Hi
I try to enable the paging funktion on my yeatsr S50 PBX System. Finally it works but when I press the * key to connect to the caller privatly, the audio from the callee to the caller ist mute or not availible.
Alle phones are yearlink phones and the firmware of the Yeastar PBX ist the newest one.
It seems everything configured well and I dont find an error for this issue.
Greetings
WALTER
Hi Do you use 2 way paging? I just tested it works with my S100.
Hi
no I want to use 1 way Paging. But when I push the * Key it should be able to speak with each other.
When I configure the Paging Group with 2 way Intercom Audio will also not been tranferred from the callee to the caller. (But to all other Phones)
What make/model of phones? Where are they located in relation to the PBX- same LAN, different LAN, remote?
Hi
all phones are Yearlink phones. (T46G,T48G,T21G). All phones are in the same Network as the local PBX. No remote.
I believe it to be a bug.
I set up 1 way paging and selected a number of phones as members. When paging I see:
The above is the INVITE SDP of the page to the phone. It is telling the phone what IP and port to use for the return audio path and is using a time condition of no constraints and permanent. The sendrcv is telling the phone that the session is two-way. The phone subsequently responds with its 200OK/SDP.
However the page is indeed one-way as no audio is heard during the initial page and before attempting to use the "*". This implies that the audio stream is being stopped at the PBX as audio can be heard in the packet trace from both devices, but is simply not allowed to pass thru the PBX at this point.
I then subsequently press the * and this is clearly seen in the trace as well, but no audio passes back to the paging extension.
The above shows the call flow and I have a phone Linkus connected remotely. You can the RTP from the devices flowing, but you will not see the RTP stream between Linkus and the PBX as they use a different mechanism to transfer audio. Nevertheless, when doing a page, it works. When doing a standard Intercom (2-way) it works, when doing a page and the the * as seen in the above it does not work.
It does appear to be a bug in the PBX.
Thank you for your reply.
I hope that yeastar will fix this soon.
Hi There,
The problem is not SIP SDP. With feedback from the engineer, the RTP would still receive by PBX, but it would not send to the paging originate party when in simplex mode.
But the user experience is not good when the call is in private session when it is still in Simplex mode.
I would submit this problem to R&D to check how to fix it.
Thanks Pixy, but I never indicated it was a SIP SDP issue, but rather that the * dial code does not allow 2-way when answering a 1-way page. To me, it is a PBX issue as the SDP clearly represents 2-way enabled....but...there is no 2 way.
Hi @Pixy Tse, were you able to fix it?
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