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bart at icpage.com Guest
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Posted: Sat Jun 21, 2008 11:11 am Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF digits
from SIP is very short sounding or distorted (barely audible) on the ZAP
and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is keeping me
from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses yet.
There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
[7147832205-inn] ROUTING TO: CUST 03 [*7142318000*7147832205*]
-- Executing [7147832205 at call-cust-03:12] Dial("SIP/innov-09a73f78",
"Zap/g5/*7142318000*2205*|10|r") in new stack
[Jun 19 15:26:15] DEBUG[12160]: chan_zap.c:1949 zt_call: Dialing
'*7142318000*2205*'
[Jun 19 15:26:15] DEBUG[12160]: chan_zap.c:2025 zt_call: Deferring
dialing...
-- Called g5/*7142318000*2205*
[Jun 19 15:26:15] DEBUG[12160]: chan_zap.c:4378 zt_handle_event: Ignoring
wink on channel 97
[Jun 19 15:26:16] DEBUG[12160]: chan_zap.c:4441 zt_handle_event: Sent
deferred digit string: T*7142318000*2205
[Jun 19 15:26:19] DEBUG[12160]: chan_zap.c:1452 zt_train_ec: Engaged echo
training on channel 97
[Jun 19 15:26:21] DEBUG[12160]: chan_zap.c:1415 zt_enable_ec: Echo
cancellation already on
-- Zap/97-1 answered SIP/innov-09a73f78
[Jun 19 15:26:30] DTMF[12160]: channel.c:2204 __ast_read: DTMF begin '*'
received on SIP/innov-09a73f78
[Jun 19 15:26:30] DTMF[12160]: channel.c:2215 __ast_read: DTMF begin
passthrough '*' on SIP/innov-09a73f78
[Jun 19 15:26:30] DEBUG[12160]: chan_zap.c:1050 zt_digit_begin: Started
VLDTMF digit '*'
[Jun 19 15:26:30] DTMF[12160]: channel.c:2129 __ast_read: DTMF end '*'
received on SIP/innov-09a73f78, duration 100 ms
[Jun 19 15:26:30] DTMF[12160]: channel.c:2176 __ast_read: DTMF end accepted
with begin '*' on SIP/innov-09a73f78
[Jun 19 15:26:30] DTMF[12160]: channel.c:2192 __ast_read: DTMF end
passthrough '*' on SIP/innov-09a73f78
[Jun 19 15:26:30] DEBUG[12160]: chan_zap.c:1085 zt_digit_end: Ending VLDTMF
digit '*'
I'm using:
Asterisk Source Version : 1.4.21
Zaptel Source Version : 1.4.11
Libpri Source Version : 1.4.4
Addons Source Version : 1.4.7
Please help, I'm stuck on 1.2 until resolved - Thanks
Bart |
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stotaro at totarotechn... Guest
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Posted: Sat Jun 21, 2008 12:03 pm Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF digits
from SIP is very short sounding or distorted (barely audible) on the ZAP
and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is keeping me
from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses yet.
There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
|
Hello, here are a few pointers that might help. Are you using
RFC2833, inband, info? My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.
This is pretty interesting and might solve your issue. It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it. Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248
Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Thanks,
Steve Totaro |
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bart at icpage.com Guest
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Posted: Sat Jun 21, 2008 1:26 pm Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get what's it doing, but
wonder what else would be effected afterwards - I guess I could switch back
if it turns out to be a bad idea
Bart
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF digits
from SIP is very short sounding or distorted (barely audible) on the ZAP
and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is keeping
| me
Quote: | from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses
| yet.
Quote: | There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
|
Hello, here are a few pointers that might help. Are you using
RFC2833, inband, info? My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.
This is pretty interesting and might solve your issue. It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it. Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248
Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Thanks,
Steve Totaro |
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stotaro at totarotechn... Guest
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Posted: Sun Jun 22, 2008 9:35 am Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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Bart,
Did you try the method of inband along with changing the frequencies
at the same time?
Thanks,
Steve T
On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | OK, tried changing DTMF tone as described on URL and no difference
Bart
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get what's it doing, but
wonder what else would be effected afterwards - I guess I could switch back
if it turns out to be a bad idea
Bart
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF digits
from SIP is very short sounding or distorted (barely audible) on the ZAP
and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is keeping
| me
Quote: | from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses
| yet.
Quote: | There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
|
Hello, here are a few pointers that might help. Are you using
RFC2833, inband, info? My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.
This is pretty interesting and might solve your issue. It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it. Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248
Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Thanks,
Steve Totaro
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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stotaro at totarotechn... Guest
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Posted: Sun Jun 22, 2008 9:44 am Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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Also, when you tried inband, did you set it on the phone as well as sip.conf?
Thanks,
Steve T
On Sun, Jun 22, 2008 at 10:35 AM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
Quote: | Bart,
Did you try the method of inband along with changing the frequencies
at the same time?
Thanks,
Steve T
On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | OK, tried changing DTMF tone as described on URL and no difference
Bart
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get what's it doing, but
wonder what else would be effected afterwards - I guess I could switch back
if it turns out to be a bad idea
Bart
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF digits
from SIP is very short sounding or distorted (barely audible) on the ZAP
and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is keeping
| me
Quote: | from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses
| yet.
