vicidial.org

by **ruben23** » Fri Nov 04, 2011 11:29 pm

Asterisk 1.4.27.1
SVNtrunk agc
Ubuntu server 10.04 LTS
Scratch_install

Hi guys i have setup and Inbound number tested it with trunkinbound context and even tried creating the exact number extension, but the problme when i dialed the number it keeps getting this error, it not hitting the extensions, whihc seems impossible coz all are configure already based on the inboudn config.

this is the error code:

Code: Select all: [Nov 5 00:23:08] -- Added extension '348' priority 1 to vicidial-auto [Nov 5 00:23:08] == Parsing '/etc/asterisk/sip_notify.conf': [Nov 5 00:23:08] Found [Nov 5 00:23:43] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found. [Nov 5 00:23:46] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found. [Nov 5 00:23:49] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found. [Nov 5 00:23:53] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found. [Nov 5 00:23:56] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found. [Nov 5 00:24:00] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found. [Nov 5 00:24:01] == Parsing '/etc/asterisk/manager.conf': [Nov 5 00:24:01] Found [Nov 5 00:24:01] == Manager 'sendcron' logged on from 127.0.0.1 [Nov 5 00:24:01] == Parsing '/etc/asterisk/manager.conf': [Nov 5 00:24:01] Found [Nov 5 00:24:01] == Manager 'sendcron' logged on from 127.0.0.1 [Nov 5 00:24:01] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 5 00:24:01] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found.

Any idea guys..?

by **williamconley** » Sat Nov 05, 2011 9:12 am

use sip debug and verify that this call is indeed landing in trunkinbound. if it is, you can add extension "s" and point it to the agi script. then you may want to add a line to capture the real did and have that extension dialed instead of "s" so you can differentiate between DIDs.

by **boybawang** » Sat Nov 05, 2011 10:14 am

try hard coding it on extensions.conf to really test

by **randy_delgado_03** » Sat Nov 05, 2011 10:47 am

I agree with boy bawang, we're currently running blended campaigns and configured the inbound dialplan #include on extensions.conf in a different context.

by **ruben23** » Sat Nov 05, 2011 3:05 pm

@williamconley

i cant verified on the sip debug if its landing on the trunkinbound when i tried it but i tried this two:

Code: Select all: [trunkinbound] ; DID call routing process exten => _16543434456,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of receive$ ;exten => _X.,1,AGI(agi-DID_route.agi)

and

Code: Select all: ; inbound VICIDIAL call with CID delivery through T1 PRI ;exten => 16543434456,1,Answer ; Answer the line ;exten => 16543434456,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1) ;exten => 16543434456,3,Hangup

First one is on trunkinbound context and the second one is on default context.

by **williamconley** » Sat Nov 05, 2011 5:38 pm

perhaps you should post the sip debug. usually it states clearly which context it is searching for an exten before failing to "s" and trying again (and then states the context again for "s"). if it is not in default, then the inbound call is authenticating to the wrong account and not being sent to "trunkinbound" and that may be your issue entirely.

but experimenting by putting an "s" exten in the trunkinbound is a good test to see if it is in that context as well (to avoid wading through the debug).

by **striker** » Mon Nov 07, 2011 4:13 am

you should try this

exten => s,1,AGI(agi-DID_route.agi)

in trunkinbound and also try in default context.

by **ruben23** » Mon Nov 07, 2011 8:12 am

hi guys i tried your advice on trunkinbound and i got this logs:

Code: Select all: [ == Manager 'sendcron' logged off from 127.0.0.1 [Nov 7 08:07:43] -- Executing [s@trunkinbound:1] AGI("SIP/did1-0000005a", "agi-DID_route.agi") in new stack [Nov 7 08:07:43] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi [Nov 7 08:07:43] ERROR[14738]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Nov 7 08:07:43] ERROR[14738]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Nov 7 08:07:43] -- AGI Script agi-DID_route.agi completed, returning 0 [Nov 7 08:07:43] -- Executing [9998811112@default:1] Wait("SIP/did1-0000005a", "2") in new stack [Nov 7 08:07:45] -- Executing [9998811112@default:2] Answer("SIP/did1-0000005a", "") in new stack [Nov 7 08:07:45] -- Executing [9998811112@default:3] Playback("SIP/did1-0000005a", "ss-noservice") in new stack [Nov 7 08:07:45] -- <SIP/did1-0000005a> Playing 'ss-noservice' (language 'en') [Nov 7 08:07:51] -- Executing [9998811112@default:4] Playback("SIP/did1-0000005a", "vm-goodbye") in new stack [Nov 7 08:07:51] -- <SIP/did1-0000005a> Playing 'vm-goodbye' (language 'en') [Nov 7 08:07:51] -- Executing [9998811112@default:5] Hangup("SIP/did1-0000005a", "") in new stack [Nov 7 08:07:51] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/did1-0000005a' [Nov 7 08:07:51] -- Executing [h@default:1] DeadAGI("SIP/did1-0000005a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Nov 7 08:07:51] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed

