DTMF issue on playback command with openzap
I did run some test to reproduce the problem. I'm using Version 1.0.pre4 (8797:8820M). Test 1 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the record command is active. The call is originate through openzap. Event_Socket trace: http://pastebin.freeswitch.org/4677 Result: There's no problem. The DTMF handle correctly. I'm receiving all the DTMF in the right order without any duplication. Test 2 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the playback command is active. The call is originate through openzap. Event_Socket trace: http://pastebin.freeswitch.org/4678 Result: There's a problem. The DTMF don't seem to be handles correctly. I'm not receiving the DTMF in the right order, because there's some duplication in the DTMF and I even miss some DTMF. Test 3 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the record command is active. The call is originate through SIP. Event_Socket trace: http://pastebin.freeswitch.org/4679 Result: There's no problem. The DTMF handle correctly. I'm receiving all the DTMF in the right order without any duplication. Test 4 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the playback command is active. The call is originate through SIP. Event_Socket trace: http://pastebin.freeswitch.org/4680 Result: There's no problem. The DTMF handle correctly. I'm receiving all the DTMF in the right order without any duplication. SL
What is on the other end of this test interpreting the DTMF for you to verify its bad? /b On Jun 23, 2008, at 12:52 PM, Steve Laroche wrote:
Test 2 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the playback command is active. The call is originate through openzap.
Event_Socket trace: http://pastebin.freeswitch.org/4678
Result: There’s a problem. The DTMF don’t seem to be handles correctly. I’m not receiving the DTMF in the right order, because there’s some duplication in the DTMF and I even miss some DTMF.
Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap. Steve Laroche, ing. jr Architecte / Analyste principal / Tech Lead IP5 Corporation 507, Place d'Armes Bureau 510 Montréal, Québec H2Y 2W8 T: 514.935.9300 [557] F: 514.788.4291 slaroche@ip5.com<mailto:slaroche@ip5.com> P Est-ce vraiment nécessaire d'imprimer ce courriel ? Is it really necessary to print this e-mail? From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 23, 2008 2:04 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap What is on the other end of this test interpreting the DTMF for you to verify its bad? /b On Jun 23, 2008, at 12:52 PM, Steve Laroche wrote: Test 2 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the playback command is active. The call is originate through openzap. Event_Socket trace: http://pastebin.freeswitch.org/4678 Result: There's a problem. The DTMF don't seem to be handles correctly. I'm not receiving the DTMF in the right order, because there's some duplication in the DTMF and I even miss some DTMF.
You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:
Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
I originate an outbound call to my land line phone. After that, the script execute a playback using G.711. While executing the playback, I enter DTMF on my land line phone. From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 23, 2008 2:20 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote: Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
please describe the entire path of the call and what protocol is involved on the example that does not work. starting with the telephone that placed the call all the way to the telephone or ivr that revived it. On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org> wrote:
You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b
On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:
Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, do you need more info ? From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, June 23, 2008 2:33 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap please describe the entire path of the call and what protocol is involved on the example that does not work. starting with the telephone that placed the call all the way to the telephone or ivr that revived it. On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org<mailto:brian@freeswitch.org>> wrote: You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote: Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org<mailto:Freeswitch-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I understand your problem so you need not repeat anything you have already posted. please describe in detail exactly what you are doing and how when you say openzap specify if it's analog or digital etc I still cannot tell how you are using openzap in this scheme are you saying you are doing something like: originate openzap/1/a/12121231234 1000 to call your pots phone and xfer to 1000 where an ivr is collecting dtmf? On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <slaroche@ip5.com> wrote:
Anthony, do you need more info ?
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, June 23, 2008 2:33 PM *To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] DTMF issue on playback command with openzap
please describe the entire path of the call and what protocol is involved on the example that does not work.
starting with the telephone that placed the call all the way to the telephone or ivr that revived it.
On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org> wrote:
You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on.
