Grandstream GXP-1200 supports reinvite

[GaryHockin]

Hi Guys

Does anyone know if the Grandstream GXP-1200 series iphone supports reinvite with 3cx. We are discovering that our phones and routing calls through the server at all times (despite PBX delivers audio being turned off and supports reinvite being turned on). This appears to be causing a delay of the line and some crackling.

I had read somewhere that the grandstreams dont support reinvite with 3cx but do with other PBX's (like Asterisc).

Anyone any thoughts.

Gary Hockin
Launceston College

 

[LeonidasG_3CX]

Hi,

In Version 8 the GXP1200 Phone is automatically bound to media server no matter how you set it in the PBX, this is because Grandstream themselves had a small bug in their firmware that they just fixed recently in Version 9.

So back in V8, the grandstream could only use the "Bound to Media Server" option.


Grandstream did in the end fix the issue and you can now make us of the phone with re-invites.
To make it work with Re-Invites, go to your PBX's Updates > Other > Phone Autodetection and download the update.

Restart your PhoneSystem and try making a call and see what happens.

 

[GaryHockin]

Can I please confirm whether this is firmware version 9 or 3cx version 9 !

Thanks

Gary Hockin

 

[LeonidasG_3CX]

Can I please confirm whether this is firmware version 9 or 3cx version 9 !

This is for Version 8 AND Version 9.

 

[GaryHockin]

Ok

If I can just clarify

We have firmware 1.2.3.5 on our phones, all grandstream gxp1200

We have version 8.0.10824.716 of 3cx installed

We have ensured PBX delivers audio are unticked and Supports Reinvite and Support replaces header are ticked.


I have installed the phone autodetection update and have installed the new provisioning template.

I have got two phones plugged into a network hub (a hub not a switch so I can sniff packets correctly) and have a laptop going into the same hub running wireshark.

I can see RTP streams coming from
Phone A to Server
Server to Phone A
Phone B to Server
Server to Phone B
But NOT a direct RTP stream from Phone A to Phone B, is this to be expected? I assumed there would be a direct RTP stream to Phone A to Phone B.

Below is a copy of our logs from a call, it would appear the phone is not supporting reinvite

11:18:02.293 [CM102001]: Authentication failed for SipReq: REGISTER 172.16.0.17 tid=9a637b1adb981b981 cseq=REGISTER [email protected]:5064 / 19883 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
11:18:00.297 [CM102001]: Authentication failed for SipReq: REGISTER 172.16.0.17 tid=5c02cf78fb6f7ddf8 cseq=REGISTER [email protected]:5066 / 26859 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
11:17:59.127 [CM503008]: Call(1): Call is terminated
11:17:56.553 [CM503007]: Call(1): Device joined: sip:[email protected]:5060;transport=udp;user=phone
11:17:56.553 [CM503007]: Call(1): Device joined: sip:[email protected]:5060;transport=udp;user=phone
11:17:52.434 [CM505001]: Ext.405: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[no-reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [Grandstream GXP1200 1.2.3.5] PBX contact: [sip:[email protected]:5060]
11:17:52.434 [CM503002]: Call(1): Alerting sip:[email protected]:5060;transport=udp;user=phone
11:17:52.356 [CM503025]: Call(1): Calling Ext:Ext.405@[Dev:sip:[email protected]:5060;transport=udp;user=phone]
11:17:52.325 [CM503004]: Call(1): Route 1: Ext:Ext.405@[Dev:sip:[email protected]:5060;transport=udp;user=phone]
11:17:52.325 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=phone>
11:17:52.325 [CM505001]: Ext.403: Device info: Device Identified: [Man: Grandstream;Mod: GXP Series;Rev: General] Capabilities:[no-reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [Grandstream GXP1200 1.2.3.5] PBX contact: [sip:[email protected]:5060]
11:17:52.294 [CM503001]: Call(1): Incoming call from Ext.403 to <sip:[email protected]:5060;user=phone>

Thanks

Gary