Thank you Warren, I will temporarily skip this step, as I don't have the problem anymore, though I don't know why (for that and learning purposes the logs maybe would be still useful). I found some different settings for Asterisk and Sipgate (actually I found the settings for private users on the Sipgate website, before that I found the settings for business customers and assumed there wouldn't be a difference). When I had the problem, my sip.conf looked like this:
[general] > port=5060 > bindaddr=0.0.0.0 > context=other > language=de > > register => <SIPID>:<SIP_PASS>@sipgate.de/<SIPID> > > > [sipgate] > type=peer > context=from_external_voip_provider > username=<SIPID> > defaultuser=<SIPID> > fromuser=<SIPID> > secret=<SIP_PASS> > host=sipgate.de > fromdomain=sipgate.de > qualify=yes > insecure=invite > nat=yes > Now my sip.conf looks like this (source: http://www.sipgate.de/faq/index.php?do=displayArticle&article=540&id=257): (I have commented the additions / changes) > [general] > port=5060 > bindaddr=0.0.0.0 > context=other > language=de > > qualify=no ; added > disallow=all ; added > allow=alaw ; added > allow=ulaw ; added > allow=g729 ; added > allow=gsm ; added > allow=slinear ; added > srvlookup=yes ; added > > register => <SIPID>:<SIP_PASS>@sipgate.de/<SIPID> > > [sipgate] > type=friend ; changed from peer to friend > context = from_external_voip_provider > username=<SIPID> > ;defaultuser=<SIPID> ; removed > fromuser=<SIPID> > secret=<SIP_PASS> > host=sipgate.de > fromdomain=sipgate.de > qualify=yes > insecure=invite > nat=yes > canreinvite=no ;added > dtmfmode=rfc2833 ;added > The dialplan in both cases was this: > [from_external_voip_provider] > exten => <SIPID>,1,Answer(1000) > exten => <SIPID>,n,VoiceMail(<some_number>,u) > exten => <SIPID>,n,Hangup() > (I left out the Dial command for testing purposes after I found the voicemail prompt problems) If anyone has an idea why it now works without problems, please let me know for learning purposes. I still have to read up on the options. When I have more time I will probably also set the old settings again to learn how I could have identified the problem. 2012/6/17 Warren Selby <wcse...@selbytech.com> > Please excuse the top post, I'm on my phone. > > Before we have a better idea of what's going on, please provide the > dialplan snippet that the call is using as well as the cli logs of the > calls where you hear the whole prompt and where you only hear part of the > prompt. > > Also, if you can clarify the infrastructure setup as well, that would be > helpful. > > Thanks, > --Warren Selby, dCAP > > On Jun 17, 2012, at 11:25 AM, Stefan at WPF <stefan.at....@googlemail.com> > wrote: > > Hmm, I tried calling myself (the asterisk voicemail) from another SIP > provider, same problem. What always works reliable is using and calling the > voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably > hear the complete prompt. Doesn't this contradict the assumption that the > problem is on the mobile phone side? > > 2012/6/17 Doug Lytle <supp...@drdos.info> > >> Stefan at WPF wrote: >> >>> Which end do you mean with "channel not answered"? The asterisk >>> >> >> The Asterisk side. If the answer didn't fix the issue, then my guess is >> that it's on the cellular provider's side (Which isn't unheard of). >> >> >> Doug >> >> >> -- >> Ben Franklin quote: >> >> "Those who would give up Essential Liberty to purchase a little Temporary >> Safety, deserve neither Liberty nor Safety." >> >> >> -- >> ______________________________**______________________________**_________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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