I have Grandstream GXP2160 Phones (I have tried firmware 1.05.16 - 1.0.5.33 as well as the latest 1.07.25 with the exact same symptoms.
Freepbx 13 with Asterisk 13 Servers (both primary and secondary servers are exactly the same)
After inputting my backup server ip address into the secondary sip server field my phone will not complete an assisted transfer successfully … I am using the GXP2160 transfer button (phone arrow phone icon). If I take the secondary server ip address out of the secondary server data field, I can successfully complete assisted transfers.
During my troubleshooting I have tried many different things however the bottom line is that I can enter any url or IP into the secondary sip server field and the attended transfers fail. I even entered www.google.com into the secondary sip server field and I am unable to complete an assisted transfers.
Going to the phones gui and looking at the Accounts / Account 1 / General Settings I have my primary servers ip address in the “Sip Server” field. I have my backup servers ip address in my “Secondary Sip Server” field and I have my primary servers ip address in the outbound proxy and I have my backup servers ip address in my backup outbound proxy.
One thing that I am noticing on my asterisk console of both servers is the errors below, these errors show randomly all the time, not just when I am attempting the attended transfers …
[2016-07-12 23:16:50] WARNING[3918]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1114 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [2016-07-12 23:16:50] WARNING[3918]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1093 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response [2016-07-12 23:16:50] WARNING[3918]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1093 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [2016-07-12 23:16:51] WARNING[3918]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1092 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransm
Any assistance that you can provide would be greatly appreciated.
This hints strongly toward a Grandstream firmware bug. You may get lucky and find someone here who has already dealt with this, but you should probably bring this up in the GS forum or their ticket system:
Hi, thanks for the reply Lorne. I don’t believe it has anything to do with the firmware upgrade. I am using the downgraded firmware as well as the downgraded Endpoint Manager that you had given me.
Has anyone gotten the secondary server to work properly in the GXP series phones???
Has anyone here ever gotten the secondary sip server to work with a GXP2160 or other GXP phone??? Am I wasting my time trying to get this to work?? It seems pretty odd that even when I put www.google.com in the secondary sip server field that it would mess up “attended” transfers.
Just another note, I had a Yealink T26P here that I just setup the exact same way as I was the Grandstream GXP2160. Primary Server, Secondary Server.
I logged into the console of each server. I dialed out of the T26P I could see the call being made out of the primary server. Dialed into a DID which also landed at the primary server. Shutdown asterisk on primary server.
Dialed out once again from the T26P and I could see the call being made out of the Secondary Server. Dialed that same DID once again which I saw land on the secondary server and hence the T26P rang.
No wierd errors at the console of either server. OK unless I can get the Grandstreams to do the same thing and as cleanly as the 5 minute setup on the Yealink, I’m sending the Grandstreams back!!!
you do know that with a decent session border controller you don’t need the second sip extension. the sbc can do the switch for you. this greatly simplifies the phone configs
We were able to have a 2160 working on two SIP servers (Both are External Servers) you have to lower the registration Timeout settings in order it should work.