Bridge to other FS server has no audio until DTMF
I started a new thread in case anyone muted it... it wasn't simply a network issue. It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B. Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue... I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update. Here's the log on freeswitch B: EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms 2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something. Any ideas as to the root cause of this? -Avi Marcus ---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org> Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again. - FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU). FS server B is to join a conference if that matters. I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU. I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please? -Avi Marcus
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything. On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated. <action application="bridge" data="sofia/external/number@yyy.bestfone.com "/> (And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.) -Avi Marcus On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer). What you’re describing seems different to me. On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/number@yyy.bestfone.com "/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2. What api should I call with api on answer..? On Thu, Oct 7, 2021, 3:19 PM David Villasmil <david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/number@yyy.bestfone.com "/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
execure_on_answer=playback::silence_stream://100 should solve it. /b PS, the non pc term that this has been said to be is https://en.wikipedia.org/wiki/Mexican_standoff On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi@avimarcus.net> wrote:
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2.
What api should I call with api on answer..?
On Thu, Oct 7, 2021, 3:19 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/ number@yyy.bestfone.com"/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO ] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
That’s the one! On Thu, 7 Oct 2021 at 16:11, Brian West <brian@freeswitch.com> wrote:
execure_on_answer=playback::silence_stream://100 should solve it.
/b PS, the non pc term that this has been said to be is https://en.wikipedia.org/wiki/Mexican_standoff
On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi@avimarcus.net> wrote:
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2.
What api should I call with api on answer..?
On Thu, Oct 7, 2021, 3:19 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/ number@yyy.bestfone.com"/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO ] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
I had to do this to get it to execute on the B leg: <action application="export" data="nolocal:execute_on_answer=playback silence_stream://100"/> ... but it didn't help. Only DTMF worked... either manually dialed or via queue_dtmf Freeswitch A waited for my DTMF to actually send the silence. Version 1.10.6 -release-18-1ff9d0a60e 64bit 2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort noise payload to 13 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/ JOIN_CLASS_7229999@voip.bestfone.com] has been answered EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999@voip.bestfone.com playback(silence_stream://100) 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms -- 20 seconds later when I pressed a button -- 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done playing file silence_stream://100 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/ JOIN_CLASS_7229999@voip.bestfone.com) Callstate Change DOWN -> ACTIVE 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) Running State Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet for [5] ts=960 dur=160/160/2000 seq=26795 lw=960 This seemingly shouldn't be an issue. FS1 already has active media from the A leg, so it should initiate to the B leg. The B leg has been instructed to play a file, so it should initiate to the A leg... But if this is somehow unavoidable, perhaps we need a workaround config, where we have a simple variable in the bridge string to avoid the standoff? -Avi Marcus On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian@freeswitch.com> wrote:
execure_on_answer=playback::silence_stream://100 should solve it.
/b PS, the non pc term that this has been said to be is https://en.wikipedia.org/wiki/Mexican_standoff
On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi@avimarcus.net> wrote:
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2.
What api should I call with api on answer..?
On Thu, Oct 7, 2021, 3:19 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/ number@yyy.bestfone.com"/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
I started a new thread in case anyone muted it... it wasn't simply a network issue.
It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.
Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...
I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.
Here's the log on freeswitch B:
EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG ] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO ] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something.
Any ideas as to the root cause of this?
-Avi Marcus
---------- Forwarded message --------- From: Avi Marcus <avi@avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org>
Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.
- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU).
FS server B is to join a conference if that matters.
I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU.
I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?
-Avi Marcus
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
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-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
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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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Is there a way to "fix" the standoff? Perhaps a header to send or a channel variable to set? I'd like to do bypass_media to cut one of my servers out of the media path, but then I won't be able to queue an rfc2833 digit press. Thanks, -Avi Marcus BestFone On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus <avi@avimarcus.net> wrote:
I had to do this to get it to execute on the B leg: <action application="export" data="nolocal:execute_on_answer=playback silence_stream://100"/>
... but it didn't help. Only DTMF worked... either manually dialed or via queue_dtmf Freeswitch A waited for my DTMF to actually send the silence. Version 1.10.6 -release-18-1ff9d0a60e 64bit
2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort noise payload to 13 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/ JOIN_CLASS_7229999@voip.bestfone.com] has been answered EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999@voip.bestfone.com playback(silence_stream://100) 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
-- 20 seconds later when I pressed a button --
2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done playing file silence_stream://100 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/ JOIN_CLASS_7229999@voip.bestfone.com) Callstate Change DOWN -> ACTIVE 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) Running State Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet for [5] ts=960 dur=160/160/2000 seq=26795 lw=960
This seemingly shouldn't be an issue. FS1 already has active media from the A leg, so it should initiate to the B leg. The B leg has been instructed to play a file, so it should initiate to the A leg... But if this is somehow unavoidable, perhaps we need a workaround config, where we have a simple variable in the bridge string to avoid the standoff?