/b
On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:
Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I'm driving the call with an IVR outside of the freeswitch using event_socket. I'm using digital lines. In fact, they are T1 with DMS100 protocol connected to a sangoma card. The driver I'm using for the sangoma card is wanpipe-3.2.5. This card have echo canceller activated on. IVR originate a call through event_socket : bgapi originate openzap/1/a/5147884250 &park() The call is originated to a PSTN phone. When IVR receive the events from event_socket confirming the call is connected, IVR send a playback command : SendMsg uuid call-command: execute execute-app-name: playback execute-app-arg: /usr/local/freeswitch/sounds/mission_vm.wav Audio file is about 25sec long, so when the playback is executed on line, I enter DTMF on my PSTN phone. I'm collecting the DTMF events with my system that drives the event_socket. It's basically my test. Thanks. SL From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, June 24, 2008 6:39 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap I understand your problem so you need not repeat anything you have already posted. please describe in detail exactly what you are doing and how when you say openzap specify if it's analog or digital etc I still cannot tell how you are using openzap in this scheme are you saying you are doing something like: originate openzap/1/a/12121231234 1000 to call your pots phone and xfer to 1000 where an ivr is collecting dtmf? On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <slaroche@ip5.com<mailto:slaroche@ip5.com>> wrote: Anthony, do you need more info ? From: freeswitch-dev-bounces@lists.freeswitch.org<mailto:freeswitch-dev-bounces@lists.freeswitch.org> [mailto:freeswitch-dev-bounces@lists.freeswitch.org<mailto:freeswitch-dev-bounces@lists.freeswitch.org>] On Behalf Of Anthony Minessale Sent: Monday, June 23, 2008 2:33 PM To: freeswitch-dev@lists.freeswitch.org<mailto:freeswitch-dev@lists.freeswitch.org> Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap please describe the entire path of the call and what protocol is involved on the example that does not work. starting with the telephone that placed the call all the way to the telephone or ivr that revived it. On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org<mailto:brian@freeswitch.org>> wrote: You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote: Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org<mailto:Freeswitch-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org<mailto:Freeswitch-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
just for fun try it with the echo canceler off. I know for a fact that sangoma's echo canceler messes up the soft dtmf. see the power of details? ;) On Tue, Jun 24, 2008 at 6:43 PM, Steve Laroche <slaroche@ip5.com> wrote:
I'm driving the call with an IVR outside of the freeswitch using event_socket. I'm using
digital lines. In fact, they are T1 with DMS100 protocol connected to a sangoma card.
The driver I'm using for the sangoma card is wanpipe-3.2.5. This card have echo
canceller activated on.
IVR originate a call through event_socket :
bgapi originate openzap/1/a/5147884250 &park()
The call is originated to a PSTN phone.
When IVR receive the events from event_socket confirming the call is connected, IVR
send a playback command :
SendMsg uuid
call-command: execute
execute-app-name: playback
execute-app-arg: /usr/local/freeswitch/sounds/mission_vm.wav
Audio file is about 25sec long, so when the playback is executed on line, I enter DTMF
on my PSTN phone.
I'm collecting the DTMF events with my system that drives the event_socket. It's
basically my test.
Thanks.
SL
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Tuesday, June 24, 2008 6:39 PM
*To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] DTMF issue on playback command with openzap
I understand your problem so you need not repeat anything you have already posted.
please describe in detail exactly what you are doing and how when you say openzap specify if it's analog or digital etc
I still cannot tell how you are using openzap in this scheme
are you saying you are doing something like:
originate openzap/1/a/12121231234 1000
to call your pots phone and xfer to 1000
where an ivr is collecting dtmf?
On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <slaroche@ip5.com> wrote:
Anthony, do you need more info ?
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, June 23, 2008 2:33 PM
*To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] DTMF issue on playback command with openzap
please describe the entire path of the call and what protocol is involved on the example that does not work.
starting with the telephone that placed the call all the way to the telephone or ivr that revived it.
On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org> wrote:
You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on.
