- FreeSWITCH-users - FreeSWITCH Mailing Lists

"No Digits to send" on purpose?
by Jeroen C. van Gelderen 02 Dec '10

02 Dec '10

Hi Guys, I need to bridge an FXS port (when taken off-hook) to an FXO port without dialing any digits like so: <action application="bridge" data="freetdm/span_fxo_csd/1"/> This results in the following error: ftmod/ftmod_analog/ftmod_analog.c: if (ftdm_strlen_zero(ftdmchan->caller_data.dnis.digits)) { ftdm_log_chan_msg(ftdmchan, FTDM_LOG_ERROR, "No Digits to send!\n"); For now I have worked around this by using <action application="bridge" data="freetdm/span_fxo_csd/1/w"/> or <action application="bridge" data="freetdm/span_fxo_csd/1/F"/> I would like to avoid the unnecessary delay this incurs. Any suggestions? Under what circumstances is it bad to send zero digits? Can I simply take out the check for zero-length dnis.digits? Cheers, -Slim -- Jeroen C. "Slim" van Gelderen

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Re: [Freeswitch-users] mod_perl comple problem
by Madovsky 02 Dec '10

02 Dec '10

I solved it by installing yum install db4-devel thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, December 02, 2010 2:37 PM Subject: mod_perl comple problem have this problem at complie : /usr/bin/ld: cannot find -ldb fedora10 64bits. thanks Franck

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No CDR
by Kilburn Abrahams 02 Dec '10

02 Dec '10

Hi List Asterisk has useful feature called NoCDR. Prefer not to record CDR's for internal dialing. I am using mod_xml_cdr. Is there an equivalent feature or a work around to achieve this Thanks Kilburn

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mod_perl comple problem
by Madovsky 02 Dec '10

02 Dec '10

have this problem at complie : /usr/bin/ld: cannot find -ldb fedora10 64bits. thanks Franck

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can't get channel variables after hangup
by Madovsky 02 Dec '10

02 Dec '10

I set this in my dialplan : <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="api_hangup_hook=system /usr/local/bin/php /usr/local/freeswitch/scripts/php/test.php ${uuid}"/> and this in test.php: exec("/usr/local/freeswitch/bin/fs_cli -x \"uuid_getvar ".$argv[1]." test_var\"", $exTabRet, $req); it results : -ERR No Such Channel! I tried to replace ${uuid} by ${call_uuid} without success. Thanks Franck

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How to force the usage of g729 instead of g729b
by Jan Riedinger 02 Dec '10

02 Dec '10

By setting global_codec_prefs and inbound_codec_prefs it's possible to specify, which codecs are supported for inbound calls. How can I specify that the switch shall support g729, but not g729b? The incoming INVITE message contains a SDP with the media attribute annexb=yes. How has Freeswitch to answer to indicate that g729 is supported but g729b is not supported? Thank you in advance Jan -- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger(a)sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan(a)sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger AG Charlottenburg - HRB 71973

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Re: [Freeswitch-users] Sangoma D100 usage
by Javier Gallart 02 Dec '10

