Barton Fisher
2008-Jun-21 16:11 UTC
[asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
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
Steve Totaro
2008-Jun-21 17:03 UTC
[asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:> 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
Barton Fisher
2008-Jun-22 16:13 UTC
[asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP
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> 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:> Yep - tried both and combination thereof - The bad effect of inband modewas> audio went one way after first press > My test app reads back the ANI & DNIS at answer (which works), thenprompts> 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 isno> 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-CommercialDiscussion> Subject: Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Portafter> 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: >> 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: >>> 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 theZAP>>> and ignored. ZAP to ZAP connections work perfect. >>> >>> Just so you know, with 1.2 this is not an issue and this issue iskeeping>> 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 >> >> >> >> >> >> _______________________________________________ >> -- 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 >> > > > > >