vicidial.org

My current inbound flow works great.

However, our management wants to convert from the usual softphone to ip hardphones and they want to utilize the On-Hook Feature of GoAutoDial.

If I enable this feature, the phone would ring but when I answer it it says "Not a valid extension, please try again".

Do I need to modify any code for this to work? Configuration that I may have overlooked?

Thank you.

what protocal you have selected for the onhook phones

if it is sip based IPPhones , then select SIP as protocal .


still have problem
post the cli data

striker wrote:what protocal you have selected for the onhook phones

if it is sip based IPPhones , then select SIP as protocal .


still have problem
post the cli data

Hi striker.. Yes, it is a sip based IP Phone.. I already selected sip as protocol.

CLI output:
[Feb 7 08:48:41] -- Executing [8888872XXX@trunkinbound:1] AGI("SIP/AccelaSa
nsay1-000000a3", "agi-DID_route.agi") in new stack
[Feb 7 08:48:41] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_r
oute.agi
[Feb 7 08:48:41] -- AGI Script agi-DID_route.agi completed, returning 0
[Feb 7 08:48:41] -- Executing [s@NewLeaf:1] AGI("SIP/AccelaSansay1-000000a3
", "agi-VDAD_inbound_calltime_check.agi|NewLeaf-----YES-----NewLeaf-------------
-------") in new stack
[Feb 7 08:48:41] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_
inbound_calltime_check.agising '/etc/asterisk/manager.conf': [Feb 7 08:43:57] F
[Feb 7 08:48:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)Feb 7 08:43:57] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 08:48:41] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)Feb 7 08:43:57] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
[Feb 7 08:48:41] -- AGI Script agi-VDAD_inbound_calltime_check.agi complete
d, returning 001] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:44:01] F
[Feb 7 08:48:41] -- Executing [s@NewLeaf:2] BackGround("SIP/AccelaSansay1-0
00000a3", "nss") in new stackr 'sendcron' logged on from 127.0.0.1
[Feb 7 08:48:41] -- <SIP/AccelaSansay1-000000a3> Playing 'nss' (language 'e
n')b 7 08:44:02] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:44:02] F
[Feb 7 08:48:49] == CDR updated on SIP/AccelaSansay1-000000a3
[Feb 7 08:48:49] -- Executing [2030@NewLeaf:1] Goto("SIP/AccelaSansay1-0000
00a3", "trunkinbound|2030|1") in new stacklogged off from 127.0.0.1
[Feb 7 08:48:49] -- Goto (trunkinbound,2030,1)er.conf': [Feb 7 08:44:06] F
[Feb 7 08:48:49] -- Executing [2030@trunkinbound:1] AGI("SIP/AccelaSansay1-
000000a3", "agi-DID_route.agi") in new stackgged on from 127.0.0.1
[Feb 7 08:48:49] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_r
oute.agi08:44:27] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:44:27] F
[Feb 7 08:48:49] -- AGI Script agi-DID_route.agi completed, returning 0
[Feb 7 08:48:49] -- Executing [99909*17***DID@default:1] Answer("SIP/Accela
Sansay1-000000a3", "") in new stack[8600052@default:1] MeetMe("Local/8600052@def
[Feb 7 08:48:49] -- Executing [99909*17***DID@default:2] AGI("SIP/AccelaSan
say1-000000a3", "agi-VDAD_ALL_inbound.agi") in new stack-f13d,1 was answered.
[Feb 7 08:48:49] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_
ALL_inbound.agi"agi://127.0.0.1:4577/call_log") in new stack
[Feb 7 08:48:49] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)Feb 7 08:44:27] -- AGI Script agi://127.0.0.1:4577/call_log completed, ret
[Feb 7 08:48:49] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)Feb 7 08:44:27] -- Executing [14136950531@default:2] Dial("Local/8600052@d
[Feb 7 08:48:49] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:48:49] F
ound 7 08:44:27] -- Called 14136950531@AccelaSansay1
[Feb 7 08:48:49] == Manager 'sendcron' logged on from 127.0.0.1 7 08:44:27] F
