No Video Feed

Just did a fresh install of the latest FreePBX and Asterisk 13. I got everything working fine except for the video calling. I enabled video calling in the SIP settings and selected the right codecs for my phone (h264, h263, h263p). I also tried adding these codecs manually within each extension.

When I call another phone the audio works great but I only get a black screen from their end and vice versa. I can see my video feed just fine on my phone. Does anyone know what would prevent the video feed from coming through?

Are both phones on the same PBX or are they geographically separated (different networks, PBXes, firewalls in the way, etc)?

They are on the same PBX. I have the phones right next to each other in fact.

What kind of video end-points are you using?
Is the PBX on the same LAN as the end-points?

If there are any firewalls between the devices, (Windows Firewall, etc) try disabling them to see if this yields a different result. Also some firewalls (Sonicwall) require H.323 Transformations to be turned on, for NATing purposed.

The PBX is on the same LAN as the end-points. I’m using the Yealink T49G phones. I’ll try disabling the firewall in FreePBX.

Are you using the Endpoint Manager to configure your phones?

I vaguely recall that there’s a special way you have to hold your tongue when you configure the Yealink phones to get this to work. That’s all I’ve got for you though. There have been a couple of threads about these phones on here in the past couple of weeks.

Turned the firewall off and still get the black screen.

Yes I’m using endpoint manager. I don’t see any settings that would affect video though.

Start a call trace and watch to see which codecs are chosen for the connection. Even though you have the video codecs in the list, the system may be negotiating a codec that doesn’t have video support.

With the phones and PBX on the same LAN, the firewall stuff is all red herrings.

You’ve asked us the same question three times now and still haven’t provided anything that can help troubleshoot your problem. We need a call trace and an extract from the ‘/var/log/asterisk/full’ log file before anyone is going to be able to give you something you can work with.

This is going to be a problem in your configuration. Without logs or traces, we’ll never know what setting you messed up.

Still my first week using FreePBX. I’ll look into how I do all that =P

I think I did a call trace but I’m not sure =P This is a lot of text!