/b
On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:
Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, Yes, it does work with echo canceller off. But without echo canceller, when we bridge 2 calls together, we have echo on the line for the voice and the DTMF. For the DTMF, its crazy, one DTMF can loop forever. I'm genna contact sangoma to do a follow up on that, but do you have any T1 card provider that works without any problem that you guys work with? Thanks. SL From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, June 25, 2008 10:30 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap just for fun try it with the echo canceler off. I know for a fact that sangoma's echo canceler messes up the soft dtmf. see the power of details? ;) On Tue, Jun 24, 2008 at 6:43 PM, Steve Laroche <slaroche@ip5.com<mailto:slaroche@ip5.com>> wrote: I'm driving the call with an IVR outside of the freeswitch using event_socket. I'm using digital lines. In fact, they are T1 with DMS100 protocol connected to a sangoma card. The driver I'm using for the sangoma card is wanpipe-3.2.5. This card have echo canceller activated on. IVR originate a call through event_socket : bgapi originate openzap/1/a/5147884250 &park() The call is originated to a PSTN phone. When IVR receive the events from event_socket confirming the call is connected, IVR send a playback command : SendMsg uuid call-command: execute execute-app-name: playback execute-app-arg: /usr/local/freeswitch/sounds/mission_vm.wav Audio file is about 25sec long, so when the playback is executed on line, I enter DTMF on my PSTN phone. I'm collecting the DTMF events with my system that drives the event_socket. It's basically my test. Thanks. SL From: freeswitch-dev-bounces@lists.freeswitch.org<mailto:freeswitch-dev-bounces@lists.freeswitch.org> [mailto:freeswitch-dev-bounces@lists.freeswitch.org<mailto:freeswitch-dev-bounces@lists.freeswitch.org>] On Behalf Of Anthony Minessale Sent: Tuesday, June 24, 2008 6:39 PM To: freeswitch-dev@lists.freeswitch.org<mailto:freeswitch-dev@lists.freeswitch.org> Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap I understand your problem so you need not repeat anything you have already posted. please describe in detail exactly what you are doing and how when you say openzap specify if it's analog or digital etc I still cannot tell how you are using openzap in this scheme are you saying you are doing something like: originate openzap/1/a/12121231234 1000 to call your pots phone and xfer to 1000 where an ivr is collecting dtmf? On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <slaroche@ip5.com<mailto:slaroche@ip5.com>> wrote: Anthony, do you need more info ? From: freeswitch-dev-bounces@lists.freeswitch.org<mailto:freeswitch-dev-bounces@lists.freeswitch.org> [mailto:freeswitch-dev-bounces@lists.freeswitch.org<mailto:freeswitch-dev-bounces@lists.freeswitch.org>] On Behalf Of Anthony Minessale Sent: Monday, June 23, 2008 2:33 PM To: freeswitch-dev@lists.freeswitch.org<mailto:freeswitch-dev@lists.freeswitch.org> Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap please describe the entire path of the call and what protocol is involved on the example that does not work. starting with the telephone that placed the call all the way to the telephone or ivr that revived it. On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org<mailto:brian@freeswitch.org>> wrote: You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote: Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org<mailto:Freeswitch-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org<mailto:Freeswitch-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org<mailto:Freeswitch-dev@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I'm not sure but I think PIKA cards in openzap use hardware dtmf to avoid this issue, maybe we can figure out how to do the same thing with Sangoma. On Wed, Jun 25, 2008 at 10:40 AM, Steve Laroche <slaroche@ip5.com> wrote:
Anthony,
Yes, it does work with echo canceller off. But without echo canceller, when we bridge 2 calls together, we have echo on the line for the voice and
the DTMF. For the DTMF, its crazy, one DTMF can loop forever. I'm genna contact sangoma to do a follow up on that, but do you have any T1
card provider that works without any problem that you guys work with?
Thanks.
SL
* *
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Wednesday, June 25, 2008 10:30 AM
*To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] DTMF issue on playback command with openzap
just for fun try it with the echo canceler off. I know for a fact that sangoma's echo canceler messes up the soft dtmf.
see the power of details? ;)
On Tue, Jun 24, 2008 at 6:43 PM, Steve Laroche <slaroche@ip5.com> wrote:
I'm driving the call with an IVR outside of the freeswitch using event_socket. I'm using
digital lines. In fact, they are T1 with DMS100 protocol connected to a sangoma card.
