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Hello , I’ll just started with asterisk and I would
liket to to dial between your two
phones with to cisco ATA 186 , but I have a problem
The two cisco ATA and the server in the same networks
and i have the ring in the phone but i’am not able to place a call Between the twe phone . In attachement the sip.conf and a log file Any suggestement . Regards RAbii ------------------------------sip.conf----------------------------------------------- [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all
addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't
know about here [2000] type=friend ; This device takes and makes
calls username=2000 ; Username on device secret=9overthruster7 ; Password for device host=dynamic ; This host is not on the same
IP addr every time context=from-sip ; Inbound calls from this host
go here mailbox=100 ; Activate the message waiting
light if this dtmfmode=rfc2833 ; voicemailbox
has messages in it [2001] ; Duplicate of 2000, except
with different auth data type=friend username=2001 secret=11bbanzai9 host=dynamic context=from-sip mailbox=101 dtmfmode=rfc2833 ------------------------------------------------------------------------------------------------- ~ -------------------------------Log
--------------------------------------------------- Asterisk*CLI> We're at 10.100.18.125 port 18294 Answering/Requesting with root capability 1 Answering with non-codec capability 0x1(G723) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp
SIP/2.0 Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad From: "NafthaChimie" <sip:[EMAIL PROTECTED]>;tag=as49257c3d To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 01 Dec 2004 14:35:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 11153 s=session c=IN IP4 10.100.18.125 t=0 0 m=audio 18294 RTP/AVP 4 101 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.100.18.124:5060 Asterisk*CLI> Sip read: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad From: "NafthaChimie" <sip:[EMAIL PROTECTED]>;tag=as49257c3d To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=556017164 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Cisco ATA 186 v3.1.0 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER,
REGISTER Content-Length: 0 Warning: Media type not available 10 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp
SIP/2.0 Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad From: "NafthaChimie" <sip:[EMAIL PROTECTED]>;tag=as49257c3d To: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;tag=556017164 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.100.18.124:5060 Destroying call '[EMAIL PROTECTED]' ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ |
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