no audio on voicemail number

Does the 10000 test number work? Ivor's speaking clock on 020 7043 1320 is another Sipgate number you could try.

thanks for your response, yes both 10000 and 020 7043 1320 work fine, I can hear audio when I ring them.

anybody? please? this problem is driving me insane.

Bit of a tough one to narrow down. Seems a bit odd that 10000 works but

50000 doesn't. If the debug output doesn't show anything of any interest, then I would take a packet capture of each call and compare the differences.

Are you seeing anything from 217.10.79.35 being dropped, perchance? I compared calls to each number, RTP comes from 217.10.79.30 when using the test number and 217.10.79.35 when dialling voicemail.

ok thanks but which debug do you think I should run to catch the info you speak of?

What is the problem?

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ps: here is a small section of my config which shows the settings I have for voicemail with sipgate.co.uk. Does anything look wrong to you?

voice class codec 1 codec preference 1 g711ulaw

dial-peer voice 3 voip description **voicemail** destination-pattern 50000 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad

seems to me there must be a wrong setting in there somewhere....

in fact here's the whole router config is interested:

I meant check ACL logging in case RTP is being dropped unexpectedly, although you'd think having the SIP UA running on the edge would eliminate this kind of nonsense...

My only experience with Cisco voice kit is having a UC500 demo kit for a few weeks and using a 7941G with Asterisk. A quick google came up with this:

There are several interesting looking debug commands, but you'll have to get someone else to interpret the output :-) If the debug output is no use, there's always packet capture.

I did the packet trace thing and the data is streaming in 'as though' the phone is receiving audio. I compared the packet stream with a call containing audio to a call (to voicemail) containing silence and they were practically identical. Even though I couldn't hear anything on the voicemail number the data was pouring into the phone so it's a case of the phone (or the router) can't turn it back into audio. There must be something about the voicemail stream that is different to a normal PSTN call. This made me think of codec but I've tested every codec available in the dial peer and nothing will bring on the audio. It's so wierd.

anyway it looks as as though I might have to throw in the towel on this. As a final test I bought some credit with draytel.org and gave them a try. I used exactly the same settings on my equipment as I did with sipgate and whaddya know...their voicemail worked perfectly first time. The voicemail of both draytel and tescointernetphone work fine on my setup but sipgate doesn't. There has got to be something wrong or slightly different about sipgate the voicemail system but multiple emails to sipgate support have drawn a blank.

If you open up a packet capture in wireshark, you'll be able to play the captured audio stream back [if it's SIP/RTP, not sure about SCCP]. One thing I noticed with the voicemail call, the call setup packets coming from Sipgate didn't specify a packetisation time ['ptime']. May or may not be significant.

The only difference I've been able to discern is what peer the RTP goes to/from, and the lack of a ptime in the SDP part of the call.

This looks useful:

The debug output I got from Asterisk when calling Sipgate voicemail and other numbers indicated the accepted range of codecs were identical. Also the call setup would fail if the codecs don't overlap - an essential feature of SIP.

Sipgate support is pretty much non-existent, and such support that does exist will be unlikely to include Call Manager. I'm not knocking Sipgate however; the service is excellent for the price!