I have had very little luck getting the two talking together.
For a very short time I did have calls originating on my FXO card routed to the phone working.
Phone1/2 on router ---> handytone works handytone ---> router phone1/2 works Phone1/2 on router ---> asterisk broken* asterisk ---> router phone1/2 broken**
* Can see asterisk receive the number and start following calling plan but no sound comes through, phone eventualy times out.
** Worked for a short time for no apparent reason, now sees phone as bussy, phone does not ring.
Router:
hostname: gateway-2.cybericom.co.uk
phone 1: p3000
phone 2: p3001
IP: 10.10.10.2Asterisk:
hostname: babybell.cybericom.co.uk
IP 10.10.10.3-If I use a phone connected to the router, I see asterisk receive the number and start following dial plan, but I dont hear anything and asterisk retries sending the following packet:
Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ors-20101 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=fSd-1369 To: <sip:[EMAIL PROTECTED]>;tag=as07550202 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232
v=0 o=root 16230 16230 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 13984 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
to 10.10.10.2:5060
-If I get a call via my fxo and it is supposed to be routed to a phone on the router I get a 404 not found but it seems asterisk is not asking for a specific phone by name, here is the first packet:
-- Executing Dial("Zap/1-1", "SIP/p3000|10|tr") in new stack
We're at 10.10.10.3 port 13934
Answering with preferred capability 2
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:gateway-2.cybericom.co.uk SIP/2.0
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK04efdbfe
From: "Cybericom" <sip:[EMAIL PROTECTED]>;tag=as179b70c5
To: <sip:gateway-2.cybericom.co.uk>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 10 Feb 2004 15:16:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 232v=0
o=root 16258 16258 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 13934 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 10.10.10.2:5060
-- Called p3000is there not supposed to be a p3000@ in the sip: line?
here is my sip.conf if something is wrong please let me know, thanks: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to context=in-sip ; Default for incoming calls callerid=Call canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=1800 defaultexpirey=600 tos=throughput
[p3000] type=friend host=dynamic user=p3000 ;secret= dtmfmode=rfc2833 mailbox=3000 callerid="Reception" <3000> qualify=yes context=wellingborough-road
[p3001] type=friend host=dynamic user=p3001 ;secret= dtmfmode=rfc2833 mailbox=3001 callerid="Reception" <3001> qualify=yes context=wellingborough-road
Router has box for registrar set to 10.10.10.3 place for naming phone set to p3000 and p3001 place for port set to 5060
Thank you for any help
Regards Chris Lee _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
