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Asterisk Unable To Create/find Sip Channel For This Invite

For example SIP-to-SIP calls or DAHDI-to-DAHDI calls.3. The correct syntax is canreinvite=yes/no Connecting media paths direct to an endpoint behind NAT won't be pretty. Please see the documentation referenced in the WARNING that was generated. Of course Asterisk will initiate the direct media only if the media is not needed in Asterisk, e.g. have a peek here

The remote end responds with a 200 saying "OK, my audio stream is on IP y.y.y.y port y, and I choose codec B", which is sent back to the first phone. ADDITIONAL INFORMATION ****** i`m running asterisk on a dedicated linux server hosting, with CentOS 5.3. Why not get out of the loop and let themexchange these bits directly with each other? http://superuser.com/questions/810951/how-do-i-check-the-ulimit-for-another-user-and-change-open-files share|improve this answer edited Mar 18 '15 at 7:13 answered Mar 18 '15 at 7:01 arheops 9,0261917 add a comment| Your Answer draft saved draft discarded Sign up or http://forums.asterisk.org/viewtopic.php?p=198402

Closing issue. The two legs have different Call-Ids, and so are different SIP calls. Providers offering unlimited calling plans may have restrictions. Show David Woolley added a comment - 29/Oct/09 5:26 AM This appears to be the normal response to a retransmitted incoming INVITE.

If *one* of them have canreinvite=noor something else that stops a direct audio relationship from phone A to B,Asterisk stays in the middle of things, shipping bits between the phones(the audio When dtmfmode=rfc2833, asterisk will send the RTP stream through asterisk. This means it looks and smells almost like a SIP proxy.Note: Asterisk is still really not a SIP proxy in this case. So phone A sends an invite to Asterisk.Asterisk starts a *new* SIP dialog and sends an invite to Phone B.If Phone B accepts the call ("answers"), it sends an "200 OK"

The correct syntax is canreinvite=yes/no Connecting media paths direct to an endpoint behind NAT won't be pretty. Then you should see you are going to the from-trunk context. Please see the documentation referenced in the WARNING that was generated. Business PBX Solutions Provider Solution Details Become an ITSP Now!

Tell us about it. VOIP Event Calendar PBX Internet Speed Test About Voip-info.org Business VOIP Business Voip Providers IP PBX Asterisk Based PBX Hosted PBX Virtual PBX VOIP Billing PBX Phone System SBCs / Softswitch Asterisk sends a "200 OK" SIP message to phone A. How to make a shell read the whole script before executing it?

canreinvite=yes used to disable re-invites if you had NAT=yes. click for more info Asterisk 1.8 added the media_address= configuration option which can be used to explicitly specify the IP address to use in the SDP for media (audio, video, and text) streams.Background infoIn general When I check my log , it was given following warning : [Mar 17 13:33:03] WARNING[657] acl.c: Cannot create socket [Mar 17 13:33:03] ERROR[657] rtp.c: Unable to allocate socket: Too many This comes in 2 flavors:3a) During call setup the media will be forwarded via Asterisk.

Do they mean the same across all major operating systems?-1Call cannot come into asterisk0Installation of asterisk server in ubuntu-1How to Make my asterisk server to make Outbound Calls and Recieve Inbound0Asterisk navigate here With dtmfmode=info canreinvite works properly. Well I'm not sure what the complete set is, but one of those conditions is that both SIP channels must be marked "canreinvite=yes". Hide Permalink Tarek Khoury added a comment - 29/Oct/09 5:57 AM how can i solve this problem?

From messages in the archives and the Asterisk handbook one finds out that the Cisco ATA-186 does not handle the (re)INVITE well. Hello Nobody Logout Sign In or Sign Up (Why?) HomeRefine Search    Messages per Month     Sort by Relevance Date, Forward Date, Backward Start a set with this searchInclude this search in one of system (system) 2014-06-04 19:30:39 UTC #9 Home Categories FAQ/Guidelines Terms of Service Privacy Policy Powered by Discourse, best viewed with JavaScript enabled   Search for:Search Want your own MarkMail? Check This Out That is, the audio path is direct, whereas the SIP messages went via intervening proxies.[This is horrendously over-simplified, but it's enough to make the point]Now, the second thing to understand is

Asterisk uses itself as the end-points of media streams when setting up the call. On the minus side, this means more workload on the Asterisk server. If they support the same codec, if theycan talk to each other.

The way this works is: Phone sends INVITE saying "I am on IP x.x.x.x:x, I can use codecs A,B,C" Asterisk decides where the next leg is.

If either has "canreinvite=no", then Asterisk falls back to the default behaviour of setting up two separate legs.In your case, you need the default behaviour when calling the provider, because the It *only* controls Asterisk generating reINVITEs for the specific purpose of setting up a direct media path. If the clients use different codecs, Asterisk will not issue a re-invite. You're not closing something you need to close. –EJP Mar 18 '15 at 8:45 add a comment| 2 Answers 2 active oldest votes up vote 1 down vote accepted Just try

native-bridging (see also below): if both bridged channels use the same technology then (WRONG! However I recommend you avoid this. A published paper stole my unpublished results from a science fair What is the difference between perspective distortion and barrel or pincushion distortion? this contact form pedantic=no register => ::@sip.broadvoice.com [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=xxxx secret=xxxx username=xxxx insecure=very context=from-pstn authname=xxxx ;dtmfmode=inband ;dtmf=inband dtmfmode=rfc2833 dtm=rfc2833 canreinvite=no qualify=yes As you can see, I've tried both rfc2833 and inband

Carrier says they are delivering the call and it is failing at the PBX.