After a month of battling with the infamous Cisco 7970 - 71 IP PHone SIP I have been unable to do a 3-way conference call.
I am using
Free Pbx 14.0.3.19
Asterisk 13.23.1
OS: Sangoma
I have gotten the phone to fully registered and I am able to make calls and use the other soft keys, however, I have been completely unsuccessful on of making a 3-way conference called with the Cisco phone.
I have attached a picture with the logs from “set sip debug on”
I am not sure but i do not think so, server-side conferencing is more like when you have more then 3 participants on a conference, (and then a number is created with desire password, etc) i know that in this case a conference room can be created from the server side (FREEPBX)
IT seems that this should have not be a problem due to the fact that every other softkey work, dnd, call forwarding, transfer , etc
However, when it comes to Conference I am unable to find any configuration that works or any related topic about it, as if it suppose to work as soon as the phone registered just like every other softkey
I do think that this Cisco IP Phones 3-way conference called take place locally, it does offer the option for conference as softkey and everything seem to work until the last step “To joing everybody together.” Nothing happens and the errors attach in the picture it what came up
I hope this answers your questions, if not please do elaborate a bit more so may understand profoundly.
My questions arose because you said you saw what you attached when you pressed the conference button. In the SDP, there’s something called “x-cisco-remotecc-request” which I’d hazard is the phone requesting server-side (remote) call conferencing. That’s not something that Asterisk knows how to handle; and, unless I’m mistaken, it’s not something that FreePBX knows how to handle either.
It’s 100% a Cisco thing especially with the 79XX series. These phones are meant for the Cisco UCM and expect that to be their PBX. Cisco did make these or the 8XXX or the 9XXX phones with the idea that people would use them with random PBX systems. They made them for their PBX system.
As far as I know things like 3-Way Calling are among the items that require the Cisco Patch for Asterisk. The downside of this patch, it’s third party and lacks full Asterisk support. They have never created a patch for Asterisk 14, 15 and doesn’t seem to have a patch for 16.
In fact the current patch is only for Asterisk 13.21.1 and the current release is 13.23.1 So unless you are running FreePBX with Asterisk 13.21.1 this patch isn’t going to help. If you are running a version less than 13.21.1 then you should look in their archives for a patch to match your current version. Because updating your Asterisk version (in the Distro) would take you to 13.23.1 and again make the patch pointless.
I had to downgrade asterisk to the version of the path, regardless, the issues still persisted.
Which is why I thought some configuration was missing, I was wondering if anybody has successfully been able to apply this patch and get the 7970 - 71 to work with 3-way conference, since must instructions are for 8XXX series.
it has been fixed, apparently even thought, the asterisk engine was patch, there was some configuration that should have been added automatically and for some reason it dint, so i had to added manually, after that, things seem to be working.
