Sorry for the second mail, about the infrastructure: phone - asterisk - HW firewall including NAT - Sipgate SIP Provider
About Software: Asterisk 1.8.13.0 running on Raspbian Debian Linux (http://www.raspbian.org/, Raspbian includes up to date Asterisk paackages while the normal Raspberry Pi Debian does not) running on a Raspberry Pi http://www.raspberrypi.org/:-) 2012/6/17 Stefan at WPF <stefan.at....@googlemail.com> > 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? 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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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