[asterisk-users] Missing voicemail prompt beginning

[Stefan at WPF]

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.wpf 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 <wcselby at 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.wpf 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 <support at 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."
>>>
>>>
>>> --
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>>
>> --
>> _____________________________________________________________________
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>>
>> --
>> _____________________________________________________________________
>> -- 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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