DTMF :

[gmcust3]

When I am using DTMF directly using my soft phone like Eyebeam , DTMF works 100% times very smoothly.

Now, when I try it using vicidial , 40% time it connects and other time it doesn't take the dtmf tone.

I tried increasing the dtmf tone lenght to 300 but no change.

I tried changing dtmfmode from rfc2833 to inband but no major change.

My sip :

[Sip]
disallow=all
allow=g729
type=friend
username=username
Fromuser=username
secret=password
host=domain
FromDomain=domain
dtmfmode=inband

Any guidance ?

[williamconley]

dtmf under what circumstances? are you talking about during an agent on the phone with a prospect IN A CONFERENCE? if so, you have to use the dtmf in the agent screen, not in your sip phone.

read the manual (actually, i don't know if it's in there, i just like saying that)

and update to the latest version! you should be ashamed of yourself for using 2.0.5 still, you certainly have the experience to upgrade ... uh-oh, you made changes to the 2.0.5 interface that aren't upgrade-safe, didn't you ?

[gmcust3]

you have to use the dtmf in the agent screen, not in your sip phone.

As a rule , I have to follow this ?

Cant use KeyBoard or Softphone keys ?

you should be ashamed of yourself for using 2.0.5 still, you certainly have the experience to upgrade ... uh-oh, you made changes to the 2.0.5 interface that aren't upgrade-safe, didn't you ?

Honestly, I never faced any issues with current version , thats why never planned to upgrade. Plus, yes, its agent screen also, which is also making me stick to the old verison , { though thats not the priority , using the modified agent screen }

I tried changing few setting in Xlite , but its very erractic. Sometimes It works and sometimes It doesnt.

Apart form Xlite and Eyebeam , any other soft phone which I can try ? If yes, which one you suggest { definitely free , cant help Sorry , Vici made our habbit bad of running behind open source thing }?

[williamconley]

your problem is that your DTMF tone has to go from your soft phone, into a conference room, then get transcoded and sent back out as "sound". This is notoriously unreliable (try conference-calling on your cell phone and seeing if you can get your DTMF tone to pass from one of the people on your conference to the other, it just isn't reliable).

thus the "on-screen" DTMF tone available to agents.

[gmcust3]

Tried from the Agent Portal but Out of 10 times, 7 times happens and 3 times It doesn't ...

[williamconley]

we've found it to be quite reliable. verify you are using the preferred dtmfmode for the carrier in vicidial's carrier setup. or just use them all one at a time until it works.

i know you did this when you were using the soft phone, but that has changed.

[gmcust3]

Any way to track/log the DTMF hitting the asterisk and then what sen to the VOIP and in how much duration or something like it ?

[williamconley]

"the DTMF hitting the asterisk"?

you mean ... have asterisk capture the DTMF from the agent and then pass it to the "Prospect" channel for you reliably?

Since we're all in a meetme room ... you may have some chance at success (as I believe conferences catch DTMF ...), but I'm not sure how much use it will be unless you're prepared to modify the source for the meetme rooms to share data in such a manner.

Otherwise, of course, you'd have to set all the agents that are in the meetme room to "capture" DTMF forever ... this could be expensive for the processor.

[gmcust3]

Now , Here are the details :

Situation : Conference IVR
Number : 7278280956
Room ID: 1175

VOIP :

70.36.100.56
UserName : IVR
Password : IVR1

Sip Settings :

[SipA]
disallow=all
allow=g729
type=friend
username=IVR
Fromuser=IVR
secret=IVR1
host=70.36.100.56
FromDomain=70.36.100.56
dtmfmode=info

SoftPhone : Xlite 3.0 41150 Build

When I try IVR from Agent Interface, It works sometimes and It works doesn't sometimes.

Same is the feedback with the SoftPhone.

I tried a different VOIP but same feedback.

I Tried Vnow 1.3 and Also GoAutoDial but same feedback.

Planning to Re-Install but I will consider this as Silly if it works after Re-Install.

[williamconley]

Try Vicibox from Vicibox.com and see if your results differ (VicidialNOW and GoAutoDial are actually the same thing).

And try debugging with the agi logs and/or asterisk cli with agi debugging turned on. Single Call debugging (test scenario, no other users or even registered phones to reduce noise in the system).

[gmcust3]

And try debugging with the agi logs and/or asterisk cli with agi debugging turned on.

Single Call debugging (test scenario, no other users or even registered phones to reduce noise in the system). : How ? sip debug on ?

