Thank you Warren,<br><br>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).<br>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).<br>
When I had the problem, my sip.conf looked like this:<br><br><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">[general]<br>port=5060<br>bindaddr=0.0.0.0<br>
context=other<br>language=de <br><br>register => <SIPID>:<SIP_PASS>@<a href="http://sipgate.de/">sipgate.de/</a><SIPID><br><br><br>[sipgate]<br>type=peer<br>context=from_external_voip_provider<br>username=<SIPID><br>
defaultuser=<SIPID><br>fromuser=<SIPID><br>secret=<SIP_PASS><br>host=<a href="http://sipgate.de">sipgate.de</a><br>fromdomain=<a href="http://sipgate.de">sipgate.de</a><br>qualify=yes<br>insecure=invite<br>
nat=yes<br></blockquote><div><br>Now my sip.conf looks like this (source: <a href="http://www.sipgate.de/faq/index.php?do=displayArticle&article=540&id=257">http://www.sipgate.de/faq/index.php?do=displayArticle&article=540&id=257</a>):<br>
(I have commented the additions / changes)<br><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">[general]<br>port=5060<br>bindaddr=0.0.0.0<br>context=other<br>
language=de <br><br>qualify=no ; added<br>disallow=all ; added <br>allow=alaw ; added<br>allow=ulaw ; added <br>allow=g729 ; added <br>allow=gsm ; added<br>allow=slinear ; added <br>srvlookup=yes ; added<br>
<br>register => <SIPID>:<SIP_PASS>@<a href="http://sipgate.de/">sipgate.de/</a><SIPID><br><br>[sipgate]<br>type=friend ; changed from peer to friend <br>context = from_external_voip_provider<br>
username=<SIPID><br>;defaultuser=<SIPID> ; removed<br>fromuser=<SIPID><br>secret=<SIP_PASS><br>host=<a href="http://sipgate.de">sipgate.de</a><br>fromdomain=<a href="http://sipgate.de">sipgate.de</a><br>
qualify=yes<br>insecure=invite<br>nat=yes<br>canreinvite=no ;added<br>dtmfmode=rfc2833 ;added<br></blockquote><div><br>The dialplan in both cases was this:<br><blockquote style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
[from_external_voip_provider]<br>exten => <SIPID>,1,Answer(1000)<br>exten => <SIPID>,n,VoiceMail(<some_number>,u)<br>exten => <SIPID>,n,Hangup()<br></blockquote><div>(I left out the Dial command for testing purposes after I found the voicemail prompt problems) <br>
</div><br><br> 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.<br>
</div><br> <br></div><br><br><div class="gmail_quote">2012/6/17 Warren Selby <span dir="ltr"><<a href="mailto:wcselby@selbytech.com" target="_blank">wcselby@selbytech.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF"><div>Please excuse the top post, I'm on my phone. </div><div><br></div><div>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. </div>
<div><br></div><div>Also, if you can clarify the infrastructure setup as well, that would be helpful. <br><br>Thanks,<div>--Warren Selby, dCAP</div></div><div><div class="h5"><div><br>On Jun 17, 2012, at 11:25 AM, Stefan at WPF <<a href="mailto:stefan.at.wpf@googlemail.com" target="_blank">stefan.at.wpf@googlemail.com</a>> wrote:<br>
<br></div><div><span></span></div><blockquote type="cite"><div>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?<br>
<br><div class="gmail_quote">2012/6/17 Doug Lytle <span dir="ltr"><<a href="mailto:support@drdos.info" target="_blank">support@drdos.info</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>Stefan at WPF wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Which end do you mean with "channel not answered"? The asterisk<br>
</blockquote>
<br></div>
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).<div><div><br>
<br>
Doug<br>
<br>
<br>
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