so is it working for trunkinbound..? i used this

Code: Select all: exten => s,1,AGI(agi-DID_route.agi)

by **boybawang** » Mon Nov 07, 2011 8:41 am

if you look closely at /etc/asterisk/extensions.conf you will see the context [trunkinbound] with the dial plan under it

by **ruben23** » Mon Nov 07, 2011 8:49 am

@boybawang

Yes i see it and i have used couple dial plan on the trunkinbound context this two:

Code: Select all: exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi)

and

Code: Select all: exten => _X.,1,AGI(agi-DID_route.agi)

and

Code: Select all: exten => _16468620828.,1,AGI(agi-DID_route.agi)

still getting this error log:

[/code][Nov 5 00:23:43] NOTICE[1440]: chan_sip.c:15147 handle_request_invite: Call from '16468620828' to extension 's' rejected because extension not found.

Code: Select all

by **boybawang** » Mon Nov 07, 2011 10:47 am

do you have this exact number 16468620828 in your INBOUND DID's maybe you only put in 6468620828

by **williamconley** » Mon Nov 07, 2011 4:11 pm

ruben23 wrote:hi guys i tried your advice on trunkinbound and i got this logs:

Code: Select all
[ == Manager 'sendcron' logged off from 127.0.0.1 [Nov 7 08:07:43] -- Executing [s@trunkinbound:1] AGI("SIP/did1-0000005a", "agi-DID_route.agi") in new stack [Nov 7 08:07:43] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi [Nov 7 08:07:43] ERROR[14738]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Nov 7 08:07:43] ERROR[14738]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Nov 7 08:07:43] -- AGI Script agi-DID_route.agi completed, returning 0 [Nov 7 08:07:43] -- Executing [9998811112@default:1] Wait("SIP/did1-0000005a", "2") in new stack [Nov 7 08:07:45] -- Executing [9998811112@default:2] Answer("SIP/did1-0000005a", "") in new stack [Nov 7 08:07:45] -- Executing [9998811112@default:3] Playback("SIP/did1-0000005a", "ss-noservice") in new stack [Nov 7 08:07:45] -- <SIP/did1-0000005a> Playing 'ss-noservice' (language 'en') [Nov 7 08:07:51] -- Executing [9998811112@default:4] Playback("SIP/did1-0000005a", "vm-goodbye") in new stack [Nov 7 08:07:51] -- <SIP/did1-0000005a> Playing 'vm-goodbye' (language 'en') [Nov 7 08:07:51] -- Executing [9998811112@default:5] Hangup("SIP/did1-0000005a", "") in new stack [Nov 7 08:07:51] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/did1-0000005a' [Nov 7 08:07:51] -- Executing [h@default:1] DeadAGI("SIP/did1-0000005a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Nov 7 08:07:51] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed

so is it working for trunkinbound..? i used this

Code: Select all
exten => s,1,AGI(agi-DID_route.agi)

YES, that means it worked. Now configure the DID to do something other than go to extension 99988881111 or whatever it is. (Because that extension plays no service, goodbye, and hangs up ... which is what it is doing.)

by **boybawang** » Mon Nov 07, 2011 7:41 pm

no service error means that either the DID is installed on the inbound did list but not configured, or its not yet in the inbound did list

by **ruben23** » Mon Nov 07, 2011 8:41 pm

Hi,

this is my Inbound DID settings:

[/code]

by **williamconley** » Mon Nov 07, 2011 10:19 pm

because this is configured as "s" instead of the actual DID and it works, this means that the method they are using to send you the DID is not being "caught" by your server.

temporarily configure the DEFAULT DID to your requirements (to get your call routed where you want it right now) and then investigate the options available to you to identify the DID specifically when the call comes in.

Often this can be done through sip debug ... search for the header that contains the actual DID and then you can research how to "move" this information to the exten so Vicidial can be made aware of the number.