---
[2016-04-12 16:19:40] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK25853b13;rport=5061
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Call-ID: [email protected]
CSeq: 110 INFO
Contact: <sip:[email protected]:5060>
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:40] VERBOSE[1898] chan_sip.c: --- (9 headers 0 lines) ---
[2016-04-12 16:19:40] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK4124881538
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 11 INFO
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="41d74decb8187611e3706c6fe3621684", algorithm=MD5
Content-Type: application/media_control+xml
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
[2016-04-12 16:19:40] VERBOSE[1898] chan_sip.c: --- (13 headers 1 lines) ---
[2016-04-12 16:19:40] VERBOSE[1898][C-0000000a] chan_sip.c: Receiving INFO!
[2016-04-12 16:19:40] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK4124881538;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 11 INFO
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:42] VERBOSE[16442][C-0000000a] chan_sip.c: Reliably Transmitting (NAT) to 10.1.10.33:5060:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK663cec31;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 111 INFO
User-Agent: FPBX-13.0.101(13.7.1)
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---
[2016-04-12 16:19:42] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK663cec31;rport=5061
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Call-ID: [email protected]
CSeq: 111 INFO
Contact: <sip:[email protected]:5060>
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:42] VERBOSE[1898] chan_sip.c: --- (9 headers 0 lines) ---
[2016-04-12 16:19:42] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK677755004
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 12 INFO
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="41d74decb8187611e3706c6fe3621684", algorithm=MD5
Content-Type: application/media_control+xml
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
[2016-04-12 16:19:42] VERBOSE[1898] chan_sip.c: --- (13 headers 1 lines) ---
[2016-04-12 16:19:42] VERBOSE[1898][C-0000000a] chan_sip.c: Receiving INFO!
[2016-04-12 16:19:42] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK677755004;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 12 INFO
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:44] NOTICE[1898] chan_sip.c: -- Re-registration for [email protected]
[2016-04-12 16:19:44] NOTICE[1898] chan_sip.c: Outbound Registration: Expiry for trunking.voipdnsservers.com is 46 sec (Scheduling reregistration in 31 s)
[2016-04-12 16:19:44] VERBOSE[16442][C-0000000a] chan_sip.c: Reliably Transmitting (NAT) to 10.1.10.33:5060:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK09b3a78c;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 112 INFO
User-Agent: FPBX-13.0.101(13.7.1)
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---
[2016-04-12 16:19:44] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK09b3a78c;rport=5061
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Call-ID: [email protected]
CSeq: 112 INFO
Contact: <sip:[email protected]:5060>
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:44] VERBOSE[1898] chan_sip.c: --- (9 headers 0 lines) ---
[2016-04-12 16:19:44] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK981026496
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 13 INFO
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="41d74decb8187611e3706c6fe3621684", algorithm=MD5
Content-Type: application/media_control+xml
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
[2016-04-12 16:19:44] VERBOSE[1898] chan_sip.c: --- (13 headers 1 lines) ---
[2016-04-12 16:19:44] VERBOSE[1898][C-0000000a] chan_sip.c: Receiving INFO!
[2016-04-12 16:19:44] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK981026496;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 13 INFO
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:46] VERBOSE[16442][C-0000000a] chan_sip.c: Reliably Transmitting (NAT) to 10.1.10.33:5060:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK01789fed;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 113 INFO
User-Agent: FPBX-13.0.101(13.7.1)
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---
[2016-04-12 16:19:46] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK01789fed;rport=5061
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Call-ID: [email protected]
CSeq: 113 INFO
Contact: <sip:[email protected]:5060>
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:46] VERBOSE[1898] chan_sip.c: --- (9 headers 0 lines) ---
[2016-04-12 16:19:46] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK1961126868
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 14 INFO
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="41d74decb8187611e3706c6fe3621684", algorithm=MD5
Content-Type: application/media_control+xml
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
[2016-04-12 16:19:46] VERBOSE[1898] chan_sip.c: --- (13 headers 1 lines) ---
[2016-04-12 16:19:46] VERBOSE[1898][C-0000000a] chan_sip.c: Receiving INFO!
[2016-04-12 16:19:46] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK1961126868;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 14 INFO
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:48] VERBOSE[16442][C-0000000a] chan_sip.c: Reliably Transmitting (NAT) to 10.1.10.33:5060:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK1fddf27b;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 114 INFO
User-Agent: FPBX-13.0.101(13.7.1)
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---
[2016-04-12 16:19:48] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK1fddf27b;rport=5061
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Call-ID: [email protected]
CSeq: 114 INFO
Contact: <sip:[email protected]:5060>
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:48] VERBOSE[1898] chan_sip.c: --- (9 headers 0 lines) ---
[2016-04-12 16:19:48] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK3744415821
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 15 INFO
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="41d74decb8187611e3706c6fe3621684", algorithm=MD5
Content-Type: application/media_control+xml
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
[2016-04-12 16:19:48] VERBOSE[1898] chan_sip.c: --- (13 headers 1 lines) ---
[2016-04-12 16:19:48] VERBOSE[1898][C-0000000a] chan_sip.c: Receiving INFO!
[2016-04-12 16:19:48] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK3744415821;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 15 INFO
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:49] VERBOSE[16442][C-0000000a] chan_sip.c: Reliably Transmitting (NAT) to 10.1.10.33:5060:
INFO sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK15aa2c20;rport
Max-Forwards: 70
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 115 INFO
User-Agent: FPBX-13.0.101(13.7.1)
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---
[2016-04-12 16:19:49] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK15aa2c20;rport=5061
From: <sip:[email protected]:5061>;tag=as57e92102
To: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
Call-ID: [email protected]
CSeq: 115 INFO
Contact: <sip:[email protected]:5060>
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:49] VERBOSE[1898] chan_sip.c: --- (9 headers 0 lines) ---
[2016-04-12 16:19:49] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK2736133337
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 16 INFO
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="41d74decb8187611e3706c6fe3621684", algorithm=MD5
Content-Type: application/media_control+xml
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
[2016-04-12 16:19:49] VERBOSE[1898] chan_sip.c: --- (13 headers 1 lines) ---
[2016-04-12 16:19:49] VERBOSE[1898][C-0000000a] chan_sip.c: Receiving INFO!
[2016-04-12 16:19:49] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK2736133337;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 16 INFO
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:50] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
BYE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK3649504345
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 17 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="303", realm="asterisk", nonce="1e2215af", uri="sip:[email protected]:5061", response="4b017c21b95888379547d6a9e40876f5", algorithm=MD5
Max-Forwards: 70
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:50] VERBOSE[1898] chan_sip.c: --- (11 headers 0 lines) ---
[2016-04-12 16:19:50] VERBOSE[1898][C-0000000a] chan_sip.c: Sending to 10.1.10.33:5060 (NAT)
[2016-04-12 16:19:50] VERBOSE[1898][C-0000000a] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)
[2016-04-12 16:19:50] VERBOSE[1898][C-0000000a] chan_sip.c:
<--- Transmitting (NAT) to 10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.33:5060;branch=z9hG4bK3649504345;received=10.1.10.33;rport=5060
From: "Thor Jr" <sip:[email protected]:5061>;tag=509328488
To: <sip:[email protected]:5061>;tag=as57e92102
Call-ID: [email protected]
CSeq: 17 BYE
Server: FPBX-13.0.101(13.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] bridge_channel.c: Channel SIP/303-00000014 left 'simple_bridge' basic-bridge <239fbfec-72b6-495a-8847-a5094e842cc8>
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] app_macro.c: Spawn extension (macro-dial-one, s, 47) exited non-zero on 'SIP/303-00000014' in macro 'dial-one'
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] app_macro.c: Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/303-00000014' in macro 'exten-vm'
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Spawn extension (ext-local, 305, 2) exited non-zero on 'SIP/303-00000014'
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Executing [[email protected]:1] Macro("SIP/303-00000014", "hangupcall,") in new stack
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Executing [[email protected]:1] GotoIf("SIP/303-00000014", "1?theend") in new stack
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Goto (macro-hangupcall,s,3)
[2016-04-12 16:19:50] VERBOSE[16470][C-0000000a] bridge_channel.c: Channel SIP/305-00000015 left 'simple_bridge' basic-bridge <239fbfec-72b6-495a-8847-a5094e842cc8>
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Executing [[email protected]:3] ExecIf("SIP/303-00000014", "0?Set(CDR(recordingfile)=)") in new stack
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Executing [[email protected]:4] Hangup("SIP/303-00000014", "") in new stack
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/303-00000014' in macro 'hangupcall'
[2016-04-12 16:19:50] VERBOSE[16442][C-0000000a] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/303-00000014'
[2016-04-12 16:19:55] VERBOSE[1898] chan_sip.c: Reliably Transmitting (NAT) to 10.1.10.33:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK0a74e0d6;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]:5061>;tag=as4a37faf8
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.101(13.7.1)
Date: Tue, 12 Apr 2016 22:19:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[2016-04-12 16:19:55] VERBOSE[1898] chan_sip.c:
<--- SIP read from UDP:10.1.10.33:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.30:5061;branch=z9hG4bK0a74e0d6;rport=5061
From: "Unknown" <sip:[email protected]:5061>;tag=as4a37faf8
To: <sip:[email protected]:5060>;tag=1782631055
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP VP-T49G 51.80.0.80
Content-Length: 0

