[Asterisk-Users] Iconnecthere
Armand A. Verstappen
armand at nl.envida.net
Mon Aug 11 04:26:50 MST 2003
Hi,
> I seem to have my configuration working except for outgoing and incoming
> calls for the rest of the world. For now I am concerned more about
> outgoing calls than anything else. Whenever I try to make an outgoing
> call I get these messages from the sip debug in the console
>
> s=session
> c=IN IP4 64.36.104.203
> t=0 0
> m=audio 6620 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> (no NAT) to 4.42.235.170:5060
> -- Called 13057400221 at packet8.net
> Sip read:
> SIP/2.0 404 Not Found 4
> Via: SIP/2.0/UDP 64.36.104.203:5060;branch=z9hG4bK37d8c90a
> From: "asterisk" <sip:asterisk at 64.36.104.203>;tag=as220b2c68
> To: <sip:13057400221 at 4.42.235.170>;tag=1m6lkhivci11cjdooja30ex45
> Call-ID: 72a3aaff62d64ab064938da36f1dcdb9 at 64.36.104.203
> CSeq: 102 INVITE
> Content-Length: 0
>
>
> Notice in particular the From line. Now notice a working session from
> eStara softphone:
>
>
> v=0
> o=eStara 22079953 22079953 IN IP4 64.36.104.202
> s=eStara
> c=IN IP4 64.36.104.202
> t=0 0
> m=audio 8014 RTP/AVP 0 4 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 64.36.104.202:5060
> From: Anonymous <sip:17862324057 at packet8.net>;tag=1d436a9
> To: <sip:13057400221 at packet8.net>
> Call-ID: ed3d1f22-3805-4117-a5ed-8df16f7cd5f9 at 64.36.104.202
> CSeq: 22079953 INVITE
> Content-Length: 0
>
>
> Notice how the from is different, my SIP service will not accept calls
> unless the proper from name is configured, how can I configure this?
> Here are the relevant sections from my sip.conf file
>
> [general]
> port = 5060 ; Port to bind to
> context = from-sip ; Default for incoming calls
> maxexpirey=13600 ; Max length of incoming registration we
> allow
> defaultexpirey=3600 ; Default length of incoming/outoing
> registration
> register => 17862324057:xxxxxxxxxxx at packet8.net/5500
>
> [packet8.net]
> type=friend
> username=17862324057
> secret=xxxxxxxxxxx
> host=packet8.net
> context=demo
Well, it's dialing out, that's one thing. Now to set the outgoing
caller-id and name, you'll need to do something like:
exten => s,1,SetCallerID(17862324057) ; your own callerid here
exten => s,2,SetCIDName(Anonymous) ; your proper from name
exten => s,3,Dial(SIP/${EXTEN}@packet8.net)
Note that the 's' extension should almost certainly be set to something
else for your configuration, I can't guess since you didn't volunteer
the relevant parts of your extensions.conf. But the SetCallerID() and
SetCIDName() functions will setup the from line accordingly (the
original link I sent you has this for Iconnecthere and fwd, btw)
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
armand at nl.envida.net 3531 AH Utrecht
tel: +31 (0)30 298 2255 Postbus 19127
fax: +31 (0)30 298 2111 3501 DC Utrecht
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030811/4f8d495f/attachment.pgp
More information about the asterisk-users
mailing list