Thanks all. It works fine now.
-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead.
-- Original message -- From: [EMAIL PROTECTED]
Thanks all. It
If you don't set the callerID in the channel, it will get passed on as-is.
Don't change it, and it will stay the same.
-TIm
On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
I have a follow up question. How do I pass on the caller ID of the call I'm
forwarding to the other party? I can
I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf
[4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed
whatever the did is needs to be put in the extensions.conf told to dial your cellphone.
Example:
exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED]
assuming that your using a SIP carrier, replace 1234567890 with your cellphone 1.2.3.4 with the carrier's IP or carriers context name in
It sounds like a good idea, I tried it and get this error
Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o
Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
In
Do you have gafachi-o in your sip.conf?
Since it's not a valid host name, you need to have an entry in sip.conf to
tell asterisk how to make a call to gafachi-o.
That's why it is telling you No such host.
On September 8, 2006 12:57, [EMAIL PROTECTED] wrote:
It sounds like a good idea, I
Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have anATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you
Check your Dial() string to make sure that you haven't mistyped and put
gafachi-o instead of gafachi-out. Specifiying the full host name will also
work.
As a hint, you can refresh these changes with out restarting your server (and
therefore without disrupting any calls in progress)
I looked through the forums but could not find exactly what I needed. I need help setting up call forwarding in sip.conf, where the call forwards to PSTN number without a sip phone but just the channels in sip.conf without any hardware or softphone. Any help will be greatly appreciated.
Call forwarding doesn't go in sip.conf, it has to go in the dialplan.
You set up your outbound provider in sip.conf
In your dialplan, you use the Dial application like this:
exten = _NXXNXX,1,Dial(SIP/outgoingprovider/${EXTEN})
This will dial out to a PSTN number based on the extension
I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.
In Sip.conf I have the channel
[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
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