I am the Only person Now and can debug and Post the data right Now !!!

[williamconley]

with sip debugging on and with agi debugging on. the method of turning those two on in the cli varies from asterisk version 1.2 to 1.4, but can usually just be "sip debug" or "agi debug". if you posted your asterisk version in your signature, it would be helpful.

you may also need to check the agi logs and the cli logs in /var/log/astguiclient

and remember even unused phones will register constantly. especially if there are a lot of them, they will generate sip traffic all day. so turning them OFF helps.

[gmcust3]

ohhh

So many lines , difficult to trap them and understand them .

DTMF goes as

Signal = 5
Duration=250

If I give a Pause of half a sec , then 80% time DTMF gos through.

Does it mean that I need to enter each letter very slowly ?

[gmcust3]

Below are the Logs of Asterisk , sip debug , when I entered the Room ID and It didnt take.

Code: Select all



<-- SIP read from 99.99.99.999:50452:
ACK sip:9117278280956@888.888.888.888 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.212:50452;branch=z9hG4bK-d87543-365b7e5ba1748528-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:cc112@99.99.99.999:50452>
To: "9117278280956"<sip:9117278280956@888.888.888.888>;tag=as652c93e9
From: "cc112"<sip:cc112@888.888.888.888>;tag=a0369805
Call-ID: M2JkNzdmZTYzODYxYzU1ZGE2ZDQzMTU3ZjYwZDU5NzM.
CSeq: 2 ACK
Proxy-Authorization: Digest username="cc112",realm="asterisk",nonce="4ae24962",uri="sip:9117278280956@888.888.888.888",response="f11e9488b370963c38d2a1c90c21a0b8",algorithm=MD5
User-Agent: X-Lite release 1011s stamp 41150
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
set_destination: Parsing <sip:70.36.100.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 70.36.100.56, port 5060
Reliably Transmitting (NAT) to 70.36.100.56:5060:
INFO sip:70.36.100.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK360a0d53;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:username@888.888.888.888>
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
CSeq: 104 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc112" <sip:cc112@70.36.100.56>;privacy=off;screen=no
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=250

---
vici*CLI>
<-- SIP read from 70.36.100.56:5060:
SIP/2.0 200 OK
CSeq: 104 INFO
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK360a0d53;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:70.36.100.56:5060;transport=udp>
Content-Length: 0


--- (8 headers 0 lines) ---
set_destination: Parsing <sip:70.36.100.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 70.36.100.56, port 5060
Reliably Transmitting (NAT) to 70.36.100.56:5060:
INFO sip:70.36.100.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK25f207dc;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:username@888.888.888.888>
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
CSeq: 105 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc112" <sip:cc112@70.36.100.56>;privacy=off;screen=no
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=250

---
vici*CLI>
<-- SIP read from 70.36.100.56:5060:
SIP/2.0 200 OK
CSeq: 105 INFO
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK25f207dc;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:70.36.100.56:5060;transport=udp>
Content-Length: 0


--- (8 headers 0 lines) ---
set_destination: Parsing <sip:70.36.100.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 70.36.100.56, port 5060
Reliably Transmitting (NAT) to 70.36.100.56:5060:
INFO sip:70.36.100.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK7e1be5a7;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:username@888.888.888.888>
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
CSeq: 106 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc112" <sip:cc112@70.36.100.56>;privacy=off;screen=no
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=7
Duration=250

---
vici*CLI>
<-- SIP read from 70.36.100.56:5060:
SIP/2.0 200 OK
CSeq: 106 INFO
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK7e1be5a7;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:70.36.100.56:5060;transport=udp>
Content-Length: 0


--- (8 headers 0 lines) ---
set_destination: Parsing <sip:70.36.100.56:5060;transport=udp> for address/port to send to
set_destination: set destination to 70.36.100.56, port 5060
Reliably Transmitting (NAT) to 70.36.100.56:5060:
INFO sip:70.36.100.56:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK1137434c;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:username@888.888.888.888>
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
CSeq: 107 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc112" <sip:cc112@70.36.100.56>;privacy=off;screen=no
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=250

---
vici*CLI>
<-- SIP read from 70.36.100.56:5060:
SIP/2.0 200 OK
CSeq: 107 INFO
Via: SIP/2.0/UDP 888.888.888.888:5060;branch=z9hG4bK1137434c;rport
From: "cc112" <sip:username@70.36.100.56>;tag=as2b6e7e6c
Call-ID: 01eeed680b0db37171d30f3e3d3c6a8c@70.36.100.56
To: <sip:17278280956@70.36.100.56>;tag=170129110805482208289069
Contact: <sip:70.36.100.56:5060;transport=udp>
Content-Length: 0


