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[Freeswitch-users] DTMF issues/question


 
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marc at avvatel.com
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PostPosted: Thu Sep 11, 2008 5:00 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco 7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband DTMF.

Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

- Marc

--
Marc Lewis
Avvatel Corporation


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anthony.minessale at g...
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PostPosted: Thu Sep 11, 2008 5:06 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

2833 is the default.
INBAND is not a function of the sip module it's implemented on top of the abstraction layer.

Are you doing trans coding or different ptimes between the 2 legs?

you might want to turn up the debug log "press f8" and you will pretty verbose logging of the 2833.
also consider taking a pcap of the network traffic.

What is the provider? 2833 is notoriously done wrong to the point that doing it exactly right can put you at a disadvantage.





On Thu, Sep 11, 2008 at 4:58 PM, Marc Lewis <marc@avvatel.com (marc@avvatel.com)> wrote:
Quote:

I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco 7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband DTMF.

Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

- Marc

--
Marc Lewis
Avvatel Corporation


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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brian at freeswitch.org
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PostPosted: Thu Sep 11, 2008 5:12 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

On Sep 11, 2008, at 4:58 PM, Marc Lewis wrote:

Quote:

I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried
this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco
7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband
DTMF.

No that isn't the case. We don't do anything inband by default.
RFC2833 is our preferred method.

Quote:

Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

Are you using the latest SVN or what code are you using?

Quote:


- Marc

--
Marc Lewis
Avvatel Corporation


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Brian West
sip:brian@freeswitch.org







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marc at avvatel.com
Guest





PostPosted: Thu Sep 11, 2008 5:37 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

I've got logging turned up, and it appears that FS is doing the right thing, but PointOne (my termination provider) isn't honoring it.

I am doing transcoding on some calls (GSM or G726-32 on the phone, PCMU always out to PointOne).

If you want to test, I would be happy to add your IP to my proxy's trusted IP list for a time so you can route calls out through it. Contact me off-list and I can get you (or Brian) set up on it.


This was testing with dtmf-type set to rfc2388 for both my internal and external sofia profiles. Phones register on internal and calls go out through external. Here are the logs from a couple of digits:

2008-09-11 14:39:48 [DEBUG] switch_core_io.c:734 switch_core_session_write_frame() Engaging Write Buffer at 320 bytes to accomodate 640->320
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [3] ts=2394311623 dur=160/160/2000 seq=42444
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=320/320/2000 seq=42445
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=480/480/2000 seq=42446
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=640/640/2000 seq=42447
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=800/800/2000 seq=42448
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=960/960/2000 seq=42449
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1120/1120/2000 seq=42450
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1280/1280/2000 seq=42451
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1440/1440/2000 seq=42452
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1600/1600/2000 seq=42453
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1760/1760/2000 seq=42454
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1920/1920/2000 seq=42455
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42456
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42457
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42458
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [7] ts=2394314503 dur=160/160/2000 seq=42465
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=320/320/2000 seq=42466
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=480/480/2000 seq=42467
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=640/640/2000 seq=42468
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=800/800/2000 seq=42469
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=960/960/2000 seq=42470
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1120/1120/2000 seq=42471
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1280/1280/2000 seq=42472
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1440/1440/2000 seq=42473
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1600/1600/2000 seq=42474
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1760/1760/2000 seq=42475
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1920/1920/2000 seq=42476
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42477
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42478
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42479

- Marc


Anthony Minessale wrote:
Quote:
2833 is the default.
INBAND is not a function of the sip module it's implemented on top of the abstraction layer.

Are you doing trans coding or different ptimes between the 2 legs?

you might want to turn up the debug log "press f8" and you will pretty verbose logging of the 2833.
also consider taking a pcap of the network traffic.

What is the provider? 2833 is notoriously done wrong to the point that doing it exactly right can put you at a disadvantage.





On Thu, Sep 11, 2008 at 4:58 PM, Marc Lewis <marc@avvatel.com (marc@avvatel.com)> wrote:
Quote:

I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco 7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband DTMF.

Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

- Marc

--
Marc Lewis
Avvatel Corporation


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

Quote:


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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Marc Lewis
Avvatel Corporation
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marc at avvatel.com
Guest





PostPosted: Thu Sep 11, 2008 5:38 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

Brian West wrote:
Quote:
Quote:
On Sep 11, 2008, at 4:58 PM, Marc Lewis wrote:

Quote:
I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried
this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco
7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband
DTMF.

