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--- RE: Xfer to PSTN extensions not working (http://www1.pbxes.com/forum/threadid.php?threadid=1255617052)

Xfer to PSTN extensions not working

T-fers to PSTN extensions disconnect call immediately. For example, via digital receptionist:

Oct 15 16:20:36 VERBOSE[5840] logger.c: We're at 188.40.65.148 port 39948
Oct 15 16:20:36 VERBOSE[5840] logger.c: Video is at 188.40.65.148 port 39336
Oct 15 16:20:36 VERBOSE[5840] logger.c: Adding codec 0x8 (alaw) to SDP
Oct 15 16:20:36 VERBOSE[5840] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Oct 15 16:20:37 VERBOSE[5840] logger.c: -- Playing 'custom/DK_general_phone_tree' (language 'en')
Oct 15 16:20:41 VERBOSE[5840] chan_sip.c: Hangup call SIP/UNO127536-3f48, SIP callid 726d6d9d210094ca47a664ab6df7c320@94.231.97.6
Via direct t-fer:

Oct 15 16:23:57 VERBOSE[21236] chan_sip.c: SIP call transfer received for call 6c808bc441c411a25969634218926f6d@188.40.65.148 (REFER)!
Oct 15 16:23:57 VERBOSE[21236] logger.c: Transfer to 900 in from-internal
Oct 15 16:23:57 VERBOSE[21236] logger.c: Transfer from 703220-801 in from-internal
Oct 15 16:23:57 VERBOSE[21236] logger.c: -- Stopped music on hold on SIP/UNO127536-4d41
Oct 15 16:23:57 VERBOSE[9168] chan_sip.c: Hangup call SIP/703220-801-edae, SIP callid 6c808bc441c411a25969634218926f6d@188.40.65.148
Oct 15 16:23:58 VERBOSE[9168] chan_sip.c: Hangup call SIP/UNO127536-4d41, SIP callid 4d7fc9060a03215303277187171ca366@94.231.97.6
When I dial PSTN extensions directly, it works fine. T-fers to ANY PSTN do not work, not just to 900...tried 908 as well (different country with different trunk!).

Please advise.

RE: Xfer to PSTN extensions not working

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.

RE: Xfer to PSTN extensions not working

Hi,

A new discovery which might explain a bit. When I RECEIVE phone calls, I can make neither attended nor blind transfers. Pressing *2 or ## just sends the digits as tones.

When I make a phone call, though, I get the voice "transfer".

This seems like a programming problem. Please advise.

Thank you

RE: Xfer to PSTN extensions not working

I would like to kindly ask you, when asking for help on an issue, and someone takes the time to reply to your inquiry, providing some insight while asking some questions to pinpoint the issue you are experiencing, to take the time and reply to their questions.

If the questions seem difficult or might be hard to understand, we are here to explain them and help you along. So please take the time and answer my questions to help us troubleshoot your issue with transfers.

• Do transfers to other extensions complete properly?
• What kind of SIP UAs are you using to initiate the transfers?
• 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?

Thanks for listening and have a good day.

RE: Xfer to PSTN extensions not working

I'm sorry for the misunderstanding:

• Do transfers to other extensions complete properly?

NO, neither blind transfer NOR attended transfer works to other extensions.

• What kind of SIP UAs are you using to initiate the transfers?

I don't understand what you mean with SIP UA?

• When the direct call to the PSTN number is successful, which vocoder is used for it?

How do I determine this?

• Do the trunks of the ITSPs you selected to handle the PSTN call, support the G.729 or another vocoder, except the G.711a?

I use sipgate.de. I don't know what vocoders they use.


Please understand, I'm just a layman and not an expert. I don't understand all this stuff...Sorry!

Danke trotzdem fuer Ihre Hilfe!

RE: Xfer to PSTN extensions not working

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.

RE: Xfer to PSTN extensions not working

• 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

RE: Xfer to PSTN extensions not working

Yes we have experienced the same problem: no transfer on incoming calls: when I dial *2 it send tones and hang up the other party, but no transfer.

We are using G.711a on all trunks with Snom M3 handsets and transfers have been working well in the past.

I think it is a PBXes issue, because other clients using the service are reporting the same problem.