[Feb 7 08:48:49] -- Executing [192*168*001*202*2030@default:1] Dial("Local/
192*168*001*202*2030@default-7571,2", "IAX2/Vicidial1:goautodial@192.168.1.202:4
569/2030|55|oT") in new stackuting [58600052@default:1] MeetMe("Local/58600052@d
[Feb 7 08:48:49] -- Called Vicidial1:goautodial@192.168.1.202:4569/2030
[Feb 7 08:48:49] -- Call accepted by 192.168.1.202 (format ulaw)
[Feb 7 08:48:49] -- Format for call is ulaw:1] Answer("Local/58600052@defau
[Feb 7 08:48:50] -- IAX2/Vicidial2-7505 is ringing
[Feb 7 08:48:51] -- Started music on hold, class 'default', on SIP/AccelaSa
nsay1-000000a3wav|CREDITCR_20120207-084426_4136950531_nfdoronila_304018") in new
[Feb 7 08:48:55] -- Stopped music on hold on SIP/AccelaSansay1-000000a3
[Feb 7 08:48:55] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)df72,1", "3600") in new stack
[Feb 7 08:48:55] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)Feb 7 08:44:28] -- SIP/AccelaSansay1-000000a1 is making progress passing i
[Feb 7 08:48:55] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0
)Feb 7 08:44:59] -- SIP/AccelaSansay1-000000a1 answered Local/8600052@defau
[Feb 7 08:48:55] -- Playing 'generic_hold' (escape_digits=) (sample_offset
0)eb 7 08:45:00] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:45:00] F
[Feb 7 08:48:56] -- IAX2/Vicidial2-7505 stopped sounds
[Feb 7 08:48:56] -- IAX2/Vicidial2-7505 answered Local/192*168*001*202*2030
@default-7571,20] == Spawn extension (default, 58600052, 1) exited non-zero on
[Feb 7 08:48:56] > Channel Local/192*168*001*202*2030@default-7571,1 was
answered.:45:00] -- Executing [h@default:1] DeadAGI("Local/58600052@default
[Feb 7 08:48:56] == Starting Local/192*168*001*202*2030@default-7571,1 at def
ault,8331*696*Y2070848490000304075*aj*2030,1 failed so falling back to exten 's'
[Feb 7 08:45:00] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:48:56] == Starting Local/192*168*001*202*2030@default-7571,1 at def
ault,s,1 still failed so falling back to context 'default'
[Feb 7 08:48:56] -- Sent into invalid extension 's' in context 'default' on
Local/192*168*001*202*2030@default-7571,1logged off from 127.0.0.1
[Feb 7 08:48:56] -- Executing [i@default:1] Playback("Local/192*168*001*202
*2030@default-7571,1", "invalid") in new stack
[Feb 7 08:48:56] -- <Local/192*168*001*202*2030@default-7571,1> Playing 'in
valid' (language 'en')0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------
[Feb 7 08:48:56] WARNING[8718]: file.c:1297 waitstream_core: Unexpected control
subclass '-1'00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
[Feb 7 08:48:56] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:48:56] WARNING[8718]: file.c:1297 waitstream_core: Unexpected control
subclass '-1'efault-df72,1'
[Feb 7 08:48:56] -- Executing [h@default:1] DeadAGI("Local/192*168*001*202*
2030@default-7571,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-
----16-----ANSWER-----7-----0") in new stack
[Feb 7 08:48:56] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----16-----ANSWER-----7-----0 completed, returning 0
[Feb 7 08:48:56] == Spawn extension (default, 192*168*001*202*2030, 1) exited
non-zero on 'Local/192*168*001*202*2030@default-7571,2'
[Feb 7 08:49:00] == Auto fallthrough, channel 'IAX2/Vicidial2-7505' status is
'UNKNOWN'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----AN
[Feb 7 08:49:00] -- Executing [h@default:1] DeadAGI("IAX2/Vicidial2-7505",
"agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------")
in new stack:06] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:49:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----0--------------- completed, returning 0
[Feb 7 08:49:00] -- Hungup 'IAX2/Vicidial2-7505'rom 127.0.0.1
[Feb 7 08:49:01] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:49:01] F
ound-fdcd,2", "8600052|F") in new stackn' logged on from 127.0.0.1
