| |
| News & Reviews |
Welcome to the Voxilla VoIP Forum.
Voxilla has been a trusted source for accurate, up-to-date information on the IP Communications industry since 2002. A dedicated staff of reporters and engineers produce feature articles and product reviews to keep industry watchers abreast of the people, companies, and trends driving a fast moving market.
You are currently viewing our boards as a guest which gives you limited access to view most discussions and access our other features. By joining our free community you will have access to post topics, communicate privately with other members (PM), respond to polls, upload content and access many other special features. Registration is fast, simple and absolutely free so please, join our community today!
If you have any problems with the registration process or your account login, please contact contact us.
Voxilla VoIP Forum |
Asterisk doesn't close sip channelTechnical support, how-to guides, troubleshooting, and general assistance, from beginner to seasoned pro, this is where to discuss Asterisk, the most powerful open source PBX. |
| | LinkBack | Thread Tools | Rate Thread | Display Modes |
| |||
| I have a SPA3000 and Asterisk running quite nicely after about a week playing around. The only serious problem I'm having now is that SIP channels to dialpad do not close. I run Asterisk on a Fedora box which is also my firewall, so it has a public IP address. iptables is set to allow RTP and port 5060 thru on udp. The call itself is fine, as are calls thru the PSTN and sipgate. sip.conf reads: Code: [dialpad] type=peer auth=md5 md5secret=x username=x fromuser=x host=66.35.222.58 rtptimeout=60 Code: ; international dialpad
exten => _00X.,1,Dial(SIP/${EXTEN:2}@dialpad,60,)
exten => _00X.,2,Congestion Code: baggygreen*CLI> sip show channel 45 baggygreen*CLI> * SIP Call Direction: Outgoing Call-ID: 4582dcf031bb6ac27e05c23e27d08088@a.b.c.d Our Codec Capability: 985087 Non-Codec Capability: 0 Their Codec Capability: 1 Joint Codec Capability: 1 Format unknown Theoretical Address: 66.35.222.58:5060 Received Address: 66.35.222.58:5060 NAT Support: No Our Tag: 1594628730 Their Tag: 57f972d7 SIP User agent: Asterisk PBX Username: 61212345678 Peername: x Original uri: sip:61212345678@66.35.222.58 Need Destroy: 1 Last Message: Rx: BYE Promiscuous Redir: No Route: sip:61212345678@66.35.222.58:5060;maddr=66.35.222.58 DTMF Mode: rfc2833 Thanks, - Simon |
| Thread Tools | |
| Display Modes | Rate This Thread |
| |
| | ||||
| Thread | Thread Starter | Forum | Replies | Last Post |
| Getting AAH to ring differently from each channel | cliffsur | Asterisk Support Forum | 6 | November 16th, 2005 10:49 AM |
| How do i diable second channel in iax sofphone | degsod | Asterisk Support Forum | 0 | October 2nd, 2005 09:22 PM |
| help needed in bluetooth channel | jimson | Asterisk Support Forum | 0 | July 27th, 2005 07:59 AM |
| Bad Channel Question | stufried | Asterisk Support Forum | 0 | June 26th, 2005 10:56 PM |
| ZAP channel error | sandman | Asterisk Support Forum | 1 | February 20th, 2005 11:40 AM |