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Got SIP response 503 "no route destination" from voip

PostPosted: Mon Oct 08, 2012 5:32 pm
by pagmasakpan
When the calls coming in then it suddenly hang up ..and this is an outbound setup

Here is the asterisk cli log:
Code: Select all
Executing [915473598353@default:1] [1;36;40mAGI[0;37;40m("[1;35;40mLocal/915473598353@default-a6f4,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack

[Kgo*CLI>
[0K[Oct  8 18:19:41]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

[Kgo*CLI>
[0K[Oct  8 18:19:41]     -- Executing [915473598353@default:2] [1;36;40mDial[0;37;40m("[1;35;40mLocal/915473598353@default-a6f4,2[0;37;40m", "[1;35;40mSIP/voipProvider/15473598353||tTo[0;37;40m") in new stack

[Kgo*CLI>
[0K[Oct  8 18:19:41]     -- Called voipProvider/15473598353

[Kgo*CLI>
[0K[Oct  8 18:19:41]     -- Got SIP response 503 "No Route" back from x.x.x.x <<----- ip address of our voipProvider

[0K[Oct  8 18:32:46] [1;33;40mNOTICE[0;37;40m[31114]: [1;37;40mchan_local.c[0;37;40m:[1;37;40m599[0;37;40m [1;37;40mlocal_call[0;37;40m: No such extension/context 91547359835@default while calling Local channel



under the extensions.conf

[globals]
SIPtrunk=SIP/voipProvider

dial plan entry:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(SIP/voipProvider/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,n,Hangup

Campaign Dial Plan prefix: 9


where do i miss?
pls help thanks!.

Re: Got SIP response 503 "no route destination" from voip

PostPosted: Mon Oct 08, 2012 11:45 pm
by striker
check with your voip provider.
either you are dialling wrong number or number that is not permitted by your provider.

Re: Got SIP response 503 "no route destination" from voip

PostPosted: Mon Oct 08, 2012 11:47 pm
by williamconley
1) Excellent job posting your specs, but please do try to include your GoAutoDial .iso version (2.0? 2.1?) as that can often help.

2) FYI (doesn't hurt anything, just something to notice): You are using SIP/voipProvider... in the "dial" command, but you have defined a Globals String ... that you are not using. If you used ${SIPtrunk} in the Dial String, that globals string would be getting used (and the result would be the same ...). Which means you could either skip defining the globals string or include it (or leave it as it is, your choice, just an observation). I generally recommend using the globals string in the dial command so it can be altered more easily, but that's just how I'm used to doing it.

3) I have issues believing your posted log was not edited. It appears that your system both dialed it and refused to dial it, and then your carrier added a failure as well. Ordinarily you do not get "dial" and "no such extension" in the same request, as the No Such Extension is an indicator that it could not find the dialplan ... and dial is IN the dialplan it could not find. Also, you would not ordinarily get "503 no route" if you first got "no such extension" ... and getting no such extension AFTER getting a response from teh carrier is just plain freaky (since no such extension means it can't dial the carrier ...).

4) At any rate: If you have No Such Extension, this means you do not have a dialplan (exten=> entry) that matches the number dialed, and if you get "no route" from a carrier, your carrier is refusing to complete your call. Those are the two errors you show, but how you managed to get them in that order is beyond me. :)

5) Happy hunting.

Re: Got SIP response 503 "no route destination" from voip

PostPosted: Tue Oct 09, 2012 8:21 am
by pagmasakpan
@William: Thanks, i figured it out now..that's why so long waiting time on agents..and regarding to the no such extensions only few of the numbers are having on that error..
maybe the other country that we are going to dial was not set properly in our dial plan..since we are dialing only u.s states..

@striker: yes i did..

anyway guyz i found a new warning message in the asterisk log..it says "RTCP read too short"
i need some info and clarification regarding this warning message..is this our side or to our voip provider?

FYI: our server is under NAT

until now i cant find any solution about that..

any idea? thanks ..

Re: Got SIP response 503 "no route destination" from voip

PostPosted: Tue Oct 09, 2012 8:37 pm
by williamconley
put your server on an external IP address

OR forward port 5060 UDP and port range 10000-25000 UDP to your vicidial server through your router. and turn off the firewall in your vicidial system (as a test, of course, put it back after the test).