[Feb 7 08:49:01] == Manager 'sendcron' logged on from 127.0.0.1s answered.
[Feb 7 08:49:01] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:49:01] F
oundt-fdcd,1", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 7 08:49:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 08:49:01] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:49:01] -- Started music on hold, class 'default', on SIP/AccelaSa
nsay1-000000a316] -- Executing [14136950531@default:2] Dial("Local/8600052@d
[Feb 7 08:49:02] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:49:05] -- Stopped music on hold on SIP/AccelaSansay1-000000a3
[Feb 7 08:49:05] == Spawn extension (default, 99909*17***DID, 2) exited non-z
ero on 'SIP/AccelaSansay1-000000a3'dcron' logged on from 127.0.0.1
[Feb 7 08:49:05] -- Executing [h@default:1] DeadAGI("SIP/AccelaSansay1-0000
00a3", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0----------
-----") in new stack0052|Fmq") in new stack
[Feb 7 08:49:05] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----0--------------- completed, returning 0er("Local/58600052@defau
[Feb 7 08:49:06] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:49:06] F
ound 7 08:45:17] -- Executing [8309@default:2] Monitor("Local/58600052@defa
[Feb 7 08:49:06] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 08:49:06] == Manager 'sendcron' logged off from 127.0.0.1
go*CLI>
-c9b6,1", "3600") in new stack
[Feb 7 08:45:17] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:45:17] -- SIP/AccelaSansay1-000000a2 is making progress passing i
t to Local/8600052@default-fdcd,1
[Feb 7 08:45:50] -- SIP/AccelaSansay1-000000a2 answered Local/8600052@defau
lt-fdcd,1
[Feb 7 08:45:52] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:45:52] F
ound
[Feb 7 08:45:52] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 08:45:52] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:45:52] == Spawn extension (default, 58600052, 1) exited non-zero on
'Local/58600052@default-c9b6,2'
[Feb 7 08:45:52] -- Executing [h@default:1] DeadAGI("Local/58600052@default
-c9b6,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------
--------") in new stack
[Feb 7 08:45:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----0--------------- completed, returning 0
[Feb 7 08:45:52] == Spawn extension (default, 8309, 3) exited non-zero on 'Lo
cal/58600052@default-c9b6,1'
[Feb 7 08:45:52] -- Executing [h@default:1] DeadAGI("Local/58600052@default
-c9b6,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------
--------") in new stack
[Feb 7 08:45:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----0--------------- completed, returning 0
[Feb 7 08:45:52] == Parsing '/etc/asterisk/manager.conf': [Feb 7 08:45:52] F
ound
[Feb 7 08:45:52] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 08:45:52] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 08:45:52] -- Executing [h@default:1] DeadAGI("Local/8600052@default-
fdcd,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----AN
SWER-----36-----2") in new stack
[Feb 7 08:45:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----16-----ANSWER-----36-----2 completed, returning 0
[Feb 7 08:45:52] == Spawn extension (default, 14136950531, 2) exited non-zero
on 'Local/8600052@default-fdcd,1'
[Feb 7 08:45:52] WARNING[7187]: app_meetme.c:2463 conf_run: Unable to write fra
me to channel Local/8600052@default-fdcd,2
[Feb 7 08:45:52] == Spawn extension (default, 8600052, 1) exited non-zero on
'Local/8600052@default-fdcd,2'
[Feb 7 08:45:52] -- Executing [h@default:1] DeadAGI("Local/8600052@default-
fdcd,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------
-------") in new stack
[Feb 7 08:45:52] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI
-----NODEBUG-----0--------------- completed, returning 0

