rtpmap:3 GSM/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1000@202.122.99.99 [BREAK] 2011-02-13 08:26:08.868036 [DEBUG] switch_channel.c:2788 (sofia/internal/ 1000@202.122.99.99) Callstate Change RINGING -> ACTIVE 2011-02-13 08:26:08.868036 [NOTICE] switch_ivr_originate.c:3363 Channel [sofia/internal/1000@202.122.99.99] has been answered 2011-02-13 08:26:08.868036 [DEBUG] switch_ivr_originate.c:3408 Originate Resulted in Success: [dingaling/gtalk/+14085264000@voice.google.com] 2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:709 Send signal dingaling/gtalk/+14085264000@voice.google.com [BREAK] 2011-02-13 08:26:08.868036 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+ 14085264000@voice.google.com CHANNEL KILL 2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1000@202.122.99.99 [BREAK] 2011-02-13 08:26:08.868036 [DEBUG] switch_ivr_bridge.c:1234 (dingaling/gtalk/+14085264000@voice.google.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2011-02-13 08:26:08.868036 [DEBUG] switch_core_session.c:1116 Send signal dingaling/gtalk/+14085264000@voice.google.com [BREAK] 2011-02-13 08:26:08.868036 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+ 14085264000@voice.google.com CHANNEL KILL 2011-02-13 08:26:08.868036 [DEBUG] sofia.c:4659 Channel sofia/internal/ 1000@202.122.99.99 entering state [completed][200] 2011-02-13 08:26:08.868036 [DEBUG] switch_core_state_machine.c:320 (dingaling/gtalk/+14085264000@voice.google.com) Running State Change CS_EXCHANGE_MEDIA 2011-02-13 08:26:08.868036 [DEBUG] switch_core_state_machine.c:369 (dingaling/gtalk/+14085264000@voice.google.com) State EXCHANGE_MEDIA 2011-02-13 08:26:08.868036 [DEBUG] mod_dingaling.c:1356 CHANNEL LOOPBACK 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT! 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.05 per minute to account 1000 2011-02-13 08:26:08.869918 [INFO] mod_nibblebill.c:485 Beginning new billing on d5cbf051-f0d5-4d02-b136-574e80d7c676 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:491 0 seconds passed since last bill time of 2011-02-13 08:26:08 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:498 Billing $0.000002 to 1000 (Call: d5cbf051-f0d5-4d02-b136-574e80d7c676 / 0.000000 so far) 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:321 Doing update query [UPDATE accounts SET cash=cash-0.000002 WHERE id='1000'] 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:366 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1000'] 2011-02-13 08:26:08.869918 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 1000 (balance = 96.581215) 2011-02-13 08:26:08.917032 [NOTICE] libdingaling.c:1368 SecSEND: ------------------------------------------------------------------------------- <iq type="result" to="+ 14085264000@voice.google.com/srvres-MTAuMTMuMTE4LjI2Ojk4MDI=" from=" bwibowo@gmail.com/gtalkE0340285" id="jingle:10.13.118.26-26031162:1:0388867F"> <query></query> </iq> 2011-02-13 08:26:09.221034 [DEBUG] sofia.c:4659 Channel sofia/internal/ 1000@202.122.99.99 entering state [ready][200] 2011-02-13 08:26:09.228033 [DEBUG] switch_core_session.c:771 Send signal dingaling/gtalk/+14085264000@voice.google.com [BREAK] 2011-02-13 08:26:09.228033 [DEBUG] mod_dingaling.c:1348 dingaling/gtalk/+ 14085264000@voice.google.com CHANNEL KILL 2011-02-13 08:26:09.228033 [DEBUG] switch_core_session.c:771 Send signal sofia/internal/1000@202.122.99.99 [BREAK] 2011-02-13 08:26:10.568036 [INFO] switch_rtp.c:2920 Auto Changing port from 202.43.188.15:4516 to 202.43.188.76:20256 2011-02-13 08:26:58.767084 [INFO] libdingaling.c:1366 SecRECV: ------------------------------------------------------------------------------- <presence from="bwibowo@gmail.com/gmail.16565A66" to="bwibowo@gmail.com"> <priority>24</priority> <caps:c node="http://mail.google.com/xmpp/client/caps" ver="1.1" ext="pmuc-v1 sms-v1 camera-v1 video-v1 voice-v1" xmlns:caps=" http://jabber.org/protocol/caps"></caps:c> <status>Ngantuk</status> <x xmlns="vcard-temp:x:update"> <photo></photo> </x> </presence> 2011-02-13 08:26:58.817032 [NOTICE] libdingaling.c:1368 SecSEND: ------------------------------------------------------------------------------- <presence type="subscribed" to="bwibowo@gmail.com"> <status>Ding A Ling....