Hello,
I am using the voicemail module of asterisk. When I did some test calls
from my mobile phone, sometimes the beginning of the prompt was missing,
e.g. instead of something like number 12345 not available I was only
hearing 345 not available. Verbose level 5 on the asterisk console didn't
Stefan at WPF wrote:
beginning of the prompt was missing
User answer(500) or wait(1) before the audio prompts.
Example:
exten = s,1,Answer(500)
exten = s,n,Voicemail({$ARG1}@sip,u)
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
First of all, thank you for your reply, however I see two problems with
this solution:
1) I think sometimes even more than a second from the beginning of the
prompt is missing, so I have to set a larger value, meaning in cases where
nothing of the prompt was missing, the calling person listens to
Stefan at WPF wrote:
2) Your solutions handles the symptoms of the problem, I'd like to fix
the root cause of this problem.
The root cause of the problem (Most likely) is that the channel hadn't
be answered. A wait, allows the channel to be established and audio to
pass.
Doug
--
Ben
2012/6/17 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
2) Your solutions handles the symptoms of the problem, I'd like to fix
the root cause of this problem.
The root cause of the problem (Most likely) is that the channel hadn't be
answered. A wait, allows the channel to be
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
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
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
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
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
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