[Freeswitch-users] No ring tone while recording incoming call. Please help.
Hi, I have trouble recording incoming calls with FreeSwitch. I have followed the instruction from Misc. Dialplan Tools record session (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) It works well for outgoing calls, but I have the problem with incoming calls. The person who is calling does not hear ring tone, he hears just the silence until I pick up the phone. Everything else is working, we can talk, conversation is recorded. Here is a copy of my dialplan for incoming calls /usr/local/freeswitch/conf/dialplan/public/voipms.xml for outcoming calls I have a similar code added to the /usr/local/freeswitch/conf/dialplan/default/user1.xml and it works well. I have tried to move the line between the lines ___ 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
[Freeswitch-users] No ring tone while recording incoming call. Please help.
Brian, Thank yo very much for your reply. I have tried to add transfer_ringback action, but it did not solve my problem. Destination phone is ringing, but the person who is calling does not hear ringing tone in hte handset. Is there anything in the logfile that can help you to identify the problem? Closest I can see is: 2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1738 Raw Codec Activation Success l...@8000hz 1 channel 20ms 2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1797 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/ 4163641...@67.205.74.164 receive message [TRANSCODING_NECESSARY] 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ sip:m...@192.168.0.121:5060 entering state [proceeding][180] 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/ sip:m...@192.168.0.121:5060! 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/ 4163641...@67.205.74.164 entering state [terminated][487] 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/ 4163641...@67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL] Thank you, Igor >set ringback before record_session and also set transfer_ringback >because record_session causes an pre-answer. > >/b > >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: > >> Hi, >> >> I have trouble recording incoming calls with FreeSwitch. >> >> I have followed the instruction from Misc. Dialplan Tools record >> session >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) >> It works well for outgoing calls, but I have the problem with >> incoming calls. >> >> The person who is calling does not hear ring tone, he hears just the >> silence until >> I pick up the phone. Everything else is working, we can talk, >> conversation is recorded. >> >> Here is a copy of my dialplan for incoming calls >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml >> >> >> >> > expression="XX"> >> >> >> > data="RECORD_SOFTWARE=FreeSwitch"/> >> > data="RECORD_ARTIST=FreeSwitch"/> >> > data="RECORD_COMMENT=FreeSwitch"/> >> >> >> >> >> >> >> >> > > ___ 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
[Freeswitch-users] No ring tone while recording incoming call. Please help.
Hi guys, I have solve my problem by adding before I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example and noticed this line. Tried it and it works like a charm. Thanks everybody, especially Brian and MC. Igor > Brian, >> >> Thank you very much for your reply. >> >> I have tried to add transfer_ringback action, but it did not solve my >> problem. >> Destination phone is ringing, but the person who is calling does not >> hear ringing tone in hte handset. >> >> Is there anything in the logfile that can help you to identify the problem? >> >What kind of system is the calling party connected to? It looks like a >180 is sent out by FS: > >2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ >sip:main at 192.168.0.121:5060 entering state [proceeding][180] > >At that point the server at the originating side *should* generate >pretend ringing for the calling phone. If that is not happening then >you need to see what's going on at the originating side. Is it a SIP provider? > >-MC > > >> >> Closest I can see is: >> 2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1738 Raw >> Codec Activation Success L16 at 8000hz 1 channel 20ms >> 2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1797 Play >> Ringback Tone [%(2000,4000,440.0,480.0)] >> 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 >> sofia/external/ >> 4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY] >> 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel >> sofia/internal/ sip:main at 192.168.0.121:5060 entering state >> [proceeding][180] >> 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready >> sofia/internal/ sip:main at 192.168.0.121:5060! >> 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel >> sofia/external/ >> 4163641113 at 67.205.74.164 entering state [terminated][487] >> 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup >> sofia/external/ >> 4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL] >> >> Thank you, >> >> Igor >> >> >set ringback before record_session and also set transfer_ringback >> >because record_session causes an pre-answer. >> > >> >/b >> > >> >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: >> > >> >> Hi, >> >> >> >> I have trouble recording incoming calls with FreeSwitch. >> >> >> >> I have followed the instruction from Misc. Dialplan Tools record >> >> session >> >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_sessi >> >> on) It works well for outgoing calls, but I have the problem with >> >> incoming calls. >> >> >> >> The person who is calling does not hear ring tone, he hears just >> >> the silence until I pick up the phone. Everything else is working, >> >> we can talk, conversation is recorded. >> >> >> >> Here is a copy of my dialplan for incoming calls >> >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml >> >> >> >> >> >> >> >> > >> expression="XX"> >> >> >> >> >> >> > >> data="RECORD_SOFTWARE=FreeSwitch"/> >> >> > >> data="RECORD_ARTIST=FreeSwitch"/> >> >> > >> data="RECORD_COMMENT=FreeSwitch"/> >> >> >> >> >> >> >> >> >> >> > >> data="$${base_dir}/ >> >> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ >> >> {destination_number}_${caller_id_number}.wav"/> >> >> >> >> >> >> >> > >> > >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> > ___ 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
[Freeswitch-users] How to pass Message Waiting Ind. from VOIP provider directly to the phone
Hi Guys, I have a question about configuring Message Waiting Indicator (MWI). I am running Freeswitch on Ubuntu 8.04 Desktop. I have Freeswitch connected to the external VOIP provider (voip.ms) and to the internal SIP box (Linksys RTP300) with one phone hooked to it. I let my VOIP provider to handle voicemail, and I would like Freeswitch to passthrough Message Waiting Indicator to the SIP box (and eventially to the phone), so it turns on when I have a new message in the VOIP provider voicemail . How to configure Freeswitch to achieve this? Before using Freeswitch I had SIP box hooked to my external provider directly, and I was getting Message Waiting Indicator on the phone no problem, but since I put FreeSwitch, it does not work anymore. I suspect it is because Freeswitch has its own voicemail system and it triggers MWI based on its state. I want to keep things simple and keep my voicemail at my VOIP provider. Thank you, Svetik ___ 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