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
Andy <andy@fabulous4.co.uk> wrote:
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.
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:
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
Andy <andy@fabulous4.co.uk> wrote:
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.
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.
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.
participants (2)
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Andy
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Jason White