I’m running the current asterisk 1.4.0 beta on my desktop system, which is actually a gentoo linux , kernel 2.6.18-gentoo-r6 on x86_64.
My idea is to create something like a very customizable voicemail recorder.
My asterisk is registered to my sip provider, I have managed to set up a software sip phone to connect to asterisk on localhost (port 5061!).
The setup was created by using the new gui. I created 1 user and 1 sip provider. This did’nt work for incoming or outgoing calls, so I added some lines to sip.conf and extensions.conf.
When I call my sip phone number at my provider, my softphone starts ringing. As configured, after 15 seconds, the call is sent to the voicemail. When it comes to recording, I get the following messages on the console:
As far as I am a real asterisk newbie, this does not make any sense to me.
Is this a known problem? Or any ideas how to fix this?
Maybe, I do not have enough sound channels?
Does asterisk use the sound card itself for playing and recording audio?
Do I need a second sound card to have my softphone running on the same machine?
On th other side, dialing out from the softphone to the sip provider does not work, this way I don’t even get a reaction on the console.
Maybe I didn’t find the Best-Practice-Tutorial on how to create cool voicemail systems with SIP and softphones?
Any hints and ideas are really appreciated. Thanks
i think the problem is the way you have it dial. Unless the SIP peer (provider) is named 49xxxxx etc then you are doing it wrong- you must dial in the form SIP/peer/extension, ie SIP/yourprovidername/numtodial.
good hint. Now I am able to make calls via my sip provider.
The setup created by the new asterisk gui seems to be meant in a different way than I thought. (Both my local account and the provider are created in the users.conf file). I added both in the sip.conf and made some changes in the extensions.conf.
Now asterisk dials out with
My problem not being able to record voicemails from my sip provider still persists.