Author Topic: Call from '' to extension 'xxxxxx' rejected because extension not found  (Read 46827 times)

lehey

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Hello,
I have gone through this difficulty since the beginning of my usage of Askozia for my office. It is a great product. I use a WRAP board from PCEngines that I do not use anymore. Nice second life !

I am a Voxalot user and when connecting my ATA directly to Voxalot I had no problem inbout neither outbound.
Regularly, when a call came in I had this message: Call from '' to extension '160667' rejected because extension not found. I thought first that it was because of my config, and I tried every change I thought could improve things, including rebooting the machine, which solve the problem. After enabling the SIP debug mode, here is what made me understand the problem a little bit more:
This more explicit message "Found no matching peer or user for '64.34.163.35:5060'" made me realize that this IP address does not correspond to the host I am registering on. So because the INVITE does not come from the original registration, it is rejected, since match is IP based. I found no way to help matching the incoming call with the extension, so I ended up registering this IP address for inbound. I also had the same problem with VoipBuster.
Is there another way to do that than registering proxies, by discovering them through the log when they are unknown ?
Thanks

Michael

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I'm not sure I understand what's happening on their server side which is causing this. Why is that destination shifting?

A work around for now would be to set the incoming extension destination (phone, call group, etc...) to be "public." It would then be reachable regardless of source IP.

lehey

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Great ! Thanks for your help. Actually, I had already done that. But... I had put a different extension number in the uri, that f*ed up matching for an incoming call. Now I have put the extension that matches my Voxalot account in the uri for public direct dial and my last tests demonstrate that it works ok since after a while (re registration I believe), a different IP than the one corresponding to the registration server send the INVITE. I had seen in my logs before: "Looking for xxx in public-direct-dial", but it never did its way because of extension mismatch.
Since I am quite new to Asterisk, you were of great help for my understanding of the way Asterisk works and matches providers and extensions based on peers and patterns, and the way Askozia generates dial plans.
I have nothing against Astlinux, but I have to say that I choose Askozia because of its nice interface based on M0n0wall. I am now looking forward for the storage system implementation (soon I hope !), in order to enrich the functionality of my PBX
 Since the board I use does not have any way of attaching anything else than miniPCI cards, I believe I will have to wait for NFS client implementation. Keep up the good job and thank you again.
By the way, I am in Canada and speak french. If I can help for anything that involves french speaking skills. You can contact me.
Yann

Michael

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Glad it's working!

My French skills are nonexistent so you get the job.

pimboli

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Hi,

I have the same problem :(

what did you do to solve the problem?

litnimax

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What about adding insecure=invite in custom settings for the peer (provider) ?

xSmurf

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Salut ;)

I seem to be having the same problem. I have one SIP phone configured as the default for the one SIP provider entry I have. I tried using a group as someone elsewhere had pointed out a possible bug, but no go. I'm starting to think this might be related my NATing? I don't seem to have this other error of yours though.

xSmurf

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So I've followed the guide here http://forums.askozia.com/index.php/topic,329.0.html since I have the same provider. It almost works. My SIP client receives the call, answers, but the incoming call just keeps ringing and I get this in the logs

Code: [Select]
Jul 30 16:31:04 asterisk[1442]: VERBOSE[1479]: == Using SIP RTP CoS mark 5
Jul 30 16:31:04 asterisk[1442]: VERBOSE[1479]: == Using SIP VRTP CoS mark 6
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Executing [17775555555@public-direct-dial:1] NoOp("SIP/5060-00000002", "public calling callgroup: 17775555555 - <17775555555>") in new stack
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Executing [17775555555@public-direct-dial:2] Set("SIP/5060-00000002", "HASVOICEMAIL="yes"") in new stack
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Executing [17775555555@public-direct-dial:3] Set("SIP/5060-00000002", "SENDNOTIFICATIONS="user@domain.tld"") in new stack
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Executing [17775555555@public-direct-dial:4] Set("SIP/5060-00000002", "NOVOICEMAILWHENBUSY="yes"") in new stack
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Executing [17775555555@public-direct-dial:5] Gosub("SIP/5060-00000002", "macro-main,s,1(SIP/101,101,101,to,35)") in new stack
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Executing [s@macro-main:1] Dial("SIP/5060-00000002", "SIP/101,35,to") in new stack
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: == Using SIP RTP CoS mark 5
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: == Using SIP VRTP CoS mark 6
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- Called 101
Jul 30 16:31:04 asterisk[1442]: VERBOSE[16593]: -- SIP/101-00000003 is ringing
Jul 30 16:31:08 asterisk[1442]: VERBOSE[16593]: -- SIP/101-00000003 answered SIP/5060-00000002
Jul 30 16:31:28 asterisk[1442]: WARNING[1479]: chan_sip.c:3398 in retrans_pkt: Maximum retries exceeded on transmission hash@domain.tld for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt.
Jul 30 16:31:28 asterisk[1442]: WARNING[1479]: chan_sip.c:3425 in retrans_pkt: Hanging up call hash@domain.tld- no reply to our critical packet (see doc/sip-retransmit.txt).
Jul 30 16:31:28 asterisk[1442]: VERBOSE[16593]: -- Executing [h@macro-main:1] GotoIf("SIP/5060-00000002", "1?vm-h,1") in new stack
Jul 30 16:31:28 asterisk[1442]: VERBOSE[16593]: -- Goto (macro-main,vm-h,1)
Jul 30 16:31:28 asterisk[1442]: VERBOSE[16593]: -- Executing [vm-h@macro-main:1] ExecIf("SIP/5060-00000002", "1?Hangup()") in new stack
Jul 30 16:31:28 asterisk[1442]: VERBOSE[16593]: == Spawn extension (macro-main, s, 1) exited non-zero on 'SIP/5060-00000002'

sanitized


Also as noted here and in the other post, if I don't set the DID as the public alias, I get the some "rejected because extension not found" as above error message.

scdzaak

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Re: Call from '' to extension 'xxxxxx' rejected because extension not found
« Reply #8 on: January 09, 2011, 02:18:05 PM »
Yes,

I've got the same problem as the users before.
The main problem is that the call will dropped in an extension action 'public-direct-call'.
When you look in the extension.conf you will see the rule .. but somewhere the route will be lost !

I can't find out why .. Normaly in the extension.conf you can make rules like:

[default]
include => internal
include => fromisdn
include => fromsipprovider
include => ...

[internal]
exten => ....

[fromsipprovider]
exten => ...

But in the latest version this connection is 'lost' how we have to find the correct route ?
See more technical info: http://blog.syscodata.nl
Binnenkort meer informatie ook te vinden op http://www.askozia.nl

amee

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Re: Call from '' to extension 'xxxxxx' rejected because extension not found
« Reply #9 on: December 02, 2011, 11:20:21 PM »
I have the same problem, when incoming calls rejected and log has row
chan_sip.c:20396 in handle_request_invite: Call from '' to extension '78XXXXXXX' rejected because extension not found in context 'public-direct-dial'.
I open new topic about this problem
http://forums.askozia.com/index.php/topic,1916.0.html

amee

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Re: Call from '' to extension 'xxxxxx' rejected because extension not found
« Reply #10 on: December 06, 2011, 09:04:21 AM »

A work around for now would be to set the incoming extension destination (phone, call group, etc...) to be "public." It would then be reachable regardless of source IP.

Hm.... but its working.
 ::)