extension 2030 is not in your dialplan, that is the problem, it's just a configuration issue.

mflorell wrote:extension 2030 is not in your dialplan, that is the problem, it's just a configuration issue.

Hi mflorell,

I created a DID(2030) that would route to agent and can be called in a callmenu. This works if "On-Hook Agent" option is set to "N", but it would return a 'Not a valid extension' if the option is set to "Y". As you can see in the log, the phone is actually ringing but i got that recording when i answer it.
Do i need to adjust something?

Thank you.

DIDs are not entered into the dialplan, you need to create a phones entry with that as the dialplan number.

mflorell wrote:DIDs are not entered into the dialplan, you need to create a phones entry with that as the dialplan number.

Oh.. i forgot to mention, i already created a phone extension 2030.
And I made a custom dial plan entry on the call menu so that if the caller enters the extension 2030 it will be routed to DID 2030. And DID 2030 is routed to the Agent.

I reproduced the same error on the softphone, I got it working but once i changed the on-hook feature i got the same "Sorry, not a valid extension, please try again"

Call Menu custom dialplan entries only exist within that Call Menu, which is not part of the default dialplan context. If you want a custom entry to be part of the default dialplan you need to enter it in System Settings or within a specific server's custom dialplan entry.

mflorell wrote:Call Menu custom dialplan entries only exist within that Call Menu, which is not part of the default dialplan context. If you want a custom entry to be part of the default dialplan you need to enter it in System Settings or within a specific server's custom dialplan entry.

Mmmm.. I hope you'll be more patient with me.

The whole inbound call flow works flawlessly.. which means if the caller enters the extension number when prompted in the call menu, the call is then forwarded to the agent. Im using DID to trap the numbers entered by the caller and the system does not literally forward it to a specific extension but to an agent. (Which means DID route = "Agent")

ON-HOOK Agent
(N)If this feature is disabled, the softphone rings when the agent logs in and when the agent answers he/she can hear "You are the only one in the conference", normal call can be transferred to the agent. (Working)

(Y)If this feature is enabled, the softphone does not ring when the agent logs in. The softphone only rings when there's an incoming call.. and when the agent answers he/she can hear "Sorry! Invalid Extension.. Please try again". No calls can be received by the agent.

I removed the call menu and forwarded the inbound call directly to the agent.. same thing.. it rings and when he answers it says "sorry, that's not a valid extension",
but when the "On-Hook Agent" is off then call is forwarded correctly.

Is "On-Hook Agent" on phone configuration working? I checked on the features http://www.vicidial.org/VICIDIALforum/v ... hp?t=16755 and we have the same issue.

mflorell wrote:DIDs are not entered into the dialplan, you need to create a phones entry with that as the dialplan number.

Hi Matt,

Just want to ask.. If in a normal user log-in, it creates a meet-me conference.. Is it also the same when the phone's on-hook option is enabled? Or does the creation of the conference occur when the call comes in?

Thank you very much for you help.

Yes, in both cases the conference is created and used.

mflorell wrote:Yes, in both cases the conference is created and used.

Thank you Matt for your prompt reply.
Here's the CLI messages i got:

With On-Hook Agent = N

[Feb 16 05:48:01] > Channel SIP/2030-000000e8 was answered.
[Feb 16 05:48:01] -- Executing [8700056@default:1] MeetMe("SIP/2030-000000e8", "8700056|F") in new stack
[Feb 16 05:48:01] == Parsing '/etc/asterisk/meetme.conf': [Feb 16 05:48:01] Found
[Feb 16 05:48:01] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Feb 16 05:48:01] Found
[Feb 16 05:48:01] -- Created MeetMe conference 1016 for conference '8700056'
[Feb 16 05:48:01] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 16 05:48:01] -- <SIP/2030-000000e8> Playing 'conf-onlyperson' (language 'en')

-------------------------------------------------------------------------------
With On-Hook Agent = Y

[Feb 16 05:50:45] -- Accepting AUTHENTICATED call from 192.168.1.201:
> requested format = gsm,
> requested prefs = (gsm|ulaw),
> actual format = ulaw,
> host prefs = (ulaw),
> priority = mine
[Feb 16 05:50:45] -- Executing [2030@default:1] Dial("IAX2/Vicidial1-1728", "SIP/2030|60|") in new stack
[Feb 16 05:50:45] -- Called 2030
[Feb 16 05:50:45] -- SIP/2030-000000e9 is ringing
[Feb 16 05:50:48] -- SIP/2030-000000e9 answered IAX2/Vicidial1-1728
[Feb 16 05:50:53] -- Executing [h@default:1] DeadAGI("IAX2/Vicidial1-1728", "agi://127.0.0.1:4577/call_log--HVcauses
--PRI-----NODEBUG-----16-----ANSWER-----8-----5") in new stack
[Feb 16 05:50:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -----8----
-5 completed, returning 0
[Feb 16 05:50:53] == Spawn extension (default, 2030, 1) exited non-zero on 'IAX2/Vicidial1-1728'
[Feb 16 05:50:53] -- Hungup 'IAX2/Vicidial1-1728'

------------------------------------------------------------------------------
I dont understand why it wont work if i adjust the phone's settings.