-Avi Marcus
On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian@freeswitch.com> wrote:
execure_on_answer=playback::silence_stream://100 should solve it.
/b PS, the non pc term that this has been said to be is https://en.wikipedia.org/wiki/Mexican_standoff
On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi@avimarcus.net> wrote:
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2.
What api should I call with api on answer..?
On Thu, Oct 7, 2021, 3:19 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/ number@yyy.bestfone.com"/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
I seem to remember Brian saying this was because FS is waiting for the remote end to send audio before starting itself. I believe he recommended sending an empty (silence) to force the audio stream to be sent even if fs hasn’t received anything.
On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote:
> I started a new thread in case anyone muted it... it wasn't simply a > network issue. > > It seems the bridging occurs and dialplan processes, but no media > flows - until DTMF from the A-leg. > Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> > freeswitch B. > > Calls directly from carrier to Freeswitch B are fine. > Calls from a different carrier to Freeswitch A -> media and IVR -> > Freeswitch B are also fine. > So it sounds like a carrier/unique SIP/RTP issue, but since FS is in > the media path, it's an FS issue... > > > I actually mcguyvered this right now with a queue_dtmf before the > bridge, to force the audio stream to update. > > Here's the log on freeswitch B: > > EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 > log(DEBUG class chosen: 1234567) > 2021-10-07 09:16:24.343175 [DEBUG > ] mod_dptools.c:1879 class chosen: 1234567 > EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 > javascript(conference/lookupAndJoinConference.js 1234567) > EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 > playback(class/hold-wait-teacher.wav) > 2021-10-07 09:16:24.363379 [DEBUG > ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 > entering state [completed][200] > 2021-10-07 09:16:24.363379 [DEBUG > ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 > entering state [ready][200] > 2021-10-07 09:16:24.363379 [DEBUG > ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz > 1 channels 20ms > > > > > 2021-10-07 09:16:34.903283 [DEBUG > ] switch_rtp.c:7793 Correct audio ip/port confirmed. > 2021-10-07 09:16:34.923190 [DEBUG > ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 > 2021-10-07 09:16:34.923190 [INFO > ] switch_channel.c:522 RECV DTMF 3:2080 > 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 > 2021-10-07 09:16:37.143169 [DEBUG > ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav > > > You can see a 10 second gap between call ready 200 and correct > audio/ip and file done playing (it's a 2 second file), and this doesn't > happen automatically, only when I choose to press something. > > > Any ideas as to the root cause of this? > > > -Avi Marcus > > ---------- Forwarded message --------- > From: Avi Marcus <avi@avimarcus.net> > Date: Wed, Oct 6, 2021 at 3:32 PM > Subject: Bridge to other FS server has no audio ??? > To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org> > > > Any ideas on why a call doesn't have media? It used to work, but I > think my upstream changed his SDP again. > > - FreeSWITCH Server A - call comes in and bypass_media bridges to FS > server B. Media works. > - FreeSWITCH Server A - call comes in and bridges to FS server B > (not on bypass). Media works. > - FreeSWITCH Server A - call comes in, gets answered, then bridges > to FS server B. Call looks OK, but no media is flowing (I don't hear > anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All > the same codecs are set in the json cdrs (PCMU). > > FS server B is to join a conference if that matters. > > I was assuming it had to do with codecs, but setting > absolute_codec_string to PCMU doesn't make any difference in the logs - > it's already always PCMU. > > I have NO clue what further could cause this other than codecs, > which seem to be fine. Any ideas please? > > > -Avi Marcus > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales@freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
First off, don't use export and don't use nolocal, just set it and see what happens. /b On Mon, Nov 8, 2021 at 12:36 PM Avi Marcus <avi@avimarcus.net> wrote:
Is there a way to "fix" the standoff? Perhaps a header to send or a channel variable to set?
