Asterisk PBX -- Re: Missing voicemail prompt beginning

2012/6/17 Stefan at WPF <stefan.at.wpf@xxxxxxxxxxxxxx>

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@xxxxxxxxxxxxx>

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@xxxxxxxxxxxxxx> 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@xxxxxxxxxx>

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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http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
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