Quote: | There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
|
Hello, here are a few pointers that might help. Are you using
RFC2833, inband, info? My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.
This is pretty interesting and might solve your issue. It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it. Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248
Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Thanks,
Steve Totaro
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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bart at icpage.com Guest
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Posted: Sun Jun 22, 2008 10:30 am Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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|
Yep - tried both and combination thereof - The bad effect of inband mode was
audio went one way after first press
My test app reads back the ANI & DNIS at answer (which works), then prompts
for more digits.
It's suppose to read back whatever is heard. I can see it reading back
something, back I don't hear anything.
One note: if I press say '1111111' fast, it might hear '11', but not all
digits sadly
I'm sure this is a 'bug' as it work perfectly on 1.2, but so far there is no
acknowledgement from Developers yet.
Not sure how long it should take
Bart
-----Original Message-----
From: Steve Totaro [mailto:stotaro at totarotechnologies.com]
Sent: Sunday, June 22, 2008 7:36 AM
To: bart at icpage.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after
connection when arrives as SIP
Bart,
Did you try the method of inband along with changing the frequencies
at the same time?
Thanks,
Steve T
On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | OK, tried changing DTMF tone as described on URL and no difference
Bart
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get what's it doing,
| but
Quote: | wonder what else would be effected afterwards - I guess I could switch
| back
Quote: | if it turns out to be a bad idea
Bart
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF
|
| digits
Quote: | Quote: | from SIP is very short sounding or distorted (barely audible) on the ZAP
and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is keeping
| me
Quote: | from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses
| yet.
Quote: | There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
|
Hello, here are a few pointers that might help. Are you using
RFC2833, inband, info? My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.
This is pretty interesting and might solve your issue. It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it. Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248
Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Thanks,
Steve Totaro
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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bart at icpage.com Guest
|
Posted: Sun Jun 22, 2008 11:13 am Post subject: [asterisk-users] DTMF not reproduced towards ZAP T1 Port aft |
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Yeah, it gets a bit confusing with all the scenario possible - Regardless,
you are right I should stay on 1.2 until 1.4 is ready for prime time, but
now that 1.6 is out, I'm sure I'm in for a long wait. I reposted my bug
again, since I think I may have listed it wrong - it's now
http://bugs.digium.com/view.php?id=12913 - Maybe now someone might notice
Thanks, Steve for your inputs
Bart
Asterisk has never been good at catching DTMF in rapid succession. I
have read in many places that asterisk 1.4 had many changes to DTMF.
You contradict yourself below. "The bad effect of inband mode was
Quote: | audio went one way after first press" and "One note: if I press say
| '1111111' fast, it might hear '11', but not all digits sadly"
I suppose that you were using different methods. Try pressing the
keys a little slower.
Personally, I would just go back to 1.2.X if you cannot get anyone to
acknowledge your issue. What features do you need in 1.4 anyways?
Maybe if the DTMF bugs you opened get resolved then 1.4.X could be
revisited.
Thanks,
Steve T
On Sun, Jun 22, 2008 at 11:30 AM, Barton Fisher <bart at icpage.com> wrote:
Quote: | Yep - tried both and combination thereof - The bad effect of inband mode
| was
Quote: | audio went one way after first press
My test app reads back the ANI & DNIS at answer (which works), then
| prompts
Quote: | for more digits.
It's suppose to read back whatever is heard. I can see it reading back
something, back I don't hear anything.
One note: if I press say '1111111' fast, it might hear '11', but not all
digits sadly
I'm sure this is a 'bug' as it work perfectly on 1.2, but so far there is
| no
Quote: | acknowledgement from Developers yet.
Not sure how long it should take
Bart
-----Original Message-----
From: Steve Totaro [mailto:stotaro at totarotechnologies.com]
Sent: Sunday, June 22, 2008 7:36 AM
To: bart at icpage.com; Asterisk Users Mailing List - Non-Commercial
| Discussion
Quote: | Subject: Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port
| after
Quote: | connection when arrives as SIP
Bart,
Did you try the method of inband along with changing the frequencies
at the same time?
Thanks,
Steve T
On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | OK, tried changing DTMF tone as described on URL and no difference
Bart
Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push.
As far as changing the code per the URL, I think I get what's it doing,
| but
Quote: | wonder what else would be effected afterwards - I guess I could switch
| back
Quote: | if it turns out to be a bad idea
Bart
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
Quote: | I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
external IVR system. I can hear the asterisk sending the DTMFs properly
toward ZAP at call setup. After the call connects, any further DTMF
|
| digits
Quote: | Quote: | from SIP is very short sounding or distorted (barely audible) on the
|
|
| ZAP
Quote: | Quote: | Quote: | and ignored. ZAP to ZAP connections work perfect.
Just so you know, with 1.2 this is not an issue and this issue is
|
|
| keeping
Quote: | Quote: | me
Quote: | from moving to 1.4.
I have a test system setup with a SIP DID to a test IVR system to
demonstrate this problem. I can provide full access to these systems for
testing. I've placed on Digium bugs but have not received any responses
| yet.
Quote: | There is nothing in the logs or cli that provides anything meaningful -
Below is a call where I press '*' and it is ignored.
|
Hello, here are a few pointers that might help. Are you using
RFC2833, inband, info? My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.
This is pretty interesting and might solve your issue. It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it. Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248
Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF
Thanks,
Steve Totaro
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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