I'd like to do bypass_media to cut one of my servers out of the media path, but then I won't be able to queue an rfc2833 digit press.
Thanks, -Avi Marcus BestFone
On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus <avi@avimarcus.net> wrote:
I had to do this to get it to execute on the B leg: <action application="export" data="nolocal:execute_on_answer=playback silence_stream://100"/>
... but it didn't help. Only DTMF worked... either manually dialed or via queue_dtmf Freeswitch A waited for my DTMF to actually send the silence. Version 1.10.6 -release-18-1ff9d0a60e 64bit
2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort noise payload to 13 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/ JOIN_CLASS_7229999@voip.bestfone.com] has been answered EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999@voip.bestfone.com playback(silence_stream://100) 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
-- 20 seconds later when I pressed a button --
2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done playing file silence_stream://100 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/ JOIN_CLASS_7229999@voip.bestfone.com) Callstate Change DOWN -> ACTIVE 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) Running State Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet for [5] ts=960 dur=160/160/2000 seq=26795 lw=960
This seemingly shouldn't be an issue. FS1 already has active media from the A leg, so it should initiate to the B leg. The B leg has been instructed to play a file, so it should initiate to the A leg... But if this is somehow unavoidable, perhaps we need a workaround config, where we have a simple variable in the bridge string to avoid the standoff?
-Avi Marcus
On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian@freeswitch.com> wrote:
execure_on_answer=playback::silence_stream://100 should solve it.
/b PS, the non pc term that this has been said to be is https://en.wikipedia.org/wiki/Mexican_standoff
On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi@avimarcus.net> wrote:
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2.
What api should I call with api on answer..?
On Thu, Oct 7, 2021, 3:19 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
I'm using dialplan bridge, so then the dialplan is over. How do I send silence after the bridge...? An api_on_answer with a uuid_broadcast.. seems overly complicated.
<action application="bridge" data="sofia/external/ number@yyy.bestfone.com"/>
(And I don't know why there isn't audio - I had to set up an audio to get to this options in the IVR... so there's already audio. And Server B also started a file playback so should have initiated audio.)
-Avi Marcus
On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
> I seem to remember Brian saying this was because FS is waiting for > the remote end to send audio before starting itself. I believe he > recommended sending an empty (silence) to force the audio stream to be sent > even if fs hasn’t received anything. > > On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote: > >> I started a new thread in case anyone muted it... it wasn't simply >> a network issue. >> >> It seems the bridging occurs and dialplan processes, but no media >> flows - until DTMF from the A-leg. >> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> >> freeswitch B. >> >> Calls directly from carrier to Freeswitch B are fine. >> Calls from a different carrier to Freeswitch A -> media and IVR -> >> Freeswitch B are also fine. >> So it sounds like a carrier/unique SIP/RTP issue, but since FS is >> in the media path, it's an FS issue... >> >> >> I actually mcguyvered this right now with a queue_dtmf before the >> bridge, to force the audio stream to update. >> >> Here's the log on freeswitch B: >> >> EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 >> log(DEBUG class chosen: 1234567) >> 2021-10-07 09:16:24.343175 [DEBUG >> ] mod_dptools.c:1879 class chosen: 1234567 >> EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 >> javascript(conference/lookupAndJoinConference.js 1234567) >> EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 >> playback(class/hold-wait-teacher.wav) >> 2021-10-07 09:16:24.363379 [DEBUG >> ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 >> entering state [completed][200] >> 2021-10-07 09:16:24.363379 [DEBUG >> ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 >> entering state [ready][200] >> 2021-10-07 09:16:24.363379 [DEBUG >> ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz >> 1 channels 20ms >> >> >> >> >> 2021-10-07 09:16:34.903283 [DEBUG >> ] switch_rtp.c:7793 Correct audio ip/port confirmed. >> 2021-10-07 09:16:34.923190 [DEBUG >> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 >> 2021-10-07 09:16:34.923190 [INFO >> ] switch_channel.c:522 RECV DTMF 3:2080 >> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 >> 2021-10-07 09:16:37.143169 [DEBUG >> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav >> >> >> You can see a 10 second gap between call ready 200 and correct >> audio/ip and file done playing (it's a 2 second file), and this doesn't >> happen automatically, only when I choose to press something. >> >> >> Any