FreeSwitch and Yealink
Hi guys, Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up. short FS LOG: switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going. Internal extension to extension works fine (even if the extensions are at a different physical location and subnet). I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have). I have updated firwmare to their latest version (which in the comments say freeswitch ready) Wondering if anyone else had any experience with these, or has some thoughts? Thanks Paul
Hangups with the cause you describe can be caused by a failed codec negotiation. Are you sure it's not the case ? Does the remote phone send a sip "not acceptable here" ? If needed you can force the selection of the "outgoing" codec: http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string ivan On 08/07/2013 07:44 AM, Paul wrote:
Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Ivan thanks for your reply. I'm willing to try anything :) Where would I set those variables? In the dialplan out to the sip trunk? Thanks Paul On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan@c3i.bg> wrote:
Hangups with the cause you describe can be caused by a failed codec negotiation. Are you sure it's not the case ? Does the remote phone send a sip "not acceptable here" ?
If needed you can force the selection of the "outgoing" codec:
http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string
ivan
On 08/07/2013 07:44 AM, Paul wrote:
Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You have to set the variable in the dialplan, before bridging. You can also set it directly in the bridge command. More details at: http://wiki.freeswitch.org/wiki/Codec_Negotiation But the only way to really know what is going on is to enable siptrace on your sofia profile and see what is offered before your endpoint hangups (if you post your debug, please post on freeswitch's pastebin). ivan On 08/07/2013 08:59 AM, Paul wrote:
Ivan thanks for your reply.
I'm willing to try anything :)
Where would I set those variables? In the dialplan out to the sip trunk?
Thanks
Paul
On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan@c3i.bg> wrote:
Hangups with the cause you describe can be caused by a failed codec negotiation. Are you sure it's not the case ? Does the remote phone send a sip "not acceptable here" ? If needed you can force the selection of the "outgoing" codec: http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string ivan On 08/07/2013 07:44 AM, Paul wrote:
Hi guys, Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up. short FS LOG: switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going. Internal extension to extension works fine (even if the extensions are at a different physical location and subnet). I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have). I have updated firwmare to their latest version (which in the comments say freeswitch ready) Wondering if anyone else had any experience with these, or has some thoughts? Thanks Paul _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Try to set verbose_sdp=true in vars.xml. I just experienced a problem with a China phone using Broadcom chipset. /Nandy On Fri, Aug 9, 2013 at 1:01 AM, Ivan <ivan@c3i.bg> wrote:
You have to set the variable in the dialplan, before bridging. You can also set it directly in the bridge command.
More details at:
http://wiki.freeswitch.org/wiki/Codec_Negotiation
But the only way to really know what is going on is to enable siptrace on your sofia profile and see what is offered before your endpoint hangups (if you post your debug, please post on freeswitch's pastebin).
ivan
On 08/07/2013 08:59 AM, Paul wrote:
Ivan thanks for your reply.
I'm willing to try anything :)
Where would I set those variables? In the dialplan out to the sip trunk?
Thanks
Paul
On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan@c3i.bg> wrote:
Hangups with the cause you describe can be caused by a failed codec negotiation. Are you sure it's not the case ? Does the remote phone send a sip "not acceptable here" ? If needed you can force the selection of the "outgoing" codec: http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_stringivan On 08/07/2013 07:44 AM, Paul wrote:
Hi guys, Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up. short FS LOG: switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going. Internal extension to extension works fine (even if the extensions are at a different physical location and subnet). I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have). I have updated firwmare to their latest version (which in the comments say freeswitch ready) Wondering if anyone else had any experience with these, or has some thoughts? Thanks Paul
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Paul,
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Get a real capture of the traffic with for example ngrep, dumpcap or tcpdump. # ngrep -d tun0 -O /directory/file/save.here.cap or # dumpcap -i tun0 -w /directory/file/save.here.cap Afterwards, analyse the capture with wireshark. It will give you all the information you need to analyse your issue. -- EL
Check RTP Packet Size too. -- Rogoshchenkov Nikolay On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha@prosperity4ever.com> wrote:
Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900[CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I'm gonna try all of your guys' suggestions. Ivan, I do not think its codec related, I made /log 11 and siptrace on all profiles (internal/external) as well as pcap capture of the phone itself. For some strange reason the moment the PBX send a packe tot ye Yealink phone that the remote party picked up and the call should be bridges, the Yealink phone replies with a BYE packet, I'm thinking it might be a bug in the Yealink firmware, so needless to say I have gotten a hold of their support and opened a ticket, see if something comes of that, but meanwhile going to try some of the suggestions in this thread and see if I can get any close to work around. Paul On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov <nickolayr@gmail.com> wrote:
Check RTP Packet Size too.