</status> </presence> 2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT! 2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.05 per minute to account 1000 2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:491 59 seconds passed since last bill time of 2011-02-13 08:26:08 2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:498 Billing $0.049998 to 1000 (Call: d5cbf051-f0d5-4d02-b136-574e80d7c676 / 0.000002 so far) 2011-02-13 08:27:08.868031 [DEBUG] mod_nibblebill.c:321 Doing update query [UPDATE accounts SET cash=cash-0.049998 WHERE id='1000'] 2011-02-13 08:27:08.868979 [DEBUG] mod_nibblebill.c:366 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='1000'] 2011-02-13 08:27:08.868979 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 1000 (balance = 96.531219) On Sun, Feb 13, 2011 at 5:49 AM, Michael Collins <msc@freeswitch.org> wrote:
On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo@gmail.com> wrote:
On previous git version I have working mod dingaling, but until today my dingaling is never working. If anybody has freeswitch source with dingaling works, please share the url. I will download
What does the console log show? Also, turn on dingaling debugging w/ the "dl_debug on" command. -MC
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Hi Michael, this is a known issue with mod_dingaling since mid January around the time of 1.0.7 tarbar release this year. I tested with different kinds of setups, the mod_dingaling always loops at mod_dingaling.c line 2941and 3275, which results in no audio (rtp packets) even though the jingle session seems to have already established. I asked for core dev's review of mod_dingaling on IRC a couple of times, looks like everybody is so busy that no one really takes a closer looks at the mod_dingaling.c yet. This behavior affects Gtalk to sip calling as well as gmail outbound call which is sip to jingle. Hope FS core dev will have some time to review this, we really appreciate that FS's gtalk calling can be back running again. Thanks, Chris On Sat, Feb 12, 2011 at 5:49 PM, Michael Collins <msc@freeswitch.org> wrote:
On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo@gmail.com> wrote:
On previous git version I have working mod dingaling, but until today my dingaling is never working. If anybody has freeswitch source with dingaling works, please share the url. I will download
What does the console log show? Also, turn on dingaling debugging w/ the "dl_debug on" command. -MC
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Chris Chen-4 wrote:
Hi Michael, this is a known issue with mod_dingaling since mid January around the time of 1.0.7 tarbar release this year. I tested with different kinds of setups, the mod_dingaling always loops at mod_dingaling.c line 2941and 3275, which results in no audio (rtp packets) even though the jingle session seems to have already established. I can concur with you on this issue.
I asked for core dev's review of mod_dingaling on IRC a couple of times,
looks like everybody is so busy that no one really takes a closer looks at the mod_dingaling.c yet. I could understand, particularly working hard with an empty stomach.
Hope FS core dev will have some time to review this, we really appreciate
that FS's gtalk calling can be back running again. Especially, after they have returned from their good meals paid for and/or donated by some of readers as shown http://freeswitch-users.2379917.n2.nabble.com/UPDATE-Buy-the-devs-dinner-tp6... here .
----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-inbound-audio-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.