ideas as to the root cause of this? >> >> >> -Avi Marcus >> >> ---------- Forwarded message --------- >> From: Avi Marcus <avi@avimarcus.net> >> Date: Wed, Oct 6, 2021 at 3:32 PM >> Subject: Bridge to other FS server has no audio ??? >> To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org> >> >> >> Any ideas on why a call doesn't have media? It used to work, but I >> think my upstream changed his SDP again. >> >> - FreeSWITCH Server A - call comes in and bypass_media bridges to >> FS server B. Media works. >> - FreeSWITCH Server A - call comes in and bridges to FS server B >> (not on bypass). Media works. >> - FreeSWITCH Server A - call comes in, gets answered, then bridges >> to FS server B. Call looks OK, but no media is flowing (I don't hear >> anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All >> the same codecs are set in the json cdrs (PCMU). >> >> FS server B is to join a conference if that matters. >> >> I was assuming it had to do with codecs, but setting >> absolute_codec_string to PCMU doesn't make any difference in the logs - >> it's already always PCMU. >> >> I have NO clue what further could cause this other than codecs, >> which seem to be fine. Any ideas please? >> >> >> -Avi Marcus >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire >> https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales@freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://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 >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work@gmail.com > phone: +34669448337 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales@freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch> _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
<anti-action application="set" data="execute_on_answer=playback silence_stream://100"/> ... I'm not seeing it executing in the logs for leg A or leg B. Maybe because leg A is already answered? I tried it again as an export, and upping it to 500ms, but it didn't force the audio to flow. So far only DTMF has worked. -Avi Marcus On Fri, Nov 12, 2021 at 10:38 PM Brian West <brian@freeswitch.com> wrote:
First off, don't use export and don't use nolocal, just set it and see what happens.
/b
On Mon, Nov 8, 2021 at 12:36 PM Avi Marcus <avi@avimarcus.net> wrote:
Is there a way to "fix" the standoff? Perhaps a header to send or a channel variable to set?
I'd like to do bypass_media to cut one of my servers out of the media path, but then I won't be able to queue an rfc2833 digit press.
Thanks, -Avi Marcus BestFone
On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus <avi@avimarcus.net> wrote:
I had to do this to get it to execute on the B leg: <action application="export" data="nolocal:execute_on_answer=playback silence_stream://100"/>
... but it didn't help. Only DTMF worked... either manually dialed or via queue_dtmf Freeswitch A waited for my DTMF to actually send the silence. Version 1.10.6 -release-18-1ff9d0a60e 64bit
2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort noise payload to 13 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/JOIN_CLASS_7229999@voip.bestfone.com] has been answered EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999@voip.bestfone.com playback(silence_stream://100) 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms
-- 20 seconds later when I pressed a button --
2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done playing file silence_stream://100 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) Callstate Change DOWN -> ACTIVE 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) Running State Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090) 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654 (sofia/external/JOIN_CLASS_7229999@voip.bestfone.com) State EXCHANGE_MEDIA 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet for [5] ts=960 dur=160/160/2000 seq=26795 lw=960
This seemingly shouldn't be an issue. FS1 already has active media from the A leg, so it should initiate to the B leg. The B leg has been instructed to play a file, so it should initiate to the A leg... But if this is somehow unavoidable, perhaps we need a workaround config, where we have a simple variable in the bridge string to avoid the standoff?
-Avi Marcus
On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian@freeswitch.com> wrote:
execure_on_answer=playback::silence_stream://100 should solve it.
/b PS, the non pc term that this has been said to be is https://en.wikipedia.org/wiki/Mexican_standoff
On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi@avimarcus.net> wrote:
I meant there's audio from pstn to fs1, but indeed I'm observing no audio between fs1 and fs2.
What api should I call with api on answer..?
On Thu, Oct 7, 2021, 3:19 PM David Villasmil < david.villasmil.work@gmail.com> wrote:
If you see rtp glowing both ways, then this is not the stalemate I was talking about. The scenario I’m referring to is about FS not starting sending rtp waiting for the other side to start sending, and the other side doing the same thing, thus going into a stalemate. This is solved by injecting a silence (I would do api_on_answer).
What you’re describing seems different to me.
