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- trunk config in the sip.conf file
- [trunk-smsmedia-warid]
- type=friend
- host=10.1.16.72
- context=to-warid
- disallow=all
- allow=gsm ; GSM consumes far less bandwidth than ulaw
- allow=ulaw
- allow=alaw
- nat = no
- canreinvite=no
- port=5060
- outbound calls context in the extension.conf file
- [to-warid]
- exten => 0800707020,1,Dial(SIP/trunk-smsmedia-warid/${EXTEN}, 60)
- exten => 0800707020,n,Hangup()
- sip debug output
- baraza*CLI> console dial 0800707020@to-warid
- Audio is at 172.18.0.34 port 13374
- Adding codec 0x2 (gsm) to SDP
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 10.1.16.72:5060:
- INVITE sip:0800707020@10.1.16.72:5060 SIP/2.0
- Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@172.18.0.34>;tag=as31a21a46
- To: <sip:0800707020@10.1.16.72:5060>
- Contact: <sip:asterisk@172.18.0.34>
- Call-ID: 00cb370752ba9dad0505c4042e682889@172.18.0.34
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.6.2.18
- Date: Thu, 05 May 2011 16:08:14 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 280
- v=0
- o=root 471337243 471337243 IN IP4 172.18.0.34
- s=Asterisk PBX 1.6.2.18
- c=IN IP4 172.18.0.34
- t=0 0
- m=audio 13374 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
- a=ptime:20
- a=sendrecv
- ---
- [May 5 19:08:14] NOTICE[2986]: console_video.c:133 console_video_start: voice only, console video support not present
- <--- SIP read from UDP:10.1.16.72:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport=5060
- Call-ID: 00cb370752ba9dad0505c4042e682889@172.18.0.34
- From: "asterisk"<sip:asterisk@172.18.0.34>;tag=as31a21a46
- To: <sip:0800707020@10.1.16.72:5060>
- CSeq: 102 INVITE
- Content-Length: 0
- <------------->
- --- (7 headers 0 lines) ---
- <--- SIP read from UDP:10.1.16.72:5060 --->
- SIP/2.0 500 Server Internal Error
- Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport=5060
- Call-ID: 00cb370752ba9dad0505c4042e682889@172.18.0.34
- From: "asterisk"<sip:asterisk@172.18.0.34>;tag=as31a21a46
- To: <sip:0800707020@10.1.16.72:5060>;tag=6c0828e6
- CSeq: 102 INVITE
- Reason: Q.850;cause=0;text="unknown"
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- Transmitting (no NAT) to 10.1.16.72:5060:
- ACK sip:0800707020@10.1.16.72:5060 SIP/2.0
- Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK4e2a9fb2;rport
- Max-Forwards: 70
- From: "asterisk" <sip:asterisk@172.18.0.34>;tag=as31a21a46
- To: <sip:0800707020@10.1.16.72:5060>;tag=6c0828e6
- Contact: <sip:asterisk@172.18.0.34>
- Call-ID: 00cb370752ba9dad0505c4042e682889@172.18.0.34
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.6.2.18
- Content-Length: 0
- ---
- << Hangup on console >>
- Really destroying SIP dialog '00cb370752ba9dad0505c4042e682889@172.18.0.34' Method: INVITE
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