Hi

i'm having the same issues with ON-HOOK = Y, getting not valid extension message if the agent answered the call..

I'm using eyebeam. ON-HOOK = N, its working ok..

thanks

GoAutodial 2.1 stock

Hi sadikhov,

I did have the same issue on my goautodial installation. I was able to resolved this by adding the following entries on the extensions.conf inside the default context.

; these are used for the ring_all function in VICIDIAL
exten => _8331*.,1,Playback(sip-silence)
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,Hangup

Hope this helps.

lcape wrote:Hi sadikhov,

I did have the same issue on my goautodial installation. I was able to resolved this by adding the following entries on the extensions.conf inside the default context.

; these are used for the ring_all function in VICIDIAL
exten => _8331*.,1,Playback(sip-silence)
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,Hangup

Hope this helps.

Hi lcape,

Tried adding this to my extensions.conf and it worked beautifully.

Thank you very much.

Hi,

With On-Hook = Y, when answering the softphone(eyebeam) it does not answer or connect to the web agent form thus it hangs up.
and then it will ring again then when answered it will be hanged up again.

what seems to be the problem?? thanks very much!



vicibox 2.6-388a

1) Welcome to the Party!

2) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Does this dialer work properly with on-hook = N? And to verify: Is this agent a logged in agent or a remote agent?

4) CLI output (possibly with agi debugging enabled) would be quite helpful (or /var/log/astguiclient logs). Please only post the relevant log/output. Not 3000 lines of "unrelated code". So try this with a controlled environment on a single call with no other traffic.

Will be grateful if somebody could help with the issue:

lcape wrote:Hi sadikhov,

I did have the same issue on my goautodial installation. I was able to resolved this by adding the following entries on the extensions.conf inside the default context.

; these are used for the ring_all function in VICIDIAL
exten => _8331*.,1,Playback(sip-silence)
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,Hangup

Hope this helps.

I am using GoAutoDial CE 2.1
I have setup an inbound campaign and I am using 2 on-hook agents for the same
I have also used the ring_all fucntion by mentioning the RINGALL AGI entries in the default context inside extensions.conf

I still hear the "Not a Valid Extension" when somebody is calling in
Any help would be appreciated

Thanks

post your asterisk cli log while receiving the call.

Thanks for your quick response!

Here's the CLI Log:

-- Executing [441183111032@trunkinbound:1] AGI("SIP/tsc1-0000000f", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20150611105828_441183111032_8050)
[Jun 11 10:58:28] ERROR[5074]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
-- AGI Script agi-DID_route.agi completed, returning 0
-- Executing [99909*3***DID@default:1] Answer("SIP/tsc1-0000000f", "") in new stack
-- Executing [99909*3***DID@default:2] AGI("SIP/tsc1-0000000f", "agi-VDAD_ALL_inbound.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [192*168*001*043*8001@default:1] Goto("Local/192*168*001*043*8001@default-9b87,2", "default|8001|1") in new stack
-- Goto (default,8001,1)
-- Executing [8001@default:1] Dial("Local/192*168*001*043*8001@default-9b87,2", "SIP/8001|60|") in new stack
-- Called 8001
-- SIP/8001-00000010 is ringing
-- Started music on hold, class 'tsc1', on SIP/tsc1-0000000f
-- SIP/8001-00000010 answered Local/192*168*001*043*8001@default-9b87,2
> Channel Local/192*168*001*043*8001@default-9b87,1 was answered.
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:1] Playback("Local/192*168*001*043*8001@default-9b87,1", "sip-silence") in new stack
-- <Local/192*168*001*043*8001@default-9b87,1> Playing 'sip-silence' (language 'en')
[Jun 11 10:58:34] WARNING[5113]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:2] AGI("Local/192*168*001*043*8001@default-9b87,1", "agi-VDAD_RINGALL.agi|8331*14*Y6111058280000000037*agent001*8001") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:3] Hangup("Local/192*168*001*043*8001@default-9b87,1", "") in new stack
== Spawn extension (default, 8331*14*Y6111058280000000037*agent001*8001, 3) exited non-zero on 'Local/192*168*001*043*8001@default-9b87,1'
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-9b87,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-9b87,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----1") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----1 completed, returning 0
== Spawn extension (default, 8001, 1) exited non-zero on 'Local/192*168*001*043*8001@default-9b87,2'
-- Stopped music on hold on SIP/tsc1-0000000f
-- Started music on hold, class 'tsc1', on SIP/tsc1-0000000f
== Manager 'sendcron' logged off from 127.0.0.1
[Jun 11 10:58:44] NOTICE[5074]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 212.38.95.161
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [192*168*001*043*8001@default:1] Goto("Local/192*168*001*043*8001@default-8c03,2", "default|8001|1") in new stack
-- Goto (default,8001,1)
-- Executing [8001@default:1] Dial("Local/192*168*001*043*8001@default-8c03,2", "SIP/8001|60|") in new stack
-- Called 8001
-- SIP/8001-00000011 is ringing
-- SIP/8001-00000011 answered Local/192*168*001*043*8001@default-8c03,2
> Channel Local/192*168*001*043*8001@default-8c03,1 was answered.
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:1] Playback("Local/192*168*001*043*8001@default-8c03,1", "sip-silence") in new stack
-- <Local/192*168*001*043*8001@default-8c03,1> Playing 'sip-silence' (language 'en')
[Jun 11 10:58:49] WARNING[5181]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:2] AGI("Local/192*168*001*043*8001@default-8c03,1", "agi-VDAD_RINGALL.agi|8331*14*Y6111058280000000037*agent001*8001") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:3] Hangup("Local/192*168*001*043*8001@default-8c03,1", "") in new stack
== Spawn extension (default, 8331*14*Y6111058280000000037*agent001*8001, 3) exited non-zero on 'Local/192*168*001*043*8001@default-8c03,1'
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-8c03,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-8c03,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---4-----0 completed, returning 0
== Spawn extension (default, 8001, 1) exited non-zero on 'Local/192*168*001*043*8001@default-8c03,2'
== Manager 'sendcron' logged off from 127.0.0.1
-- Stopped music on hold on SIP/tsc1-0000000f
== Spawn extension (default, 99909*3***DID, 2) exited non-zero on 'SIP/tsc1-0000000f'
-- Executing [h@default:1] DeadAGI("SIP/tsc1-0000000f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

i can see the calls landing to your extensions
Called 8001
-- SIP/8001-00000010 is ringing


who is getting the message "not valid extension"

customer or agent?

The agent is hearing the message "Not a Valid Extension" and then the call is getting disconnected from the agent's side on its own. The customer is listening to the MOH music during the entire time. The customer is never getting disconnected in the meanwhile

rathin.saran wrote:Will be grateful if somebody could help with the issue:
I am using GoAutoDial CE 2.1
I have setup an inbound campaign and I am using 2 on-hook agents for the same
I have also used the ring_all fucntion by mentioning the RINGALL AGI entries in the default context inside extensions.conf

I still hear the "Not a Valid Extension" when somebody is calling in
Any help would be appreciated

1) If you have edited "extensions.conf" please revert to the original. That file is not meant to be edited.
2) Please provide CLI which includes the entire phone call and especially the part where "not a valid extension" occurs, as that is your problem.
3) Ringall is a setting in the InGroup, and does not require customization of any configuration files.

@williamconley

Thanks for your response! I appreciate that you're looking into the issue. I would be grateful if you could help me resolve this

I was going through this post and it mentioned that there are issues with the on-hook agent for an inbound setup. Right now, I am not getting a "not a valid extension" message any more after deleted the INGROUP, DID & CAMPAIGN and started afresh.