--- (8 headers 0 lines) ---
  == Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>
<-- SIP read from 192.168.100.151:35654:
SUBSCRIBE sip:cc111@192.168.100.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:35654;branch=z9hG4bK-d87543-7d6ee27a627da816-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:cc111@192.168.100.151:35654>
To: "cc111"<sip:cc111@192.168.100.100>
From: "cc111"<sip:cc111@192.168.100.100>;tag=de67a92d
Call-ID: MWI5ZDBkYTkyMjg2NTlhOWVmMWUxNTcxOTc2NDQwODQ.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Event: message-summary
Content-Length: 0


--- (13 headers 0 lines) ---
Using latest SUBSCRIBE request as basis request
Sending to 192.168.100.151 : 35654 (NAT)
Transmitting (NAT) to 192.168.100.151:35654:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.151:35654;branch=z9hG4bK-d87543-7d6ee27a627da816-1--d87543-;received=192.168.100.151;rport=35654
From: "cc111"<sip:cc111@192.168.100.100>;tag=de67a92d
To: "cc111"<sip:cc111@192.168.100.100>;tag=as3b28dd78
Call-ID: MWI5ZDBkYTkyMjg2NTlhOWVmMWUxNTcxOTc2NDQwODQ.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a85c3e7"
Content-Length: 0


---
Scheduling destruction of call 'MWI5ZDBkYTkyMjg2NTlhOWVmMWUxNTcxOTc2NDQwODQ.' in 15000 ms
Found user 'cc111'
vici*CLI>
<-- SIP read from 192.168.100.151:35654:
SUBSCRIBE sip:cc111@192.168.100.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:35654;branch=z9hG4bK-d87543-800229182c4d2548-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:cc111@192.168.100.151:35654>
To: "cc111"<sip:cc111@192.168.100.100>
From: "cc111"<sip:cc111@192.168.100.100>;tag=de67a92d
Call-ID: MWI5ZDBkYTkyMjg2NTlhOWVmMWUxNTcxOTc2NDQwODQ.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="cc111",realm="asterisk",nonce="0a85c3e7",uri="sip:cc111@192.168.100.100",response="8c5706370b1233ba1733a335746b2830",algorithm=MD5
Event: message-summary
Content-Length: 0


--- (14 headers 0 lines) ---
Found user 'cc111'
Looking for cc111 in default (domain 192.168.100.100)
Transmitting (NAT) to 192.168.100.151:35654:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.100.151:35654;branch=z9hG4bK-d87543-800229182c4d2548-1--d87543-;received=192.168.100.151;rport=35654
From: "cc111"<sip:cc111@192.168.100.100>;tag=de67a92d
To: "cc111"<sip:cc111@192.168.100.100>;tag=as3b28dd78
Call-ID: MWI5ZDBkYTkyMjg2NTlhOWVmMWUxNTcxOTc2NDQwODQ.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call 'MWI5ZDBkYTkyMjg2NTlhOWVmMWUxNTcxOTc2NDQwODQ.'
vici*CLI>
<-- SIP read from 99.99.99.999:50452:



--- (0 headers 1 lines) ---
vici*CLI>
<-- SIP read from 192.168.100.151:35654:



--- (0 headers 1 lines) ---
Destroying call '66d4eea13264653804d9854941147158@127.0.0.1'
vici*CLI>



[gmcust3]

Is this Log of any use ? (

[williamconley]

only if you can correlate it to "worked" and "didn't work" and see some sort of difference.

also helpful to know if all requests were, indeed, received by asterisk and acted upon (so you know that the agent interface is doing ITS job and the problem occurs AFTER asterisk). If that's true, you tell me?

try again with all the available dtmf methods between asterisk (always getting the signal) and your carrier (80%).

[gmcust3]

Thats really doesnt look like an easy task.

I have asked my client to use eyebeam and configured directly with VOIP and then do the IVR.

But in that case, they have to hang up and call back the customer again and 70% customer goes on AM. But atleast 30% time its a sale.

[williamconley]

changing the dtmfmode between asterisk and the carrier is a single line entry. then testing takes a few minutes. but there are not very many possible modes (sip header, rfc2833, inline ...?)