Manual Dial Problem

[Srinivasa Rao]

Hi Everyone,

Need Help,

Following are my server details

1. Asterisk 1.4.42
2. vicidial 2.2.1
3. 4 Port Digium Card
4. Cent OS
5. Soft Phone Zoiper

I am using the vicidial application for the past one year, it is very nice.
If i call to any normal number through manual dial, it is working fine.

But if i call to ivr number through manual dial, it is asking me to enter input (for suppose dial 1 for English) , really i don't know where to enter the input.

I have entered in zoiper (my soft phone) channel text box but its not taking the input.


Please help me to sort out this problem.

[striker]

if you are dialling through the agent interface, then there is a option in the agent inerface "SEND DTMF "

to press 1 enter 1 and press the SEND DTMF BUTTON

[Srinivasa Rao]

Thank you striker for your immediate response.

I am dialing through agent interface only.

Now i have tried with the send dtmf option, but my bad luck is its not working for me.

In the asterisk console i have seen the following log


Executing [8500998@default:4] AGI("Local/8500998@default-8ed8,2", "agi-dtmf.agi") in new stack
> Channel Local/8500998@default-8ed8,1 was answered.
== Starting Local/8500998@default-8ed8,1 at default,78600055,1 failed so falling back to exten 's'
== Starting Local/8500998@default-8ed8,1 at default,s,1 still failed so falling back to context 'default'
[Jan 2 15:32:05] WARNING[3274]: pbx.c:2525 __ast_pbx_run: Channel 'Local/8500998@default-8ed8,1' sent into invalid extension 's' in context 'default', but no invalid handler

[striker]

make sure DTMF is working in your carrier by dialling directly from the softphone to a ivr.


also try relaxdtmf=yes in the chand_dahdi.so or zapata.conf.

[Srinivasa Rao]

Thank you striker for your immediate response.

DTMF is working in my carrier, I have tried with soft phone .

I have kept relaxdtmf=yes in zapata.conf, but my problem didn't solved.

[striker]

post your installtion method

[Srinivasa Rao]

Hi Striker,

What type of installation method we have followed i don't know.

But, I know the sequence of steps to install

1. Installed Mysql
2. Installed perl_modules
3. Configured ntp.conf for time synchronization
4. Installed openssh,openssl,ploticus and all
5. Installed Apache
6. Installed PHP
7. Installed libpri,dahdi,Asterisk, Asterisk-addons
8. Installed astGUIClient

[striker]

you should use the asterisk patched for vicidial or pre patched asterisk's

from download.vicidial.com
asterisk-1.4.21.2-vici.tar.gz
asterisk-1.4.27.1-vici.tar.gz
asterisk-1.4.39.1-vici.tar.gz

[williamconley]

Thank you striker for your immediate response.

DTMF is working in my carrier, I have tried with soft phone .

I have kept relaxdtmf=yes in zapata.conf, but my problem didn't solved.

please show your carrier settings used for "I have tried with soft phone" (and show the asterisk cli from that success) and then show it again fora failed attempt. possibly with sip debug on. please be sure there is no other traffic on the system during your attempt as cli output can be long (please don't post 3000 lines of irrelevant code! LOL)

[Srinivasa Rao]

Thank you williamconley

Below i am writing my asterisk log

Success Log

<------------>
-- Executing [9885098850@outbound:1] Dial("SIP/cc100-00000354", "Zap/4/9885098850|30") in new stack
-- Called 4/9885098850
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
[Jan 3 11:52:02] ERROR[20095]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe
== Manager 'sendcron' logged off from 127.0.0.1
-- Zap/4-1 answered SIP/cc100-00000354
Audio is at 10.5.0.28 port 10150
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.5.32.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.32.40:5060;branch=z9hG4bK-d8754z-20218a5b66c94224-1---d8754z-;received=10.5.32.40;rport=5060
From: "cc100"<sip:cc100@10.5.0.28>;transport=UDP;tag=ca010154
To: <sip:9885098850@10.5.0.28>;transport=UDP;tag=as1865ffb8
Call-ID: Nzg0ZjExMmNjMDk4YjViM2E5OWZkZDlmNTgzZmJmOWI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:9885098850@10.5.0.28>
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 2735 2735 IN IP4 10.5.0.28
s=session
c=IN IP4 10.5.0.28
t=0 0
m=audio 10150 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from 10.5.32.40:5060 --->
ACK sip:9885098850@10.5.0.28 SIP/2.0
Via: SIP/2.0/UDP 10.5.32.40:5060;branch=z9hG4bK-d8754z-6dfd3d29830c84ea-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc100@10.5.32.40:5060>
To: <sip:9885098850@10.5.0.28>;transport=UDP;tag=as1865ffb8
From: "cc100"<sip:cc100@10.5.0.28>;transport=UDP;tag=ca010154
Call-ID: Nzg0ZjExMmNjMDk4YjViM2E5OWZkZDlmNTgzZmJmOWI.
CSeq: 2 ACK
Proxy-Authorization: Digest username="cc100",realm="asterisk",nonce="2e654f6e",uri="sip:9885098850@10.5.0.28;transport=UDP",response="82d428a8e082b65352cf5e2f48908667",algorithm=MD5
User-Agent: Zoiper for Windows rev.1105
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

