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703
Registration Date: 01.01.1970
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15.10.2009 16:30 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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RE: Xfer to PSTN extensions not working |
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Based on your description of this issue, it sounds like a vocoder related issue. The transfer is successful up to the point the RTP stream is being invoked, at which time the voice path fails.
Looking at the log of events above, I see the SIP UAs had settled on the G.711a vocoder for the inbound call, when it was answered by the IVR. This leads to a few questions:
• Do transfers to other extensions complete properly?
• What kind of SIP UAs are you using to initiate the transfers?
• Which kind of transfers (Blind or Attended) have you attempted so far?
• When the direct call to the PSTN number is successful, which vocoder is used for it?
• Do the trunks of the ITSPs you selected to handle the PSTN call, support the G.729 or another vocoder, except the G.711a?
Keep in mind, that merging call legs has it's own set of challenges, some of which you have undoubtedly experienced. Renegotiating vocoders in the middle of a call, sometimes leads to unexpected results.
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17.10.2009 13:56 |
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703
Registration Date: 01.01.1970
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06.11.2009 13:53 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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07.11.2009 00:03 |
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703
Registration Date: 01.01.1970
Posts:
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07.11.2009 15:35 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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RE: Xfer to PSTN extensions not working |
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Please don't feel sorry if you don't understand most of this stuff. No one was born with this knowledge. It's all acquired, and as I said earlier we are here to help.
• A SIP User Agent (UA) can be found in many forms, but it has a SIP stack and allows you to dial or receive a SIP based phone call. SIP is a protocol which allows SIP UAs and Proxies to communicate between them.
It can be a soft-phone, an ATA (Analog Telephone Adaptor), a desktop or cordless SIP phone, or even a mobile phone which contains a SIP client. So in your case, via which type of the above SIP UAs are you dialing or accepting calls? In essence, what type of SIP UA is registering on your PBXes extensions?
• The vocoder related questions can be determined easily, via the SIP UA used to make the call, or in a more complicated process, via the System Log.
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07.11.2009 20:43 |
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703
Registration Date: 01.01.1970
Posts:
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RE: Xfer to PSTN extensions not working |
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• A SIP User Agent (UA) can be found in many forms, but it has a SIP stack and allows you to dial or receive a SIP based phone call. SIP is a protocol which allows SIP UAs and Proxies to communicate between them.
I use both Grandstream and Aastra phones.
• The vocoder related questions can be determined easily, via the SIP UA used to make the call, or in a more complicated process, via the System Log.
Yes, both phones accept G729.
***Please be aware, I've had pbxes for about 2 years now with the same phones, and this problem has just started recently with the server crashes. Before that it always worked.
***Plus, it's not the phones, because the disconnects also occur when calls should be transferred from the digital receptionist directly (see first post).
That means, two separate (but related?) problems:
1) no transfers to PSTNs from digital receptionist
2) no blind OR attended transfers possible on INCOMING calls (Tfers possible on outbound calls)
PS: These problems exhibit themselves regardless of trunk combinations (i.e., various SIP providers) tried.
I really think it's a PBXes problem...IS ANYONE ELSE EXPERIENCING THE SAME THING?
Thanks for the feedback
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09.11.2009 04:37 |
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eco
Premium Account
Registration Date: 05.12.2008
Posts: 15
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01.12.2009 12:41 |
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