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andy at fabulous4.co.uk
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 Posted: Fri May 15, 2009 2:14 am Post subject: [Freeswitch-users] DTMF not comming through on some calls |
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| Hi, I have an urgent issue if anyone can help. I have been running freeswitch for 3-4 weeks now without issue. In the last 2 days some of the calls coming into the switch seem to get set up in such a way that means they cannot carry DTMF. ie on that call, no dtmf signals come through from the phone. It's not that digits get dropped some calls semm to handle dtmf perfectly and others don't seem to get dtmf at all. Can anyone shed any light opn this or suggest any solutions? Many thanks Andy |
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jason at jasonjgw.net
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| Posted: Fri May 15, 2009 2:48 am Post subject: [Freeswitch-users] DTMF not comming through on some calls |
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Andy <andy@fabulous4.co.uk> wrote:
| Quote: | It's not that digits get dropped some calls semm to handle dtmf perfectly
and others don't seem to get dtmf at all.
Can anyone shed any light opn this or suggest any solutions?
|
I can't help, but you could make it a lot easier for others to help you by
including the necessary information with your question.
For example, what DTMF method is configured in the SIP profiles - RFC2833 or
Info, or are you using inband DTMF detection?
What are the phones, and how are they connected to your FreeSWITCH system?
What relevant information appears in your FreeSWITCH logs? For example, when
debug-level logging is enabled, you should see log entries related to the DTMF
detection. Check whether there are differences between the calls that work and
those which don't.
If it appears to be a bug, test whether you can reproduce it with the latest
code taken from svn trunk.
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andy at fabulous4.co.uk
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| Posted: Fri May 15, 2009 4:51 am Post subject: [Freeswitch-users] DTMF not comming through on some calls |
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Apologies. The freeswitch software is receiving incoming calls from a voip
gateway. I'm using voiptalk in the UK.
The DTMF method was efault which I believe is "info" but I've now set it
explicitly to rfc2833 inband to see if that helps. Is there a way I can tell
from the logs that this is the case and that my config changes have worked.
Most of the phones are mobiles but some landlines as well.
I've done a detailed analysis of the logs and the calls that don't work are
missing what appear to be critical actions in the debug. Namely:
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
And then a little later in the call....
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp()
Audio params changed for sofia/external/07540526194@194.145.190.143 from
194.145.190.143:11780 to 87.238.72.155:16968
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/07540526194@194.145.190.143] 77.86.49.249 port
21054 -> 87.238.72.155 port 16968 codec: 8 ms: 20
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()
AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968]
2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state()
Processing Reinvite
2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel
sofia/external/07540526194@194.145.190.143 entering state [completed][200]
2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel
sofia/external/07540526194@194.145.190.143 entering state [ready][200]
These lines appear for calls that work and not when they don't.
Hope that helps.
Cheers Andy
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jason
White
Sent: 15 May 2009 08:47
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF not comming through on some calls
Andy <andy@fabulous4.co.uk> wrote:
| Quote: | It's not that digits get dropped some calls semm to handle dtmf
perfectly and others don't seem to get dtmf at all.
Can anyone shed any light opn this or suggest any solutions?
|
I can't help, but you could make it a lot easier for others to help you by
including the necessary information with your question.
For example, what DTMF method is configured in the SIP profiles - RFC2833 or
Info, or are you using inband DTMF detection?
What are the phones, and how are they connected to your FreeSWITCH system?
What relevant information appears in your FreeSWITCH logs? For example, when
debug-level logging is enabled, you should see log entries related to the
DTMF detection. Check whether there are differences between the calls that
work and those which don't.
If it appears to be a bug, test whether you can reproduce it with the latest
code taken from svn trunk.
_______________________________________________
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
_______________________________________________
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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jason at jasonjgw.net
Guest
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| Posted: Fri May 15, 2009 5:21 am Post subject: [Freeswitch-users] DTMF not comming through on some calls |
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Andy <andy@fabulous4.co.uk> wrote:
| Quote: |
The DTMF method was efault which I believe is "info" but I've now set it
explicitly to rfc2833 inband to see if that helps. Is there a way I can tell
from the logs that this is the case and that my config changes have worked.
|
This is in the logs, and (assuming the logs you quoted were taken after any
relevant configuration change), they indicate that RFC2833 is indeed being
used. This is also the default in the supplied Sofia profiles.
| Quote: |
Most of the phones are mobiles but some landlines as well.
I've done a detailed analysis of the logs and the calls that don't work are
missing what appear to be critical actions in the debug. Namely:
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
|
That's turning on RFC2833, as I understand it, for DTMF detection.
| Quote: |
And then a little later in the call....
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp()
Audio params changed for sofia/external/07540526194@194.145.190.143 from
194.145.190.143:11780 to 87.238.72.155:16968
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/07540526194@194.145.190.143] 77.86.49.249 port
21054 -> 87.238.72.155 port 16968 codec: 8 ms: 20
2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()
AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968]
2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state()
Processing Reinvite
2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel
sofia/external/07540526194@194.145.190.143 entering state [completed][200]
2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel
sofia/external/07540526194@194.145.190.143 entering state [ready][200]
These lines appear for calls that work and not when they don't.
|
There are obviously SIP reinvite messages being received from your SIP
provider, and which FreeSWITCH is processing successfully.
What I'm wondering is whether your provider is always offering RFC2833, since,
given the above, they seem to have complex call handling arrangements.
For that, you would need to look at the SDP from the remote end in the calls
for which DTMF isn't being detected properly. Fortunately, this is logged by
FreeSWITCH when set to debug logging.
What you're looking for is a line such as
a=rtpmap:101 telephone-event/8000
If that isn't present, then something odd would appear to be going on at your
SIP provider's end, which is what I personally suspect, since FreeSWITCH is
correctly activating RFC2833 support on the channel in other cases.
You can also obtain a sip trace:
sofia profile external siptrace on
which will show you exactly what you're receiving from your provider.
Disclaimer: I'm not an expert, but I hope this helps anyway.
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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