<--- SIP read from 10.5.32.40:5060 --->

<------------->
Really destroying SIP dialog 'MmE1ZDhjYzAyNWY3ZTVlMWM2MDQ5NWY0ZjcyMjQ4NzE.' Method: REGISTER
Really destroying SIP dialog 'MDE4YTc5NjYzZDhjZWM3MzY1ZDg5ZmFmYmY5YzFiYmU.' Method: REGISTER

<--- SIP read from 10.5.32.40:5060 --->
BYE sip:9885098850@10.5.0.28 SIP/2.0
Via: SIP/2.0/UDP 10.5.32.40:5060;branch=z9hG4bK-d8754z-6bb74389790f72c5-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc100@10.5.32.40:5060>
To: <sip:9885098850@10.5.0.28>;transport=UDP;tag=as1865ffb8
From: "cc100"<sip:cc100@10.5.0.28>;transport=UDP;tag=ca010154
Call-ID: Nzg0ZjExMmNjMDk4YjViM2E5OWZkZDlmNTgzZmJmOWI.
CSeq: 3 BYE
Proxy-Authorization: Digest username="cc100",realm="asterisk",nonce="2e654f6e",uri="sip:9885098850@10.5.0.28",response="074c9e439a3928e9d5479236ec16cbc0",algorithm=MD5
User-Agent: Zoiper for Windows rev.1105
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 10.5.32.40 : 5060 (NAT)

<--- Transmitting (NAT) to 10.5.32.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.32.40:5060;branch=z9hG4bK-d8754z-6bb74389790f72c5-1---d8754z-;received=10.5.32.40;rport=5060
From: "cc100"<sip:cc100@10.5.0.28>;transport=UDP;tag=ca010154
To: <sip:9885098850@10.5.0.28>;transport=UDP;tag=as1865ffb8
Call-ID: Nzg0ZjExMmNjMDk4YjViM2E5OWZkZDlmNTgzZmJmOWI.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
-- Hungup 'Zap/4-1'
== Spawn extension (outbound, 9885098850, 1) exited non-zero on 'SIP/cc100-00000354'
Really destroying SIP dialog 'Nzg0ZjExMmNjMDk4YjViM2E5OWZkZDlmNTgzZmJmOWI.' Method: BYE

Failure Log

== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600054@default:1] MeetMe("Local/8600054@default-f445,2", "8600054|q") in new stack
> Channel Local/8600054@default-f445,1 was answered.
-- Executing [919885098850@default:1] AGI("Local/8600054@default-f445,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [919885098850@default:2] Dial("Local/8600054@default-f445,1", "Zap/4/9885098850|55|To") in new stack
-- Called 4/9885098850
Reliably Transmitting (no NAT) to 10.5.32.40:5060:
OPTIONS sip:cc106@10.5.32.40:5060;rinstance=12d53be945c54233 SIP/2.0
Via: SIP/2.0/UDP 10.5.0.28:5060;branch=z9hG4bK232686d1;rport
From: "asterisk" <sip:asterisk@10.5.0.28>;tag=as2e9303f9
To: <sip:cc106@10.5.32.40:5060;rinstance=12d53be945c54233>
Contact: <sip:asterisk@10.5.0.28>
Call-ID: 4a9d16cf29a9f4d753a0733d00e8909b@10.5.0.28
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 03 Jan 2012 06:24:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---

<--- SIP read from 10.5.32.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.0.28:5060;branch=z9hG4bK232686d1;rport=5060
Contact: <sip:10.5.32.40:5060>
To: <sip:cc106@10.5.32.40:5060;rinstance=12d53be945c54233>;tag=c76afd4a
From: "asterisk"<sip:asterisk@10.5.0.28>;tag=as2e9303f9
Call-ID: 4a9d16cf29a9f4d753a0733d00e8909b@10.5.0.28
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
User-Agent: Zoiper for Windows rev.1105
Allow-Events: message-summary, dialog
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '4a9d16cf29a9f4d753a0733d00e8909b@10.5.0.28' Method: OPTIONS
-- Zap/4-1 answered Local/8600054@default-f445,1
== Manager 'sendcron' logged off from 127.0.0.1

