Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-12 Thread Ivan Stepaniuk
Joseph wrote:
 I just double checked the setting of the remote asterisk and it has the same 
 setting as mine.
 Sip.conf has in Global:
 dtmfmode = rfc2833
 individual extension has no dtmf setting at all, so global setting take 
 precedence.

 All units Linksys, Sipura have 
 DTMF Tx Method: Auto

 Linksys has an additional setting:
 DTMF Tx Mode: Strict

 My asterisk is using old Sipura units and dtmf tones to access voicemail are 
 recognized.
 The remote asterisk is using newer Linksys units and dtmf to voicemail does 
 not work, the phone hangs up.  

 The strange part is:
 PSTN -- Asterisk (voicemail access) works OK on both sytemes.
 Asterisk (w/Linksys) -- Asterisk (w/Sipura) to Voicemail works OK
 Asterisk (w/Linksys) -- Asterisk (w/Linksys) to Voicemail DOES NOT work
 Asterisk (w/Sipura) --  Asterisk (w/Linksys) to Voicemail DOES NOT work
   
You are using the ATAs to access from one Asterisk to the other one? 
Wouldn't make sense to connect those two asterisk through SIP or IAX via 
Internet instead of calling via PSTN anyway?

Anyway, In this case it seems that this is not asterisk related, try the 
several Sipura/Linksys settings  related to DTMF, also if the asterisk 
boxes are the only thing your ATA connects to, there is no point on 
using the auto mode. 

Check the relaxed dtmf setting on yhe linksys, and also check the 
Impedance, Rx and Tx gain, As well as the DTMF duration, All this 
knobs can mess up your DTMF tones if there is something wrong. Just

-- 
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com


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Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-12 Thread Joseph
On 10/12/09 13:29, Ivan Stepaniuk wrote:
You are using the ATAs to access from one Asterisk to the other one?
Wouldn't make sense to connect those two asterisk through SIP or IAX via
Internet instead of calling via PSTN anyway?

Anyway, In this case it seems that this is not asterisk related, try the
several Sipura/Linksys settings  related to DTMF, also if the asterisk
boxes are the only thing your ATA connects to, there is no point on
using the auto mode.

Check the relaxed dtmf setting on yhe linksys, and also check the
Impedance, Rx and Tx gain, As well as the DTMF duration, All this
knobs can mess up your DTMF tones if there is something wrong. Just

--
Iv?n Stepaniuk
Alba Fot?nica S. L.
www.albafotonica.com

Thanks for the input, I'll try play with Linksys units.
The Asterisk on the other end has no Internet connection so I need to use PSTN 
line.
One thing I've noticed on the other end there is a bit echo and TDMF tones 
playing a bit loud when pressing the buttons, it it PSTN to SPA Gain or SPA to 
PSTN
Impedance is set to standard 600 but I'll try 900; but I've read from some 
sources that to find the correct match can be a black art
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080477a06.shtml


-- 
Joseph

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Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-11 Thread Joseph
I just double checked the setting of the remote asterisk and it has the same 
setting as mine.
Sip.conf has in Global:
dtmfmode = rfc2833
individual extension has no dtmf setting at all, so global setting take 
precedence.

All units Linksys, Sipura have 
DTMF Tx Method: Auto

Linksys has an additional setting:
DTMF Tx Mode: Strict

My asterisk is using old Sipura units and dtmf tones to access voicemail are 
recognized.
The remote asterisk is using newer Linksys units and dtmf to voicemail does not 
work, the phone hangs up.  

The strange part is:
PSTN -- Asterisk (voicemail access) works OK on both sytemes.
Asterisk (w/Linksys) -- Asterisk (w/Sipura) to Voicemail works OK
Asterisk (w/Linksys) -- Asterisk (w/Linksys) to Voicemail DOES NOT work
Asterisk (w/Sipura) --  Asterisk (w/Linksys) to Voicemail DOES NOT work

So it seems to me the Linksys units don't work as they suppose to.

--
Joseph

On 10/11/09 01:27, Ivan Stepaniuk wrote:
Joseph wrote:
 I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
 The other Asterisk Linksys is set dtmf = auto
If understand correctly, you have two asterisk servers and when you dial
from one the other, DTMF is not recognized. I also asume you are using
SIP to connect them as you mentioned dtmfmode. In any case, this should
be set to the same value on both sides, both rfc2833, or both info. You
don't wand inband and auto is just rfc2833 with automatic inband fallback.

--
Iv?n Stepaniuk
Alba Fot?nica S. L.
www.albafotonica.com

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[asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-10 Thread Joseph
Asterisk to Asterisk voicemail not working (accessing voicemail from another 
asterisk).
PSTN to Asterisk is working, but not between two asterisk :-(   

I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
The other Asterisk Linksys is set dtmf = auto

-- 
Joseph

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Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-10 Thread Ivan Stepaniuk
Joseph wrote:
 I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
 The other Asterisk Linksys is set dtmf = auto
If understand correctly, you have two asterisk servers and when you dial 
from one the other, DTMF is not recognized. I also asume you are using 
SIP to connect them as you mentioned dtmfmode. In any case, this should 
be set to the same value on both sides, both rfc2833, or both info. You 
don't wand inband and auto is just rfc2833 with automatic inband fallback.

-- 
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com


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Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working

2009-10-10 Thread Joseph
On 10/11/09 01:27, Ivan Stepaniuk wrote:
Joseph wrote:
 I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
 The other Asterisk Linksys is set dtmf = auto
If understand correctly, you have two asterisk servers and when you dial
from one the other, DTMF is not recognized. I also asume you are using
SIP to connect them as you mentioned dtmfmode. In any case, this should
be set to the same value on both sides, both rfc2833, or both info. You
don't wand inband and auto is just rfc2833 with automatic inband fallback.

--
Iv?n Stepaniuk
Alba Fot?nica S. L.
www.albafotonica.com

Thank for the feedback.
You are correct, I have two asterisk servers and dialing from one to another, 
using Linksys (supura unit) and sip.
I'll double check the dtmf on them tomorrow.
As on my Asterisk (have two lines) and dialing from one line to another (it is 
like dialing from one asterisk to another) I dtmf tone is recognized so I can 
access voicemail.

-- 
Joseph

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