No that isn't the case. We don't do anything inband by default.
RFC2833 is our preferred method.

Good information to know.

Quote:
Quote:
Quote:
Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

Are you using the latest SVN or what code are you using?

This is running on svn version 9372.

- Marc

Quote:
--
Marc Lewis
Avvatel Corporation
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brian at freeswitch.org
Guest





PostPosted: Thu Sep 11, 2008 5:49 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

Is this inbound or outbound ?

PSTN -> YOU


YOU -> PSTN


Which way and what makes you think you're dtmf isn't working. I just want to make sure we are on the same page.




/b

On Sep 11, 2008, at 5:34 PM, Marc Lewis wrote:
Quote:
I've got logging turned up, and it appears that FS is doing the right thing, but PointOne (my termination provider) isn't honoring it.

I am doing transcoding on some calls (GSM or G726-32 on the phone, PCMU always out to PointOne).

If you want to test, I would be happy to add your IP to my proxy's trusted IP list for a time so you can route calls out through it. Contact me off-list and I can get you (or Brian) set up on it.


This was testing with dtmf-type set to rfc2388 for both my internal and external sofia profiles. Phones register on internal and calls go out through external. Here are the logs from a couple of digits:

2008-09-11 14:39:48 [DEBUG] switch_core_io.c:734 switch_core_session_write_frame() Engaging Write Buffer at 320 bytes to accomodate 640->320
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [3] ts=2394311623 dur=160/160/2000 seq=42444
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=320/320/2000 seq=42445
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=480/480/2000 seq=42446
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=640/640/2000 seq=42447
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=800/800/2000 seq=42448
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=960/960/2000 seq=42449
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1120/1120/2000 seq=42450
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1280/1280/2000 seq=42451
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1440/1440/2000 seq=42452
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1600/1600/2000 seq=42453
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1760/1760/2000 seq=42454
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1920/1920/2000 seq=42455
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42456
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42457
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42458
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [7] ts=2394314503 dur=160/160/2000 seq=42465
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=320/320/2000 seq=42466
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=480/480/2000 seq=42467
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=640/640/2000 seq=42468
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=800/800/2000 seq=42469
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=960/960/2000 seq=42470
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1120/1120/2000 seq=42471
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1280/1280/2000 seq=42472
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1440/1440/2000 seq=42473
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1600/1600/2000 seq=42474
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1760/1760/2000 seq=42475
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1920/1920/2000 seq=42476
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42477
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42478
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42479

- Marc


Anthony Minessale wrote:
Quote:
2833 is the default.
INBAND is not a function of the sip module it's implemented on top of the abstraction layer.

Are you doing trans coding or different ptimes between the 2 legs?

you might want to turn up the debug log "press f8" and you will pretty verbose logging of the 2833.
also consider taking a pcap of the network traffic.

What is the provider? 2833 is notoriously done wrong to the point that doing it exactly right can put you at a disadvantage.





On Thu, Sep 11, 2008 at 4:58 PM, Marc Lewis <marc@avvatel.com (marc@avvatel.com)> wrote:
Quote:

I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco 7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband DTMF.

Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

- Marc

--
Marc Lewis
Avvatel Corporation


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
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Quote:

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PostPosted: Thu Sep 11, 2008 5:54 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

This is outbound, the diagram would look something like:

phone -> sofia profile internal -> sofia profile external -> my sip proxy (openser for routing)-> term provider

Inbound from PSTN works fine. All DTMF digits are recognized in voicemail, IVR's, etc. Phones also have no problems with DTMF for their voicemail, etc.

The problems arise when trying to send to other IVR's. My last tests have been to call my two different banks and american express. None of the three of them recognized the DTMF tones.

- Marc

Brian West wrote:
Quote:
Is this inbound or outbound ?

PSTN -> YOU


YOU -> PSTN


Which way and what makes you think you're dtmf isn't working. I just want to make sure we are on the same page.




/b

On Sep 11, 2008, at 5:34 PM, Marc Lewis wrote:
Quote:
I've got logging turned up, and it appears that FS is doing the right thing, but PointOne (my termination provider) isn't honoring it.