by **boybawang** » Tue Nov 08, 2011 8:05 am

sip debug is much better so you can see what the sip originator is really sending then that will be the value you will put in the inbound did , did you try also putting another entry of the same number without '1'

i have encountered this problem with vitelity before

by **williamconley** » Tue Nov 08, 2011 4:38 pm

[Nov 7 08:07:43] -- Executing [s@trunkinbound:1] AGI("SIP/did1-0000005a", "agi-DID_route.agi") in new stack

if there are lines before this indicating that the system was trying a different extension, which failed, then the system tried "s" because of the failure, you can find the prior exten and that will be your "DID". If no such failure happened, then Asterisk is not being passed the DID in a manner that this system can detect, and you need to use sip debug to find out if it is IN the sip package at all. Then you can tackle how to capture it and use it. 8-)

by **ruben23** » Wed Nov 09, 2011 6:16 pm

hi there guys the inbound DID is working now but i got problem it only take 17 seconds on every call and it automatically disconnect on every inboudn call to that DID, i tried diffrent carrier but still the same

and base on the call logs i see this error during the calls are getiing cut off

Code: Select all: [Nov 9 18:03:48] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 9 18:03:58] WARNING[1417]: chan_sip.c:2015 retrans_pkt: Maximum retries exceeded on transmission 2fba714616acf9e81142a44e1533a1bd@46.19.209.25-b2b_1 for seqno 200 (Critical Response) -- See doc/sip-retransmit.txt. [Nov 9 18:03:58] WARNING[1417]: chan_sip.c:2037 retrans_pkt: Hanging up call 2fba714616acf9e81142a44e1533a1bd@46.19.209.25-b2b_1 - no reply to our critical packet (see doc/sip-retransmit.txt). [Nov 9 18:03:58] == Spawn extension (default, 8600054, 1) exited non-zero on 'SIP/didww_us5-000000d8' [Nov 9 18:03:58] -- Executing [h@default:1] DeadAGI("SIP/didww_us5-000000d8", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18---------------") in new stack [Nov 9 18:03:58] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18--------------- completed, returning 0

any idea on this guys..?

by **ruben23** » Fri Nov 11, 2011 9:15 am

hi guys any idea on same issue i have, who experience this same issue.

by **boybawang** » Fri Nov 11, 2011 2:05 pm

try reinstalling asterisk only, also check the sip details of the voip provider, do a sip debug and look at the sip invite

by **williamconley** » Sat Nov 12, 2011 7:17 pm

was there sound during the call? if not, your firewall is blocking the sound in at least one direction.

describe your network.

by **daccad** » Thu Nov 17, 2011 12:18 pm

williamconley wrote:was there sound during the call? if not, your firewall is blocking the sound in at least one direction.

describe your network.

Hi William
In facing the same problem of Ruben

Code: Select all: [Nov 17 09:58:38] -- Executing [8600055@default:1] MeetMe("SIP/inDID-00000014", "8600055|F") in new stack [Nov 17 09:58:38] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 17 09:58:47] WARNING[3709]: chan_sip.c:2058 retrans_pkt: Maximum retries exceeded on transmission 6338f12132c4836a3138d95c16992c23@46.19.209.24-b2b_1 for seqno 200 (Critical Response) -- See doc/sip-retransmit.txt. [Nov 17 09:58:47] WARNING[3709]: chan_sip.c:2080 [color=red]retrans_pkt:[/color] Hanging up call 6338f12132c4836a3138d95c16992c23@XX.XX.XXX.XX-b2b_1 - no reply to our critical packet (see doc/sip-retransmit.txt). [Nov 17 09:58:47] == Spawn extension (default, 8600055, 1) exited non-zero on 'SIP/inDID-00000014' [Nov 17 09:58:47] -- Executing [h@default:1] DeadAGI("SIP/inDID-00000014", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18---------------") in new stack

My server is plugged direct to my ISP router (FIREWALL OFF), no active firewall on the server.
Server has 2 NICS (we used to have local and wan access to that server, not anymore)
Something very weird, If I want my server have internet access I need to use PCI NIC card, if I try with my integrated Network port, I got no internet access (both NIC are correctly configured).

I hope you can help me to get this solved, this is a real torture

by **daccad** » Sun Nov 20, 2011 12:53 pm

daccad wrote:
williamconley wrote:was there sound during the call? if not, your firewall is blocking the sound in at least one direction.

describe your network.

I william, hope you could help me.

Now My server is working with one NIC, no firewall installed and still having the same problem.
When I get the inbound call, I have no sound during the call.
All kind of firewalls are off.
I try another server, an older version in the same network environment and works perfect, that means problem must be with the server.
Here is a description of both servers.
NOT WORKING SERVER:
VERSION: 2.4-309a
BUILD: 110430-1642
GOaUTOdIAL 2.1
Intel Core i3 CPU 540 3.07GHz, RAM 4 gb DDR3

WORKING SERVER
Version: 2.0.5-174
Build #: 90522-0506
Pentium 4 2.4 GHz, 1.5 RAM DDR

by **williamconley** » Wed Nov 23, 2011 12:54 pm

have you verified the sip.conf settings are the same in both machines? (at least as related to IP addresses ... local networks and externip are crucial)

also: what is the IP of your server? Is it a private address (192.168.x.x or 10.x.x ...?)

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