<------------->
[2016-04-12 16:19:55] VERBOSE[1898] chan_sip.c: --- (8 headers 0 lines) ---
[2016-04-12 16:19:55] VERBOSE[1898] chan_sip.c: Really destroying SIP dialog '[email protected]:5061' Method: OPTIONS
[2016-04-12 16:19:57] VERBOSE[1898] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: BYE

Something I have discovered using Grandstream GXV3140 and Linlhone is that if the phones have more than one video codec enabled regardless of the PBX settings the video will fail, sometimes only in one direction. If you try disabling all but one codec on the phones that might work.

1 Like

I disabled all except h264 but that doesn’t appear to be the problem. I think cynjut was onto something checking to make sure the phones were actually trying to use h264 instead of some other codec but I don’t know how to check what they are actually using. I’ll have to look into that more.

When placing a call I see this warning in the CLI

[2016-04-13 15:48:31] WARNING[1898][C-00000031]: chan_sip.c:11175 process_sdp_a_audio: Got Siren14 offer at 32000 bps, but only 48000 bps supported; ignoring.
[2016-04-13 15:48:31] WARNING[1898][C-00000031]: chan_sip.c:11175 process_sdp_a_audio: Got Siren14 offer at 24000 bps, but only 48000 bps supported; ignoring.
[2016-04-13 15:48:31] WARNING[1898][C-00000031]: chan_sip.c:11166 process_sdp_a_audio: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.

Figured out how to view the channels its using. Seems to be using h264 and g722 as configured.

localhost*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
10.1.10.33 303 [email protected] (h264|g722) No Rx: INFO 303
10.1.10.35 305 081e4bd6444878c (g722|h264) No Tx: INFO 305

Just want to give an update if anyone in the future has the same problem not getting their video to work. After having reinstalled and started from scratch multiple times on the latest FreePBX 10.13.66 and Asterisk 13 I decided to try the 6.12.65 version with Asterisk 11. Input the same exact settings and it worked! Seems to be something wonky with the latest FreePBX or Asterisk 13 versions when it comes to the video.

I have seen the exact same thing! I was setting up a new PBX using the latest distro from FreePBX and Asterisk 13. I could not for the life of me get the video to preform nicely. It would start with a picture, but it would be so lagged it was unusable. After a small period of time, the video would drop to a black screen. I knew my setup was fine as I had it working fine on our office PBX (running Asterisk 11). So I decided to change the Asterisk version on the new PBX from 13 to 11, and everything started working instantly!

I would have to agree that there is something funky with Asterisk 13 and video… I am just wondering how widely used video calls are, as I am sure we cannot be the only two people to have run into this problem.

On that note, how many calls should you be able to have in a video conference call? Video only works well for me when its one on one, but introduce a 3rd person and two of the endpoints video is very bad quality.

I second having trouble with the SIP-T49G’s and video calling. I was running the latest distro with Asterisk13/FreePBX13 and kept getting black screens during the call, this was using chan_sip, I couldn’t get the phones to ever register using pjsip. Everything is on the same network. I reinstalled with Asterisk 11/FreePBX 12 and everything is working as should.

I couldn’t get the phones to ever register using pjsip

pjsip usually runs on a different port to sip, you have to set this up on the sip settings screen. I also read somewhere that you have to explicitly allow the codec on the extension before video would work on pjsip… its been a while since I played with it now, but from memory I was having the same video issue with pjsip as I was with the original sip driver.

Its a shame because using pjsip on the particular project I was talking about would have solved many problems and made a few things much simpler!

I am surprised no-one else has come in and complained about this. I keep going back to it being a configuration issue, but without changing any config and rolling back to Asterisk11 it works fine.

On the newer distros PJSIP runs be default on 5060, and SIP on 5061.