Hi,
I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows:
Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection.
Incomming calls work fine. No problems with this.
I have found numerous documents that desribe how to dial out to PSTN numbers via the VoIP provider, but none of them work for me, or better yet, I haven’t been able to get them up and running, most likely as I don’t really know where to start.
Currently, I continuosly get the error that it cannot create a ZAP channel (which is true as I do not use this and have not set this up; there is only one trunk defined, which is ZAP). I can see which macro gets fired upon dial out, but I do not see where I can set the second trunk. I tried to create the trunk through AMP, but that messed up my install, so I had to reinstall. So can someone please help me out and tell me (high level) what to do to get this working.
So any help is appreciated
Best regards
Guy
PS! Oh, I’m quite new at asterisk, but I guess that’s pretty clear.
Hi,
I’ve managed to make some progression, but now I’m at another dead end. I can receive call through my SIP provider, but I still can’t make calls to PSTN numbers through my VOIP provider.
In the log file, it saus the following:
May 8 10:47:11 VERBOSE[1563]: – Executing e[1;36;40mSetCallerIDe[0;37;40m(“e[1;35;40mSIP/200-8f7fe[0;37;40m”, “e[1;35;40m31437110323e[0;37;40m”) in new stack
May 8 10:47:11 VERBOSE[1563]: – Executing e[1;36;40mSetCIDNamee[0;37;40m(“e[1;35;40mSIP/200-8f7fe[0;37;40m”, “e[1;35;40m31437110323e[0;37;40m”) in new stack
May 8 10:47:11 VERBOSE[1563]: – Executing e[1;36;40mSetCIDNume[0;37;40m(“e[1;35;40mSIP/200-8f7fe[0;37;40m”, “e[1;35;40m31437110323e[0;37;40m”) in new stack
May 8 10:47:11 VERBOSE[1563]: – Executing e[1;36;40mDiale[0;37;40m(“e[1;35;40mSIP/200-8f7fe[0;37;40m”, “e[1;35;40mSIP/XXXXXXXXXX@budgetphone.nl|30|re[0;37;40m”) in new stack
May 8 10:47:11 DEBUG[1563]: SIMPLE DIAL (NO URL)
May 8 10:47:11 DEBUG[1563]: Outgoing Call for XXXXXXXXXX
May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user
May 8 10:47:11 VERBOSE[1563]: – Called XXXXXXXXXX@budgetphone.nl
May 8 10:47:11 DEBUG[1563]: (Provisional) Stopping retransmission (but retaining packet) on ‘1f6df4381299d2161fc94ea9202acb42@192.168.123.151’ Request 102: Found
May 8 10:47:11 DEBUG[1563]: Acked pending invite 102
May 8 10:47:11 DEBUG[1563]: Stopping retransmission on ‘1f6df4381299d2161fc94ea9202acb42@192.168.123.151’ of Request 102: Found
May 8 10:47:11 WARNING[1563]: Forbidden - wrong password on authentication for INVITE to '“31437110323” ;tag=as01c07be8’
May 8 10:47:11 VERBOSE[1563]: – SIP/budgetphone.nl-25eb is circuit-busy
May 8 10:47:11 DEBUG[1563]: update_user_counter(XXXXXXXXXX) - decrement outUse counter
May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user
May 8 10:47:11 VERBOSE[1563]: == Everyone is busy/congested at this time
May 8 10:47:11 DEBUG[1563]: Exiting with DIALSTATUS=CONGESTION.
So I get a bad password, even though registering with my SIP provider using that password does not fail. Above all, I have found several articles on the internet stating this, but the all have more information after the INVITE.
Below you can find my call when dialing out:
exten => _XXXXXXXXXX,4,Dial(SIP/${EXTEN}@budgetphone.nl,30,r)
Is there anything missing?
Thanks for the help
Cheers
Guy