Waiting for Ring

[Rev]

So I finally have everything setup and am testing a call list. Upon clicking on Dial next number all I get is waiting for ring and no numbers are being actually dialed I am not sure where the problem could be I've gone threw the manual several times to see if I missed something but it doesn't look like it.

[iteratel]
disallow=all
allow=ulaw
type=friend
username=***
secret=***
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,To)
exten => _91999NXXXXXX,3,Hangup

not sure if the issue is with the trunk or somewhere else

[bobchaos]

Having your install method and current version and build is usually a requirement for support on this forum (so you know), but your issue is probably right there in the code you posted:

exten => _91999NXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,To)

You'll want to make a broader dialing context, because that can only dial north american numbers that start with 999 :/ The default Vicidial extensions.conf sample file should have a context called [trunkoutbound] that should be functional as is for US/CAN. Just copy that into your carrier's dial plan, uncomment all the lines relevant to your situation and replace ${TESTSIPTRUNK} with the appropriate registration string for your carrier (Or change the value of TESTSIPTRUNK to match the registration string, if you haven't already)

[Rev]

Thank you. that solver one problem and presented another I now get and error "Cause: 20 - Subscriber absent." but this points to the provider so I am going to confirm with them to see if the sip/trunk was set-up correctly.

[Rev]

Al right I seem to be moving from one problem to another
I fix one thing and another problem pops up I get this when trying to make a call out

chan_sip.c:23838 handle_request_invite: Call from '4449' (10.0.0.95:5060) to extension '**********' rejected because extension not found in context 'default'.

[Acidshock]

Can you post your updated dial plan and the numbers you are dialing? Also what does your global string say under that carrier?

[Rev]

[iteratel]
disallow=all
allow=ulaw
type=friend
username=
secret=
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

TESTSIPTRUNK = SIP/testcarrier

exten => _91XXNXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXNXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91XXNXXXXXXX,3,Hangup

I'm just dialing my cellphone over and over again I've tried changing the dialing plan to 9XXXXXXXXXXX as well

[Rev]

[May 12 09:29:54] WARNING[13637]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:29:54] NOTICE[13637]: console_video.c:137 console_video_start: voice only, console video support not present
[May 12 09:29:54] -- Executing [4449@default:1] Dial("Console/dsp", "SIP/444 9,60,") in new stack
[May 12 09:29:54] == Using SIP RTP CoS mark 5
[May 12 09:29:54] -- Called SIP/4449
[May 12 09:29:54] WARNING[13771]: chan_oss.c:779 oss_indicate: Don't know how to display condition 22 on Console/dsp
[May 12 09:29:54] WARNING[13771]: chan_oss.c:779 oss_indicate: Don't know how to display condition 22 on Console/dsp
[May 12 09:29:54] -- SIP/4449-00000014 is ringing
[May 12 09:29:55] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:29:56] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:29:57] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:29:58] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:29:59] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:30:00] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:30:01] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:30:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 09:30:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 09:30:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 09:30:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 09:30:02] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:30:03] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:30:04] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
[May 12 09:30:05] WARNING[13771]: chan_oss.c:489 setformat: Unable to re-open DS P device /dev/dsp: No such file or directory
Vici1*CLI> console hangup
[May 12 09:30:06] == Spawn extension (default, 4449, 1) exited non-zero on 'Co nsole/dsp'
[May 12 09:30:06] -- Executing [h@default:1] AGI("Console/dsp", "agi://127.0 .0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CANCEL----------") in n ew stack
[May 12 09:30:06] -- <Console/dsp>AGI Script agi://127.0.0.1:4577/call_log-- HVcauses--PRI-----NODEBUG-----16-----CANCEL---------- completed, returning 0
[May 12 09:30:06] << Hangup on console >>
[May 12 09:30:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 09:30:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 09:30:25] > Refreshing DNS lookups.
[May 12 09:30:25] WARNING[2979]: netsock2.c:182 ast_sockaddr_split_hostport: Por t disallowed in 204.15.197.50:5060
[May 12 09:30:25] WARNING[2979]: acl.c:590 resolve_first: Unable to lookup '204.

I really do not understand this stuff

[bobchaos]

You still have the value of TESTSIPTRUNK = SIP/testcarrier where it should likely be TESTSIPTRUNK = SIP/iteratel.

Setting up a SIP trunk is very well documented on the internet as it is a common asterisk process (well, Vicidial adds the nice GUI to give it structure I guess). You should be able to google all the info you need fairly easily. VoiPinfo.org is a good place to start. You should familiarize yourself with those resources as Vicidial doesn't require much Asterisk knowledge but it does come in handy.

[Rev]

Sorry about that, I've already corrected that in the carrier information problem I'm getting is 2 part right now and I really don't know what to look for in error handling it.