I am doing transcoding on some calls (GSM or G726-32 on the phone, PCMU always out to PointOne).

If you want to test, I would be happy to add your IP to my proxy's trusted IP list for a time so you can route calls out through it. Contact me off-list and I can get you (or Brian) set up on it.


This was testing with dtmf-type set to rfc2388 for both my internal and external sofia profiles. Phones register on internal and calls go out through external. Here are the logs from a couple of digits:

2008-09-11 14:39:48 [DEBUG] switch_core_io.c:734 switch_core_session_write_frame() Engaging Write Buffer at 320 bytes to accomodate 640->320
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [3] ts=2394311623 dur=160/160/2000 seq=42444
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=320/320/2000 seq=42445
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=480/480/2000 seq=42446
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=640/640/2000 seq=42447
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=800/800/2000 seq=42448
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=960/960/2000 seq=42449
2008-09-11 14:39:56 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1120/1120/2000 seq=42450
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1280/1280/2000 seq=42451
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1440/1440/2000 seq=42452
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1600/1600/2000 seq=42453
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1760/1760/2000 seq=42454
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [3] ts=2394311623 dur=1920/1920/2000 seq=42455
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42456
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42457
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [3] ts=2394311623 dur=2080/2080/2000 seq=42458
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1201 do_2833() Send start packet for [7] ts=2394314503 dur=160/160/2000 seq=42465
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=320/320/2000 seq=42466
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=480/480/2000 seq=42467
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=640/640/2000 seq=42468
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=800/800/2000 seq=42469
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=960/960/2000 seq=42470
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1120/1120/2000 seq=42471
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1280/1280/2000 seq=42472
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1440/1440/2000 seq=42473
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1600/1600/2000 seq=42474
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1760/1760/2000 seq=42475
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send middle packet for [7] ts=2394314503 dur=1920/1920/2000 seq=42476
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42477
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42478
2008-09-11 14:39:57 [DEBUG] switch_rtp.c:1143 do_2833() Send end packet for [7] ts=2394314503 dur=2080/2080/2000 seq=42479

- Marc


Anthony Minessale wrote:
Quote:
2833 is the default.
INBAND is not a function of the sip module it's implemented on top of the abstraction layer.

Are you doing trans coding or different ptimes between the 2 legs?

you might want to turn up the debug log "press f8" and you will pretty verbose logging of the 2833.
also consider taking a pcap of the network traffic.

What is the provider? 2833 is notoriously done wrong to the point that doing it exactly right can put you at a disadvantage.





On Thu, Sep 11, 2008 at 4:58 PM, Marc Lewis <marc@avvatel.com (marc@avvatel.com)> wrote:
Quote:

I've been having a problem getting DTMF passed through from registered
phones to my termination provider. Sometimes it works, sometimes it
doesn't. My provider is set for rfc2833.

In looking through the mod_sofia.conf.xml file, and the sofia sources,
it appears that the only two options for DTMF are info and rfc2833.
I've tried various combinations and the only thing that ever seems to
even sometimes work is if pass-rfc2833 is set to true. I've tried this
with Linksys SPA 942 phones, Grandstream GXP 2000 phones and Cisco 7940
phones. Doesn't seem to make a difference. I've also tried various
dtmf-durations and that didn't seem to have an effect either.

However, after doing various searches it appears that not explicitly
setting a dtmf-type in a sofia profile, it defaults to using inband DTMF.

Is this correct? If inband DTMF is the default, then I will have my
termination provider switch my account over to using inband instead of
rfc2833.

- Marc

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Avvatel Corporation


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Quote:

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[url=sip:brian@freeswitch.org]sip:brian@freeswitch.org[/url]










Quote:


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PostPosted: Thu Sep 11, 2008 5:59 pm    Post subject: [Freeswitch-users] DTMF issues/question Reply with quote

Can you get me a trace of inbound RFC2833... call your cellphone press
1234,, the press 1234 outbound... I need the pcap file so I can see
what they are sending.

/b

On Sep 11, 2008, at 5:51 PM, Marc Lewis wrote:

Quote:
The problems arise when trying to send to other IVR's. My last
tests have been to call my two different banks and american
express. None of the three of them recognized the DTMF tones.

Brian West
sip:brian@freeswitch.org







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