https://issues.asterisk.org/jira/browse/ASTERISK-20633
Date
Votes
5 comments
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Oscar Colka Yeastar S100 is based on asterisk version 13, not sure if it have the same issue. I don't have a device can generate the SIP re-invite packets described in the ASTERISK-20633, so I couldn't test this in my office.
All my SIP devices send the re-invite with a=sendrecv to resume the on hold call.
Do you have any problem with these devices Broadsoft/Cisco Call Manager? If yes, could you send the SIP logs so we can report this to Yeastar team?
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Carlo van Dongen The problem is at one of my customers, when they call to another company and there is the call transfered, the audio is gone.
I spoke with the provider and they saw this happens.
I gonna try to make a pcap and will post it.
This happens on a S100 connected to Broadsoft.
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Oscar Colka OK, a pcap logs is welcome, so that we can check whether the issue is caused by the issue describe in the asterisk issue ASTERISK-20633
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Kane Boere Does anyone have found a solution for this issue?
I have the same problem here based in a Broadsoft platform
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Roel Koster Our customers have the same problem:
"The problem is when they call to another company and there is the call transfered, the audio is gone"
The customer has a Yeastar (asterisk) PBX, the another company uses Broadsoft, then this problem arises.
We solve this problem by connecting the trunk/phone number of the customer to another ITSP, the problem is gone.
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