First I keep seeing these warnings pop up
[May 13 08:50:25] WARNING[2979]: netsock2.c:182 ast_sockaddr_split_hostport: Port disallowed in 204.xx.xxx.xx:5060
[May 13 08:50:25] WARNING[2979]: acl.c:590 resolve_first: Unable to lookup '204.xx.xxx.xx:5060'

This is the ip address of the carrier that was provided from my carrier.

Second one I see is
NOTICE[3041]: chan_sip.c:23838 handle_request_invite: Call from '4449' (10.0.0.95:5060) to extension '416xxxxxxx' rejected because extension not found in context 'default'.

Whenever I try to make a call from a phone associated with the dialer.

[iteratel]
disallow=all
allow=ulaw
type=friend
username=0107*9000
secret=ZrET27qE
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

TESTSIPTRUNK = SIP/ITERATEL

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91XXXXXXXXXX,3,Hangup

[Rev]

Okay so I just made dramatic changes and now get the following

[ITERATEL-OUT]
disallow=all
allow=ulaw
type=friend
username=0107*9000
secret=ZrET27qE
host=xxx.xx.xxx.xx
dtmfmode=rfc2833
qualify=1000
context=trunkoutbound

[ITERATEL-IN]
disallow=all
allow=ulaw
type=friend
username=0107*9000
secret=ZrET27qE
host=xxx.xx.xxx.xx
dtmfmode=rfc2833
qualify=1000
context=trunkInbound

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITERATEL-OUT,,tTor)
exten => _1NXXNXXXXXX,3,Hangup

exten => _NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITERATEL-OUT,,tTor)
exten => _NXXNXXXXXX,3,Hangup

[May 13 09:39:38] -- Executing [xxxxxxxxxx@default:1] AGI("SIP/4449-00000018", "agi://127.0.0.1:4577/call_log") in new stack
[May 13 09:39:38] -- <SIP/4449-00000018>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 13 09:39:38] -- Executing [xxxxxxxxxx@default:2] Dial("SIP/4449-00000018", "SIP/xxxxxxxxxx@ITERATEL-OUT,,tTor") in new stack
[May 13 09:39:38] == Using SIP RTP CoS mark 5
[May 13 09:39:38] -- Called SIP/xxxxxxxxxx@ITERATEL-OUT
[May 13 09:39:38] -- Got SIP response 603 "Declined" back from
[May 13 09:39:38] -- SIP/ITERATEL-OUT-00000019 is busy
[May 13 09:39:38] == Everyone is busy/congested at this time (1:1/0/0)
[May 13 09:39:38] -- Executing [xxxxxxxxxx@default:3] Hangup("SIP/4449-00000018", "") in new stack
[May 13 09:39:38] == Spawn extension (default, xxxxxxxxxx, 3) exited non-zero on 'SIP/4449-00000018'
[May 13 09:39:38] -- Executing [h@default:1] AGI("SIP/4449-00000018", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----BUSY----------") in new stack
[May 13 09:39:38] -- <SIP/4449-00000018>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

I'm assuming this is now the problem [May 13 09:39:38] -- Got SIP response 603 "Declined" back from

[bobchaos]

Declined means your carrier is refusing to transmit the call, usually because you're using a configuration he doesn't like. If I was to guess, I'd say you probably are transmitting using a codec they don't support, but it could be many other things. Best thing to do is to enable maximum verbosity on your console, enable SIP debug mode and compare the INVITEs with corresponding replies. Feel free to post your results here if you're having a hard time interpreting them, it can look a bit confusing to untrained eyes, but it's fairly straightforward.

Also, when you run "sip show peers" and "sip show users", do you see both of your carrier entries in there?

[Rev]

Hello Bobchaos how would I compare invites?

I did the following
Core set Verbose 21

asterisk -vvvvvr to enter the command line

tried calling my cell and only got the same output as my previous post so don't think I did that right

results for show peers/users
Name/username Host Dyn Forcerport ACL Port Status
4449/4449 10.0.0.95 D N 5060 OK (8 ms)
ITERATEL-IN N 5060 OK (14 ms)
ITERATEL-OUT N 5060 OK (16 ms)


Username Secret Accountcode Def.Context ACL Forcerport
4449 test 4449 default No Yes
ITERATEL-IN ZrET27qE trunkInbound No Yes
ITERATEL-OUT ZrET27qE trunkoutbound No Yes


Also my provider has informed me that the codecs are g729 or g711 and I believe I have these 2 codecs already enabled

[bobchaos]

First step, in vicidial, disable all other codecs but those 2. It's not supposed to be required (the servers should negotiate for something they both have enabled) but I've seen weirder things. You seem to be registering fine, so base parameters for the iteratel accounts are probably OK.

To compare SIP invites, you need to enable "sip set debug on" and that should output the entire conversation between your SIP server and it's various endpoints (including the provider). Just make a call with debug enabled and you should be able to see the problem in there.

[Rev]

Hello Bob,

Sorry for the late response went away for a bit, as for the server not dialling out, the provider corrected the issue on there end and I can call out now on the phone that was setup.

Thanks for all your help.