-- Rogoshchenkov Nikolay
On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha@prosperity4ever.com> wrote:
Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
What phone model and yealink firmware are you using? On Aug 9, 2013, at 9:01 PM, Paul <pasha@prosperity4ever.com> wrote:
I'm gonna try all of your guys' suggestions.
Ivan, I do not think its codec related, I made /log 11 and siptrace on all profiles (internal/external) as well as pcap capture of the phone itself. For some strange reason the moment the PBX send a packe tot ye Yealink phone that the remote party picked up and the call should be bridges, the Yealink phone replies with a BYE packet, I'm thinking it might be a bug in the Yealink firmware, so needless to say I have gotten a hold of their support and opened a ticket, see if something comes of that, but meanwhile going to try some of the suggestions in this thread and see if I can get any close to work around.
Paul
On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov <nickolayr@gmail.com> wrote:
Check RTP Packet Size too.
-- Rogoshchenkov Nikolay
On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha@prosperity4ever.com> wrote: Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
This is happening on both T38Gs and T32Gs, I have physical access to the T32, firmware 32.70.0.132 Paul On Mon, 12 Aug, 2013 at 8:21 AM, Michael Jerris <mike@jerris.com> wrote:
What phone model and yealink firmware are you using?
On Aug 9, 2013, at 9:01 PM, Paul <pasha@prosperity4ever.com> wrote:
I'm gonna try all of your guys' suggestions.
Ivan, I do not think its codec related, I made /log 11 and siptrace on all profiles (internal/external) as well as pcap capture of the phone itself. For some strange reason the moment the PBX send a packe tot ye Yealink phone that the remote party picked up and the call should be bridges, the Yealink phone replies with a BYE packet, I'm thinking it might be a bug in the Yealink firmware, so needless to say I have gotten a hold of their support and opened a ticket, see if something comes of that, but meanwhile going to try some of the suggestions in this thread and see if I can get any close to work around.
Paul
On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov <nickolayr@gmail.com> wrote:
Check RTP Packet Size too.
-- Rogoshchenkov Nikolay
On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha@prosperity4ever.com> wrote:
Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
I have had zero FreeSwitch problems with the T32 (18 months now) and new T46 but I don't use OpenVPN. Try Yealink tech support, they have been great and fast at fixing issues (rings, dimmer, etc.) for me, usually less than a week. BTW, the phone have great logging and pcap built in, I would test and send that to Yealink. Mario G On Aug 12, 2013, at 2:20 PM, Paul <pasha@prosperity4ever.com> wrote:
This is happening on both T38Gs and T32Gs, I have physical access to the T32, firmware 32.70.0.132
Paul
On Mon, 12 Aug, 2013 at 8:21 AM, Michael Jerris <mike@jerris.com> wrote:
What phone model and yealink firmware are you using?
On Aug 9, 2013, at 9:01 PM, Paul <pasha@prosperity4ever.com> wrote:
I'm gonna try all of your guys' suggestions.
Ivan, I do not think its codec related, I made /log 11 and siptrace on all profiles (internal/external) as well as pcap capture of the phone itself. For some strange reason the moment the PBX send a packe tot ye Yealink phone that the remote party picked up and the call should be bridges, the Yealink phone replies with a BYE packet, I'm thinking it might be a bug in the Yealink firmware, so needless to say I have gotten a hold of their support and opened a ticket, see if something comes of that, but meanwhile going to try some of the suggestions in this thread and see if I can get any close to work around.
Paul
On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov <nickolayr@gmail.com> wrote:
Check RTP Packet Size too.
-- Rogoshchenkov Nikolay
On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha@prosperity4ever.com> wrote: Hi guys,
Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
short FS LOG:
switch_ivr_bridge.c:475 Channel [sofia/internal/105@10.0.0.34] has been answered sofia.c:6528 Hangup sofia/internal/105@10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
I have updated firwmare to their latest version (which in the comments say freeswitch ready)
Wondering if anyone else had any experience with these, or has some thoughts?
Thanks
Paul
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participants (7)
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EL
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Ivan
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Mario G
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Michael Jerris
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Nandy Dagondon
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Nikolay Rogoshchenkov
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Paul