Right now, there is 1 on-hook IP PHONE, a DID pointed to an INGROUP, and the INGROUP assigned to a Blended Campaign. Things are running fine with ON HOOK = N. I am trying to configure the agent's IP PHONE to RING when there's an incoming call. For that, I have set "Next Agent Call" to RINGALL in the INGROUP and ON HOOK = Y in the Phone Setting

Right now, when the IP PHONE is ringing, and an agent is picking up the call, it is getting disconnected without any message. And the call is ringing again and again

1) The following addition inside extensions.conf was supposed to solve the issue according to the current post:
; these are used for the ring_all function in VICIDIAL
exten => _8331*.,1,Playback(sip-silence)
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,AGI(agi-VDAD_RINGALL.agi,${EXTEN})
exten => _8331*.,n,Hangup

2) Here's the CLI for an incoming call:
-- Executing [441183111032@trunkinbound:1] AGI("SIP/tsc1-0000000f", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20150611105828_441183111032_8050)
[Jun 11 10:58:28] ERROR[5074]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
-- AGI Script agi-DID_route.agi completed, returning 0
-- Executing [99909*3***DID@default:1] Answer("SIP/tsc1-0000000f", "") in new stack
-- Executing [99909*3***DID@default:2] AGI("SIP/tsc1-0000000f", "agi-VDAD_ALL_inbound.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [192*168*001*043*8001@default:1] Goto("Local/192*168*001*043*8001@default-9b87,2", "default|8001|1") in new stack
-- Goto (default,8001,1)
-- Executing [8001@default:1] Dial("Local/192*168*001*043*8001@default-9b87,2", "SIP/8001|60|") in new stack
-- Called 8001
-- SIP/8001-00000010 is ringing
-- Started music on hold, class 'tsc1', on SIP/tsc1-0000000f
-- SIP/8001-00000010 answered Local/192*168*001*043*8001@default-9b87,2
> Channel Local/192*168*001*043*8001@default-9b87,1 was answered.
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:1] Playback("Local/192*168*001*043*8001@default-9b87,1", "sip-silence") in new stack
-- <Local/192*168*001*043*8001@default-9b87,1> Playing 'sip-silence' (language 'en')
[Jun 11 10:58:34] WARNING[5113]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:2] AGI("Local/192*168*001*043*8001@default-9b87,1", "agi-VDAD_RINGALL.agi|8331*14*Y6111058280000000037*agent001*8001") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:3] Hangup("Local/192*168*001*043*8001@default-9b87,1", "") in new stack
== Spawn extension (default, 8331*14*Y6111058280000000037*agent001*8001, 3) exited non-zero on 'Local/192*168*001*043*8001@default-9b87,1'
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-9b87,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-9b87,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----1") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----1 completed, returning 0
== Spawn extension (default, 8001, 1) exited non-zero on 'Local/192*168*001*043*8001@default-9b87,2'
-- Stopped music on hold on SIP/tsc1-0000000f
-- Started music on hold, class 'tsc1', on SIP/tsc1-0000000f
== Manager 'sendcron' logged off from 127.0.0.1
[Jun 11 10:58:44] NOTICE[5074]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 212.38.95.161
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [192*168*001*043*8001@default:1] Goto("Local/192*168*001*043*8001@default-8c03,2", "default|8001|1") in new stack
-- Goto (default,8001,1)
-- Executing [8001@default:1] Dial("Local/192*168*001*043*8001@default-8c03,2", "SIP/8001|60|") in new stack
-- Called 8001
-- SIP/8001-00000011 is ringing
-- SIP/8001-00000011 answered Local/192*168*001*043*8001@default-8c03,2
> Channel Local/192*168*001*043*8001@default-8c03,1 was answered.
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:1] Playback("Local/192*168*001*043*8001@default-8c03,1", "sip-silence") in new stack
-- <Local/192*168*001*043*8001@default-8c03,1> Playing 'sip-silence' (language 'en')
[Jun 11 10:58:49] WARNING[5181]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:2] AGI("Local/192*168*001*043*8001@default-8c03,1", "agi-VDAD_RINGALL.agi|8331*14*Y6111058280000000037*agent001*8001") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- AGI Script agi-VDAD_RINGALL.agi completed, returning 0
-- Executing [8331*14*Y6111058280000000037*agent001*8001@default:3] Hangup("Local/192*168*001*043*8001@default-8c03,1", "") in new stack
== Spawn extension (default, 8331*14*Y6111058280000000037*agent001*8001, 3) exited non-zero on 'Local/192*168*001*043*8001@default-8c03,1'
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-8c03,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing [h@default:1] DeadAGI("Local/192*168*001*043*8001@default-8c03,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---4-----0 completed, returning 0
== Spawn extension (default, 8001, 1) exited non-zero on 'Local/192*168*001*043*8001@default-8c03,2'
== Manager 'sendcron' logged off from 127.0.0.1
-- Stopped music on hold on SIP/tsc1-0000000f
== Spawn extension (default, 99909*3***DID, 2) exited non-zero on 'SIP/tsc1-0000000f'
-- Executing [h@default:1] DeadAGI("SIP/tsc1-0000000f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