<--- SIP read from 10.5.32.40:5060 --->

<------------->
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Really destroying SIP dialog '54aeefc028a971673e5044146468b72b@127.0.0.1' Method: INVITE
[Jan 3 11:54:53] NOTICE[20365]: channel.c:3603 __ast_request_and_dial: Unable to request channel SIP/cc102
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8500998@default:1] Answer("Local/8500998@default-331d,2", "") in new stack
-- Executing [8500998@default:2] NoOp("Local/8500998@default-331d,2", "*********************************") in new stack
-- Executing [8500998@default:3] NoOp("Local/8500998@default-331d,2", "*********************************") in new stack
-- Executing [8500998@default:4] AGI("Local/8500998@default-331d,2", "agi-dtmf.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-dtmf.agi
> Channel Local/8500998@default-331d,1 was answered.
== Starting Local/8500998@default-331d,1 at default,78600054,1 failed so falling back to exten 's'
== Starting Local/8500998@default-331d,1 at default,s,1 still failed so falling back to context 'default'
[Jan 3 11:54:53] WARNING[20371]: pbx.c:2525 __ast_pbx_run: Channel 'Local/8500998@default-331d,1' sent into invalid extension 's' in context 'default', but no invalid handler
-- Executing [h@default:1] DeadAGI("Local/8500998@default-331d,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [h@default:2] DeadAGI("Local/8500998@default-331d,1", "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
== Spawn extension (default, 8500998, 4) exited non-zero on 'Local/8500998@default-331d,2'
-- Executing [h@default:1] DeadAGI("Local/8500998@default-331d,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [h@default:2] DeadAGI("Local/8500998@default-331d,2", "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

[williamconley]

that's only half: i asked for the settings from the two calls ... AND the cli. you just have the cli ... but these calls were made through two different methods, we need to explore how they differ ...

can you describe in detail how each was executed? (ie: button pressed on agent soft phone ... field filled in on logged in agent screen and "send dtmf" pressed ... something to give us an idea of how/why they are different from one another?) I hate fishing for needles in haystacks without a magnet.

also: let's be clear on what is happening here ... is this a manual dial call trying to reach a prospect, or a "dial with prospect" 3rd party call while already with a prospect ...? what is the situation during which you are dtmf'ing?

[Srinivasa Rao]

Hi William,

Success Case :

I have created one context as named outbound in my extensions.conf and directly assigned that context to one soft phone(cc100).

[outbound]
exten => _x.,1,Dial(Zap/4/${EXTEN},30)

Now i have registered the soft phone cc100 and made a call to IVR number without using the vicidial application.Whenever the IVR asked the input, i have given that in my soft phone's text box only then it's moved to next menu successfully.

Failure Case :

Normally in the vicidial application i have used the manual dial option and entered the desired IVR number and click on dial now link then call went perfectly. Whenever the IVR asked the input, i have given that input in send dtmf text box and click on send dtmf button. Then it's taking respective send_dtmf_extension from the phones table and going to extension.conf to the extension 8500998. There its going to agi-dtmf.agi. But i am not getting whether this agi is executing or not.

[williamconley]

you say that it's going to 8500998, but:

-- Executing [8500998@default:4] AGI("Local/8500998@default-331d,2", "agi-dtmf.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-dtmf.agi
> Channel Local/8500998@default-331d,1 was answered.
== Starting Local/8500998@default-331d,1 at default,78600054,1 failed so falling back to exten 's'

were you in meetme room 8600054?

if so ... why would it fail?

[Srinivasa Rao]

Yes, i am in the meet me room 8600054.

[williamconley]

check to see why 78600054 failed then ... i forget what the 7 prefix is used for. but it obviously should not be failing.

do you have these lines:

Code: Select all

extensions.conf:51:exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
extensions.conf:334:exten => _78600XXX,1,Meetme,${EXTEN:1}|Fq
extensions.conf:335:exten => _78600XXX,n,Hangup

[Srinivasa Rao]

Thank you william,

It was very helpful for me..

Now its working fine.

But i also passed an extra parameter to the agi like below

exten => 8500998,n,AGI(agi-dtmf.agi,signalonly)

Now my system accepts dtmf

Thank you very much

[williamconley]

excellent postback.

did you make any other changes?

[Srinivasa Rao]

Ya William,

As i said, i have uncommented the code in extension.conf file which you given above and passed an extra parameter (signalonly) to the agi-dtmf.agi file.

Nothing other than that.


Thank you very much for your great support.

[bghayad]

It look like that using vicidial will require to use AGI in the extension, well: Where I can find a starting documentation to know how to create a right extensions.conf for vicidial?

Regards
Bilal

[scat]

Hi All,

Please help!

How do I configure Vicidial to set up manual dial?

sorry i am new to the software and am really lost right now.can someone redirect me on where to go?

It s that the problem is the dial plan settings that i have which shows:

exten => _52610x,1,AGI (agi://127.0.0.1:4577/call_log)
exten => _52610x.,2,Dial(SIP/${EXEN:4}@auvoip1,,tToR)
exen => _52610X,3,Hangup

what do i put on the plan to connect it with my ibeam soft phone?

Is there any other number that i need to enter?

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

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) Manual dialing calls is covered in the Vicidial Manager's Manual if I recall correctly. Since you are likely to have a panic about each stage of your process setting up your server ... my best advice for you is the same as everyone else who begins to use Vicidial: Download the free version of the manual. Start at page ONE and do NOT skip any pages. Before you reach the end, you'll have a functional server.

4) To get a functional manually dialed call ... we'd need to know if you refer to manually dialing direct from a (soft) phone to the outside world or by pressing "manual dial" in a logged in agent's screen to generate a call.