Re: [asterisk-users] DTMF-troubles with Snom

2011-01-08 Thread Jonas Kellens

Hello,

I have tried several settings.

Normally I set it to rfc 2833 on most phone types 
(Grandstream/YeaLink/Cisco SPA). Works always.


With Snom you have the option : SIP info : on/off/always

Neither of these settings make any difference...


What setting do you have in your Snom phones ??

The problem only occurs when calling with a Snom phone...
If you have 1 IVR-menu, then there's no problem.
But when you add an extra layer (a second IVR-menu), then your first 
input is used for the second IVR also. Strange !



Kind regards,
Jonas.


On 01/07/2011 04:36 PM, Bryant Zimmerman wrote:

Jonas

What is the dtmf setting on all peers involved in the call?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003



*From*: "Jonas Kellens" 
*Sent*: Wednesday, January 05, 2011 4:55 PM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion" 


*Subject*: [asterisk-users] DTMF-troubles with Snom

Hello list,

I'm having DTMF-troubles with a Snom phone. I want to know if it's the 
Snom or Asterisk that makes the trouble.



I'm playing a prompt, then make a choice for "2" :

[Jan  5 17:06:38] VERBOSE[29172] file.c: [Jan  5 17:06:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin'

(language 'nl')
[Jan  5 17:06:39] VERBOSE[29172] pbx.c: [Jan  5 17:06:39] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test1-0701", "15") in 
new stack
*[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701*


What follows is a prompt again, and it automatically chooses option 2 :

[Jan  5 17:06:41] VERBOSE[29172] file.c: [Jan  5 17:06:41] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
*[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701*


Even without pressing "2" on the Snom phone, option 2 is chosen in the 
menu.



The above is different when I do the same with a Grandstream device :

[Jan  5 17:14:15] VERBOSE[29384] file.c: [Jan  5 17:14:15] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (

language 'nl')
[Jan  5 17:14:17] VERBOSE[29384] pbx.c: [Jan  5 17:14:17] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in 
new stack
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18] > doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18] > doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
*[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714*


[Jan  5 17:14:38] VERBOSE[29384] file.c: [Jan  5 17:14:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
[Jan  5 17:14:39] VERBOSE[29384] pbx.c: [Jan  5 17:14:39] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in 
new stack
*[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714*



Here I explicitly chose option "2" by pressing on button "2".

What is going on with the Snom ? There is a difference in duration 
(160ms vs 100ms). Is that the problem ??



Kind regards,
Jonas.


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Re: [asterisk-users] DTMF-troubles with Snom

2011-01-08 Thread Bryant Zimmerman
Jonas

What is the dtmf setting on all peers involved in the call?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 


 From: "Jonas Kellens" 
Sent: Wednesday, January 05, 2011 4:55 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] DTMF-troubles with Snom

Hello list,

I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom 
or Asterisk that makes the trouble.

I'm playing a prompt, then make a choice for "2" :

[Jan  5 17:06:38] VERBOSE[29172] file.c: [Jan  5 17:06:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' 
(language 'nl')
[Jan  5 17:06:39] VERBOSE[29172] pbx.c: [Jan  5 17:06:39] -- Executing 
[...@sub-routing:52] WaitExten("SIP/test1-0701", "15") in new stack
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701

What follows is a prompt again, and it automatically chooses option 2 :

[Jan  5 17:06:41] VERBOSE[29172] file.c: [Jan  5 17:06:41] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701

Even without pressing "2" on the Snom phone, option 2 is chosen in the 
menu.

The above is different when I do the same with a Grandstream device :

[Jan  5 17:14:15] VERBOSE[29384] file.c: [Jan  5 17:14:15] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (
language 'nl')
[Jan  5 17:14:17] VERBOSE[29384] pbx.c: [Jan  5 17:14:17] -- Executing 
[...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in new stack
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18]> doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18]> doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714

[Jan  5 17:14:38] VERBOSE[29384] file.c: [Jan  5 17:14:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
[Jan  5 17:14:39] VERBOSE[29384] pbx.c: [Jan  5 17:14:39] -- Executing 
[...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in new stack
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714

Here I explicitly chose option "2" by pressing on button "2".

What is going on with the Snom ? There is a difference in duration (160ms 
vs 100ms). Is that the problem ??