On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi@avimarcus.net> wrote:
> I'm using dialplan bridge, so then the dialplan is over. How do I > send silence after the bridge...? An api_on_answer with a uuid_broadcast.. > seems overly complicated. > > <action application="bridge" data="sofia/external/ > number@yyy.bestfone.com"/> > > > (And I don't know why there isn't audio - I had to set up an audio > to get to this options in the IVR... so there's already audio. And Server B > also started a file playback so should have initiated audio.) > > > -Avi Marcus > > On Thu, Oct 7, 2021 at 1:41 PM David Villasmil < > david.villasmil.work@gmail.com> wrote: > >> I seem to remember Brian saying this was because FS is waiting for >> the remote end to send audio before starting itself. I believe he >> recommended sending an empty (silence) to force the audio stream to be sent >> even if fs hasn’t received anything. >> >> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi@avimarcus.net> wrote: >> >>> I started a new thread in case anyone muted it... it wasn't simply >>> a network issue. >>> >>> It seems the bridging occurs and dialplan processes, but no media >>> flows - until DTMF from the A-leg. >>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> >>> freeswitch B. >>> >>> Calls directly from carrier to Freeswitch B are fine. >>> Calls from a different carrier to Freeswitch A -> media and IVR -> >>> Freeswitch B are also fine. >>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is >>> in the media path, it's an FS issue... >>> >>> >>> I actually mcguyvered this right now with a queue_dtmf before the >>> bridge, to force the audio stream to update. >>> >>> Here's the log on freeswitch B: >>> >>> EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 >>> log(DEBUG class chosen: 1234567) >>> 2021-10-07 09:16:24.343175 [DEBUG >>> ] mod_dptools.c:1879 class chosen: 1234567 >>> EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 >>> javascript(conference/lookupAndJoinConference.js 1234567) >>> EXECUTE [depth=0] sofia/external/972581234567@172.123.123.123 >>> playback(class/hold-wait-teacher.wav) >>> 2021-10-07 09:16:24.363379 [DEBUG >>> ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 >>> entering state [completed][200] >>> 2021-10-07 09:16:24.363379 [DEBUG >>> ] sofia.c:7406 Channel sofia/external/972581234567@172.123.123.123 >>> entering state [ready][200] >>> 2021-10-07 09:16:24.363379 [DEBUG >>> ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz >>> 1 channels 20ms >>> >>> >>> >>> >>> 2021-10-07 09:16:34.903283 [DEBUG >>> ] switch_rtp.c:7793 Correct audio ip/port confirmed. >>> 2021-10-07 09:16:34.923190 [DEBUG >>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080 >>> 2021-10-07 09:16:34.923190 [INFO >>> ] switch_channel.c:522 RECV DTMF 3:2080 >>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 >>> 2021-10-07 09:16:37.143169 [DEBUG >>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav >>> >>> >>> You can see a 10 second gap between call ready 200 and correct >>> audio/ip and file done playing (it's a 2 second file), and this doesn't >>> happen automatically, only when I choose to press something. >>> >>> >>> Any ideas as to the root cause of this? >>> >>> >>> -Avi Marcus >>> >>> ---------- Forwarded message --------- >>> From: Avi Marcus <avi@avimarcus.net> >>> Date: Wed, Oct 6, 2021 at 3:32 PM >>> Subject: Bridge to other FS server has no audio ??? >>> To: FreeSWITCH Users Help <FreeSWITCH-users@lists.freeswitch.org> >>> >>> >>> Any ideas on why a call doesn't have media? It used to work, but I >>> think my upstream changed his SDP again. >>> >>> - FreeSWITCH Server A - call comes in and bypass_media bridges to >>> FS server B. Media works. >>> - FreeSWITCH Server A - call comes in and bridges to FS server B >>> (not on bypass). Media works. >>> - FreeSWITCH Server A - call comes in, gets answered, then bridges >>> to FS server B. Call looks OK, but no media is flowing (I don't hear >>> anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All >>> the same codecs are set in the json cdrs (PCMU). >>> >>> FS server B is to join a conference if that matters. >>> >>> I was assuming it had to do with codecs, but setting >>> absolute_codec_string to PCMU doesn't make any difference in the logs - >>> it's already always PCMU. >>> >>> I have NO clue what further could cause this other than codecs, >>> which seem to be fine. Any ideas please? >>> >>> >>> -Avi Marcus >>> >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire >>> https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and >>> PSTN services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales@freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://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 >>> https://freeswitch.com >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work@gmail.com >> phone: +34669448337 >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire >> https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales@freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://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 >> https://freeswitch.com > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales@freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
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participants (3)
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Avi Marcus
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Brian West
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David Villasmil