I will test this on one of my linux boxen when i get a chance later today. I will keep you posted. -MC On Sun, Feb 13, 2011 at 7:49 AM, mazilo <Nabble@slickdeals.endjunk.com>wrote:
Chris Chen-4 wrote:
Hi Michael, this is a known issue with mod_dingaling since mid January around the time of 1.0.7 tarbar release this year. I tested with different kinds of setups, the mod_dingaling always loops at mod_dingaling.c line 2941and 3275, which results in no audio (rtp packets) even though the jingle session seems to have already established. I can concur with you on this issue.
|
Have you considered using JIRA? Imagine if you get 200 emails and 1200 im messages a day, if you can keep track of it in your head? On Sat, Feb 12, 2011 at 7:16 PM, Chris Chen <chris.chen2004@gmail.com> wrote:
Hi Michael, this is a known issue with mod_dingaling since mid January around the time of 1.0.7 tarbar release this year. I tested with different kinds of setups, the mod_dingaling always loops at mod_dingaling.c line 2941and 3275, which results in no audio (rtp packets) even though the jingle session seems to have already established. I asked for core dev's review of mod_dingaling on IRC a couple of times, looks like everybody is so busy that no one really takes a closer looks at the mod_dingaling.c yet. This behavior affects Gtalk to sip calling as well as gmail outbound call which is sip to jingle. Hope FS core dev will have some time to review this, we really appreciate that FS's gtalk calling can be back running again. Thanks, Chris
On Sat, Feb 12, 2011 at 5:49 PM, Michael Collins <msc@freeswitch.org> wrote:
On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo@gmail.com> wrote:
On previous git version I have working mod dingaling, but until today my dingaling is never working. If anybody has freeswitch source with dingaling works, please share the url. I will download
What does the console log show? Also, turn on dingaling debugging w/ the "dl_debug on" command. -MC
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Oh, well I found it i think, again I would have preferred a JIRA ticket. it's hard to juggle issues from people over email. On Mon, Feb 14, 2011 at 11:59 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Have you considered using JIRA? Imagine if you get 200 emails and 1200 im messages a day, if you can keep track of it in your head?
On Sat, Feb 12, 2011 at 7:16 PM, Chris Chen <chris.chen2004@gmail.com> wrote:
Hi Michael, this is a known issue with mod_dingaling since mid January around the time of 1.0.7 tarbar release this year. I tested with different kinds of setups, the mod_dingaling always loops at mod_dingaling.c line 2941and 3275, which results in no audio (rtp packets) even though the jingle session seems to have already established. I asked for core dev's review of mod_dingaling on IRC a couple of times, looks like everybody is so busy that no one really takes a closer looks at the mod_dingaling.c yet. This behavior affects Gtalk to sip calling as well as gmail outbound call which is sip to jingle. Hope FS core dev will have some time to review this, we really appreciate that FS's gtalk calling can be back running again. Thanks, Chris
On Sat, Feb 12, 2011 at 5:49 PM, Michael Collins <msc@freeswitch.org> wrote:
On Sat, Feb 12, 2011 at 1:27 AM, Budi wibowo <bwibowo@gmail.com> wrote:
On previous git version I have working mod dingaling, but until today my dingaling is never working. If anybody has freeswitch source with dingaling works, please share the url. I will download
What does the console log show? Also, turn on dingaling debugging w/ the "dl_debug on" command. -MC
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-- Anthony Minessale II
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Anthony Minessale wrote:
Oh, well I found it i think, ...
By disabling VAD? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-inbound-audio-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.
no On Mon, Feb 14, 2011 at 2:43 PM, mazilo <Nabble@slickdeals.endjunk.com> wrote:
Anthony Minessale wrote:
Oh, well I found it i think, ...
By disabling VAD?
----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-inbound-audio-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Was there any update after this reply of 'no' ? I dont see any JIRA# also where I can find the solution. please update this. I am getting similar issue where on a inbound call gtalk not sending audio RTP packets. I will appreciate update on this. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-inbound-audio-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.
I have been trying to do the same. After a few hiccups and some help on the IRC channel I was able to get outbound working this morning. I didn't have to mess with any rtp settings though (which could explain my problem below). However I am having issues with inbound. It seems like it is trying but I get a strange error in the log regarding a missing port (http://pastebin.freeswitch.org/15311). - Daniel Wells From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Locutis of Borg Sent: Tuesday, February 08, 2011 12:07 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] mod_dingaling & inbound audio Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio. FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem In and out calling on SIP to provider works fine in both cases. So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users? Thank you for you help. "Resistance is Fruitful"
participants (9)
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Anthony Minessale
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budi wibowo
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Budi wibowo
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Chris Chen
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Daniel Wells
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Locutis of Borg
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mazilo
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Michael Collins
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sanjay