Kind regards,
Jonas.


--
_
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[asterisk-users] dtmf-troubles with Snom

2011-01-06 Thread Jonas Kellens

Hello list,

I'm having DTMF-troubles with a Snom phone. I want to know if it's the 
Snom or Asterisk that makes the trouble.



I'm playing a prompt, then make a choice for "2" :

[Jan  5 17:06:38] VERBOSE[29172] file.c: [Jan  5 17:06:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin'

(language 'nl')
[Jan  5 17:06:39] VERBOSE[29172] pbx.c: [Jan  5 17:06:39] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test1-0701", "15") in 
new stack
*[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701*


What follows is a prompt again, and it automatically chooses option 2 :

[Jan  5 17:06:41] VERBOSE[29172] file.c: [Jan  5 17:06:41] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
*[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701*


Even without pressing "2" on the Snom phone, option 2 is chosen in the menu.


The above is different when I do the same with a Grandstream device :

[Jan  5 17:14:15] VERBOSE[29384] file.c: [Jan  5 17:14:15] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (

language 'nl')
[Jan  5 17:14:17] VERBOSE[29384] pbx.c: [Jan  5 17:14:17] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in 
new stack
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18] > doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18] > doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
*[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714*


[Jan  5 17:14:38] VERBOSE[29384] file.c: [Jan  5 17:14:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
[Jan  5 17:14:39] VERBOSE[29384] pbx.c: [Jan  5 17:14:39] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in 
new stack
*[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714*



Here I explicitly chose option "2" by pressing on button "2".

What is going on with the Snom ? There is a difference in duration 
(160ms vs 100ms). Is that the problem ??



Kind regards,
Jonas.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] DTMF-troubles with Snom

2011-01-05 Thread Jonas Kellens

Hello list,

I'm having DTMF-troubles with a Snom phone. I want to know if it's the 
Snom or Asterisk that makes the trouble.



I'm playing a prompt, then make a choice for "2" :

[Jan  5 17:06:38] VERBOSE[29172] file.c: [Jan  5 17:06:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin'

(language 'nl')
[Jan  5 17:06:39] VERBOSE[29172] pbx.c: [Jan  5 17:06:39] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test1-0701", "15") in 
new stack
*[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin '2' received on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF begin ignored '2' on 
SIP/test1-0701
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701*


What follows is a prompt again, and it automatically chooses option 2 :

[Jan  5 17:06:41] VERBOSE[29172] file.c: [Jan  5 17:06:41] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
*[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end '2' received on 
SIP/test1-0701, duration 160 ms
[Jan  5 17:06:41] DTMF[29172] channel.c: DTMF end passthrough '2' on 
SIP/test1-0701*


Even without pressing "2" on the Snom phone, option 2 is chosen in the menu.


The above is different when I do the same with a Grandstream device :

[Jan  5 17:14:15] VERBOSE[29384] file.c: [Jan  5 17:14:15] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (

language 'nl')
[Jan  5 17:14:17] VERBOSE[29384] pbx.c: [Jan  5 17:14:17] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in 
new stack
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18] > doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
[Jan  5 17:14:18] VERBOSE[4582] dnsmgr.c: [Jan  5 17:14:18] > doing 
dnsmgr_lookup for 'ssw4.brussels.weepee.org'
*[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:21] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714*


[Jan  5 17:14:38] VERBOSE[29384] file.c: [Jan  5 17:14:38] -- 
 Playing 
'/var/lib/asterisk/sounds/vprompts/109001/prompt5041.slin' (language 'nl')
[Jan  5 17:14:39] VERBOSE[29384] pbx.c: [Jan  5 17:14:39] -- 
Executing [...@sub-routing:52] WaitExten("SIP/test6-0714", "15") in 
new stack
*[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin '2' received on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF begin ignored '2' on 
SIP/test6-0714
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end '2' received on 
SIP/test6-0714, duration 100 ms
[Jan  5 17:14:44] DTMF[29384] channel.c: DTMF end passthrough '2' on 
SIP/test6-0714***



Here I explicitly chose option "2" by pressing on button "2".

What is going on with the Snom ? There is a difference in duration 
(160ms vs 100ms). Is that the problem ??



Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users