1) If 8331 was not in your extensions.conf file, it may be a simpler solution to upgrade as there may be other problems. You'll lose the pretty goautodial screens (unless you upgrade to a later version of Goautodial instead of a standard Vicidial upgrade ...).

2) If you insist on beating your head against this (likely, LOL): Try getting agi debug output to see what the ringall.agi script is doing. Also try using "screen -r asterisk" to view the console instead of "asterisk -r" (ctrl-a ctrl-d exits, do not use ctrl-c as it will crash asterisk and you'll need to restart it!).

Code: Select all

agi debug
agi no debug

3) My best advice: Install a fresh machine with Vicibox 6.0.3. Port your database into that machine. Upgrade your database to match the new vicidial version. Update the IP using the ip update script from the old server to the new one. You'll have a brand new version of Vicidial and this problem will likely go *poof*.

Happy Hunting!

Thanks @williamconley

I will install Vicibox right away!

1) Install with the Standard (not preload) 6.0.3 Vicibox installer. From Vicibox.com

2) Use the PDF from Vicibox.com (which is not perfect, but will get you there ...) which will get you a running server, but nothing configured in Vicidial.

3) Once that has been accomplished, switch to the Vicidial Manager's Manual (from EFLO.net, I recommend the paid version ... if not today, then as soon as you've shown yourself that it works ... there's A LOT of good info in the paid one!) and configure the system by working from page one through to a "Functional" system. Don't skip any pages! Seriously. At that point, you can test your "not working on the old system" problem out and see if it works on this one. If it does: You have a decision to make ...: Just rebuild everything with this new, working server ... or bring your database over from the old server.

If you choose to bring the DB over from the old server ... here are the basic steps (which you can do instead of #3 above if you are a daredevil, but I like to be sure the server Works before porting the old database in .. that way if there's a problem, I know it's not the Vicidial install but the newly added database!):

* mysqldump on the old server to get the asterisk database (mysqldump asterisk -user cron -p1234 > old_asterisk.sql) you may also need ""
* copy this file to the new server
* mysqldump the asterisk DB on the NEW server (so you can GO BACK if you want to!!!).
* drop the asterisk DB on the new server (drop database asterisk)
* create a new blank DB on the new server (create database asterisk)
* load the old server's DB on the new server (mysql asterisk < old_asterisk.sql)
* upgrade the db by applying the sql scripts against the asterisk DB starting with the version that was already installed (this will be in the /usr/src/astguiclient/trunk/extras folder ...). One at a time (mysql asterisk < upgrade_2.4.sql). I like to delete all the lines that have already been run. The "already run" can be found by checking the DB schema number in the old servers system settings (admin->system settings). Since that db schema is already there, you can delete any mysql commands before the one that changes the DB schema to that number (I've always done this, so I can not guarantee it is necessary, or even helpful, as opposed to just a waste of time).
* run the ip update script (which appears in the ssh/console splash screen when you log in). Note that you will need to manually enter the old server's IP, it will already know the new server's IP. This will change the old server's IP address in all necessary locations in the DB

Remember that the DB can be dumped and reloaded as often as you like. No harm. you can always put the original db from the new server back and it will be just like it was before you dumped it. You may need to reboot to make everything match up, since some startup scripts may be affected by the database during startup.

happy hunting!