Asterisk don't route calls to the destination

All installation and configuration problems and questions

Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N

Asterisk don't route calls to the destination

Postby AyukRolandAgbor » Mon Dec 09, 2019 7:04 am

Hello
I have installed Vicidial. I launched a call from (phone 101) to (phone 100) to test the regidered phones, but i get this error in the Asterisk logs (in red). Any help will be appreciated.

localhost*CLI>
== Using SIP RTP CoS mark 5
> 0x7fc800070c30 -- Strict RTP learning after remote address set to: 172.17.2.120:20000
-- Executing [100@default:1] Dial("SIP/101-00000004", "SIP/100|60|") in new stack
== Using SIP RTP CoS mark 5
[Dec 9 12:49:40] ERROR[13135][C-00000004]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("100|60|", "(null)", ...): Name or service not known
[Dec 9 12:49:40] WARNING[13135][C-00000004]: chan_sip.c:6316 create_addr: No such host: 100|60|
[Dec 9 12:49:40] WARNING[13135][C-00000004]: app_dial.c:2527 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [100@default:2] Goto("SIP/101-00000004", "default,85026666666666100,1") in new stack
-- Goto (default,85026666666666100,1)
-- Executing [85026666666666100@default:1] Wait("SIP/101-00000004", "1") in new stack
-- Executing [85026666666666100@default:2] VoiceMail("SIP/101-00000004", "100,u") in new stack
> 0x7fc800070c30 -- Strict RTP switching to RTP target address 172.17.2.120:20000 as source
-- <SIP/101-00000004> Playing 'vm-theperson.gsm' (language 'en')
== Spawn extension (default, 85026666666666100, 2) exited non-zero on 'SIP/101-00000004'
[Dec 9 12:49:43] WARNING[13135][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI("SIP/101-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------)") in new stack
-- <SIP/101-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
localhost*CLI>
AyukRolandAgbor
 
Posts: 16
Joined: Fri Dec 06, 2019 9:41 am

Re: Asterisk don't route calls to the destination

Postby ambiorixg12 » Mon Dec 09, 2019 9:14 am

replace the | by , . The | was used in the beginning of Asterisk as the separator, is not longer used and Asterisk interpret the whole whole 100|60| as host
ambiorixg12
 
Posts: 448
Joined: Tue Sep 17, 2013 10:35 pm

Re: Asterisk don't route calls to the destination

Postby AyukRolandAgbor » Tue Dec 10, 2019 2:55 am

That is right!!!

Thank you very much. It did solve the problem!!!
AyukRolandAgbor
 
Posts: 16
Joined: Fri Dec 06, 2019 9:41 am

Re: Asterisk don't route calls to the destination

Postby AyukRolandAgbor » Thu Dec 19, 2019 6:46 am

Hello!

Thank you to all who took out time to respond to this my question!!!

I also tried out this. And i think it is the best option because we won't need to be doing that (the editing) every time we reload the asterisk, or reboot the system. This problem in vicidial is as a result of the mismatch of the "asterisk version" in the system and on the web interface. If the two versions are the same probably we won't have that problem. I used to procedure below to resolve it on the web interface.

Changing the asterisk version in Admin>>server>>TESTast from 1.14.2.0 to 13.21.0-vici, worked for me just right.

Thank you
AyukRolandAgbor
 
Posts: 16
Joined: Fri Dec 06, 2019 9:41 am


Return to Support

Who is online

Users browsing this forum: njr and 100 guests