Call Forwarding and Failed Transfers on 18.12

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Tommy Dome

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Mar 24, 2022, 12:05:24 AMMar 24
to sipxcom-users
I am having an issue with call forwarding on an 18.12 system located in an AWS EC2 instance with a Calltower SIP Trunk. Call forwarding works on some extensions but not others. I have done a stare and compare on a few pcaps and it looks like the ones that fail the call forward are trying to respond to the AWS subnet local IP address instead of the elastic IP. I am thinking its a codec issue because the ones that go through are using G722 and the ones that fail are using G711. I've tried to remove the G711 codecs from the phones assigned to the user but that doesn't do anything. I believe this is the same issue causing external call transfers from extension to extension to drop. 

Any ideas on how I can fix this?

Michael Picher

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Mar 24, 2022, 10:10:44 AMMar 24
to Tommy Dome, sipxcom-users
It would be pretty rare for a SIP trunk to support g.722? Usually they're g.711u or a only.

On Thu, Mar 24, 2022 at 12:05 AM Tommy Dome <dometech...@gmail.com> wrote:
I am having an issue with call forwarding on an 18.12 system located in an AWS EC2 instance with a Calltower SIP Trunk. Call forwarding works on some extensions but not others. I have done a stare and compare on a few pcaps and it looks like the ones that fail the call forward are trying to respond to the AWS subnet local IP address instead of the elastic IP. I am thinking its a codec issue because the ones that go through are using G722 and the ones that fail are using G711. I've tried to remove the G711 codecs from the phones assigned to the user but that doesn't do anything. I believe this is the same issue causing external call transfers from extension to extension to drop. 

Any ideas on how I can fix this?

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Michael Picher
Product Manager, UCaaS
 
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Tommy Laino

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Mar 24, 2022, 10:36:31 AMMar 24
to Michael Picher, Tommy Dome, sipxcom-users
I agree. However, I've tried to remove the G.722 protocols as well and no dice.

Nathaniel Watkins

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Mar 24, 2022, 1:33:54 PMMar 24
to Tommy Laino, Michael Picher, Tommy Dome, sipxcom-users
Try this:
System->Services->Media Services
Select your server

Move PCMU to the top of the Selected Codecs section.
image.png

Matt Keys

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Mar 26, 2022, 10:36:11 AMMar 26
to sipxcom-users
"Call forwarding works on some extensions but not others." -- This is true for consultative/attended transfer, or "at the same time" call forward. Example destinations this would fail is basically any freeswitch destination (voicemail, AA/IVR, conference) or to the PSTN. 

It is not true for blind transfer or "if no response" call forward. 

Matt Keys

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Mar 26, 2022, 11:00:03 AMMar 26
to sipxcom-users
I should add PCMU / G.711U is the standard on the PSTN in the US. PCMA / G.711A is the standard on the PSTN in other countries. It's not a good idea to remove PCMU or PCMA from the phone codecs. 

Other codecs (ie G729) are typically not supported unless your ITSP specifically states they support it. Even then typically it is likely to be transcoded to G711 somewhere along the way. 


Tommy Laino

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Mar 28, 2022, 1:09:48 PMMar 28
to Matt Keys, sipxcom-users
That is correct. Transfers from the AA are no issue. I would need to check closer to see if the call forwards failing are at the same or if No response.

I only removed codecs as a check because of what I was seeing in the capture. I put everything back in

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