The driver I'm using for the sangoma card is wanpipe-3.2.5. This card have echo
canceller activated on.
IVR originate a call through event_socket :
bgapi originate openzap/1/a/5147884250 &park()
The call is originated to a PSTN phone.
When IVR receive the events from event_socket confirming the call is connected, IVR
send a playback command :
SendMsg uuid
call-command: execute
execute-app-name: playback
execute-app-arg: /usr/local/freeswitch/sounds/mission_vm.wav
Audio file is about 25sec long, so when the playback is executed on line, I enter DTMF
on my PSTN phone.
I'm collecting the DTMF events with my system that drives the event_socket. It's
basically my test.
Thanks.
SL
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Tuesday, June 24, 2008 6:39 PM
*To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] DTMF issue on playback command with openzap
I understand your problem so you need not repeat anything you have already posted.
please describe in detail exactly what you are doing and how when you say openzap specify if it's analog or digital etc
I still cannot tell how you are using openzap in this scheme
are you saying you are doing something like:
originate openzap/1/a/12121231234 1000
to call your pots phone and xfer to 1000
where an ivr is collecting dtmf?
On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <slaroche@ip5.com> wrote:
Anthony, do you need more info ?
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, June 23, 2008 2:33 PM
*To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] DTMF issue on playback command with openzap
please describe the entire path of the call and what protocol is involved on the example that does not work.
starting with the telephone that placed the call all the way to the telephone or ivr that revived it.
On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org> wrote:
You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on.
/b
On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:
Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
_______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hi, Anthony Minessale wrote:
just for fun try it with the echo canceler off. I know for a fact that sangoma's echo canceler messes up the soft dtmf.
see the power of details? ;) If that is the case, I'd love to see some captured audio that will not decode.
Regards, Steve
FS ( OpenZap )-----------originate call -----------------> land line FS ( OpenZap )-----------playback----------------------> land line FS ( OpenZap )<-------------DTMF----------------------- land line From: Steve Laroche Sent: Monday, June 23, 2008 2:30 PM To: 'freeswitch-dev@lists.freeswitch.org' Subject: RE: [Freeswitch-dev] DTMF issue on playback command with openzap I originate an outbound call to my land line phone. After that, the script execute a playback using G.711. While executing the playback, I enter DTMF on my land line phone. From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 23, 2008 2:20 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote: Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
It's with the events received by FS that I know I don't receive all the correct DTMF. See the trace : http://pastebin.freeswitch.org/4678 So while FS play an audio file on the line at the other end of the call is my phone witch is connected to PSTN without any FS involded, I enter DTMF on my phone. The DTMF sequence I enter is : 12345678901234567891234567890*. Playback stop at the * and after, the IVR send hangup. From: Steve Laroche Sent: Monday, June 23, 2008 2:33 PM To: 'freeswitch-dev@lists.freeswitch.org' Subject: RE: [Freeswitch-dev] DTMF issue on playback command with openzap FS ( OpenZap )-----------originate call -----------------> land line FS ( OpenZap )-----------playback----------------------> land line FS ( OpenZap )<-------------DTMF----------------------- land line From: Steve Laroche Sent: Monday, June 23, 2008 2:30 PM To: 'freeswitch-dev@lists.freeswitch.org' Subject: RE: [Freeswitch-dev] DTMF issue on playback command with openzap I originate an outbound call to my land line phone. After that, the script execute a playback using G.711. While executing the playback, I enter DTMF on my land line phone. From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, June 23, 2008 2:20 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] DTMF issue on playback command with openzap You still didn't answer what was receiving the audio. If its OpenZAP that means its inband. But can you elaborate more on the path? If the audio has ANY gaps in it the DTMF will be double detected but I need to know more about that before I can tell what is going on. /b On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote: Instead of 123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*. While for all the other test that I did I do receive 123456789012345678901234567890*. I did this test many time and always the same result, I have an issue with DTMF when playback is running and the call is originate through openzap.
participants (4)
-
Anthony Minessale
-
Brian West
-
Steve Laroche
-
Steve Underwood