02 Dec '10

On Thu, Dec 2, 2010 at 2:56 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Sangoma D100 usage (Javier Gallart) > 2. Re: Sangoma D100 usage (Steven Ayre) > 3. Re: Sangoma D100 usage (David Ponzone) > 4. Re: How to filter Skypopen chat message in ESL mode? > (Giovanni Maruzzelli) > > > ---------- Forwarded message ---------- > From: Javier Gallart <jgallartm(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 2 Dec 2010 13:55:33 +0100 > Subject: [Freeswitch-users] Sangoma D100 usage > Hello > > we got a Sangoma D100 transcoding card, and we like to get it working with > our freeswitch. Freeswitch and the Sangoma card are running on the same > server. The module is properly loaded: > > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading > sangoma codec configuration > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma > RTP IP x.x.x.x > 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and > activated 1 Sangoma codec vocallo modules > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs, > register='all', noregister='' > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering > implementations for codec PCMU > ... > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering > implementations for codec PCMA > ... > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering > implementations for codec G729 > ... > 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946 > Successfully Loaded [mod_sangoma_codec] > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps > ... > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps > ... > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > Codec G729 18 Sangoma G729 8000hz 40ms 8000bps > ... > 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API > Function 'sangoma_codec' > > The codecs look also good: > freeswitch@internal> show codecs > codec,Sangoma G723,mod_sangoma_codec > codec,Sangoma G729,mod_sangoma_codec > codec,Sangoma GSM,mod_sangoma_codec > codec,Sangoma PCMA,mod_sangoma_codec > codec,Sangoma PCMU,mod_sangoma_codec > > What we don't know is how to actually use the card. We have forced a call > with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how > to bridge them. I've tried unsuccessfully this config in the vars.xml: > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma > PCMU,Sangoma PCMA"/> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma > PCMA"/> > > Does anybody has any experience usgin this card? > > Thanks > > Javier > > > > > > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Thu, 2 Dec 2010 13:48:42 +0000 > Subject: Re: [Freeswitch-users] Sangoma D100 usage > Sangoma is not part of the codec IANA name, just part of the interface > name (description). Try it like this: > > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/> > > Don't have mod_g729 or mod_com_g729 loaded at the same time. > > -Steve > Hi Steve removing the mod_g729 module did the trick. Thanks! Javi > > On 2 December 2010 12:55, Javier Gallart <jgallartm(a)gmail.com> wrote: > > Hello > > we got a Sangoma D100 transcoding card, and we like to get it working > with > > our freeswitch. Freeswitch and the Sangoma card are running on the same > > server. The module is properly loaded: > > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading > sangoma > > codec configuration > > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma > > RTP IP x.x.x.x > > 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 > and > > activated 1 Sangoma codec vocallo modules > > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading > codecs, > > register='all', noregister='' > > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering > > implementations for codec PCMU > > ... > > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering > > implementations for codec PCMA > > ... > > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering > > implementations for codec G729 > > ... > > 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946 > > Successfully Loaded [mod_sangoma_codec] > > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > > Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps > > ... > > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > > Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps > > ... > > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > > Codec G729 18 Sangoma G729 8000hz 40ms 8000bps > > ... > > 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding > API > > Function 'sangoma_codec' > > The codecs look also good: > > freeswitch@internal> show codecs > > codec,Sangoma G723,mod_sangoma_codec > > codec,Sangoma G729,mod_sangoma_codec > > codec,Sangoma GSM,mod_sangoma_codec > > codec,Sangoma PCMA,mod_sangoma_codec > > codec,Sangoma PCMU,mod_sangoma_codec > > What we don't know is how to actually use the card. We have forced a call > > with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know > how > > to bridge them. I've tried unsuccessfully this config in the vars.xml: > > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma > > PCMU,Sangoma PCMA"/> > > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma > > PCMA"/> > > Does anybody has any experience usgin this card? > > Thanks > > Javier > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > ---------- Forwarded message ---------- > From: David Ponzone <david.ponzone(a)ipeva.fr> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Thu, 2 Dec 2010 14:48:48 +0100 > Subject: Re: [Freeswitch-users] Sangoma D100 usage > Javier, > > as Sangoma is working quite closely with FreeSWITCH devs (some Sangoma > people are actually on this ML), I am pretty sure you can ask Sangoma > support to help you out. > > David Ponzone Direction Technique > email: david.ponzone(a)ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pièces jointes sont confidentiels et établis à > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autorisée est interdite. Tout message électronique est susceptible > d'altération. **IPeva** décline toute responsabilité au titre de ce > message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas > destinataire de ce message, merci de le détruire immédiatement et d'avertir > l'expéditeur.* > * > * > > > > Le 02/12/2010 à 13:55, Javier Gallart a écrit : > > Hello > > we got a Sangoma D100 transcoding card, and we like to get it working with > our freeswitch. Freeswitch and the Sangoma card are running on the same > server. The module is properly loaded: > > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading > sangoma codec configuration > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma > RTP IP x.x.x.x > 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and > activated 1 Sangoma codec vocallo modules > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs, > register='all', noregister='' > 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering > implementations for codec PCMU > ... > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering > implementations for codec PCMA > ... > 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering > implementations for codec G729 > ... > 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946 > Successfully Loaded [mod_sangoma_codec] > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps > ... > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps > ... > 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding > Codec G729 18 Sangoma G729 8000hz 40ms 8000bps > ... > 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API > Function 'sangoma_codec' > > The codecs look also good: > freeswitch@internal> show codecs > codec,Sangoma G723,mod_sangoma_codec > codec,Sangoma G729,mod_sangoma_codec > codec,Sangoma GSM,mod_sangoma_codec > codec,Sangoma PCMA,mod_sangoma_codec > codec,Sangoma PCMU,mod_sangoma_codec > > What we don't know is how to actually use the card. We have forced a call > with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how > to bridge them. I've tried unsuccessfully this config in the vars.xml: > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma > PCMU,Sangoma PCMA"/> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma > PCMA"/> > > Does anybody has any experience usgin this card? > > Thanks > > Javier > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: Giovanni Maruzzelli <gmaruzz(a)celliax.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Thu, 2 Dec 2010 14:55:51 +0100 > Subject: Re: [Freeswitch-users] How to filter Skypopen chat message in ESL > mode? > 1) Is that the complete skypopen.conf.xml? I mean, do you use one only > interface? > > 2) An incoming skype call goes directly (because is in > skypopen.conf.xml) to 5655 and ear the continuous message. > > 3) While the incoming skype call is hearing the message, the same > incoming skypeclient sends chatmessages to the skypopen skypeclient > > 4) You connect via telnet to the ESL port, do the "events plain message" > > 5) You get sometimes those longer events intermixed with the regular > message events, but not everytime > > Is this correct? If this is correct I will test asap and fix the > possible problem. > > Btw, I tested yesterday with skypeclient not in a call and I got no > problems at all. I don't think to be in a call would make any > difference. > > Please be certain you gave me all the info needed to exactly replicate > your problem. > > -giovanni > > On Thu, Dec 2, 2010 at 12:09 PM, xuyan yang <xyangni(a)gmail.com> wrote: > > OK. I will try to make it clear. > > In general, my test case is to forward skype call to a extention such as > > 5655. And then in the dialplan for 5655, continuously play some sound. > > During playing, you can send skype chat messages from the caller and > check > > what esl events has been generated from it. According to my test, the raw > > type event is not generated for most of chat messages. > > In details, > > this is my skypeopen.conf.xml: > > <configuration name="skypopen.conf" description="Skypopen Configuration"> > > <global_settings> > > <param name="debug" value="8"/> > > <param name="dialplan" value="XML"/> > > <param name="context" value="default"/> > > <param name="destination" value="1000"/> > > <param name="skype_user" value="user1"/> > > <param name="report_incoming_chatmessages" value="true"/> > > <param name="silent_mode" value="false"/> > > <param name="write_silence_when_idle" value="true"/> > > </global_settings> > >  > > <per_interface_settings> > > <interface id="1" name="interface1"> > > <param name="X11-display" value=":0.0"/> > > <param name="skype_user" value="xxxxxxxxx"/> > > <param name="destination" value="5655"/> > > </interface> > > </per_interface_settings> > > </configuration> > > > > > > in dialplan for default context, add the following to describe extention > > 5655: > > <extension name="skype" break="on-false"> > > <condition field="destination_number" expression="^5655$"> > > <action application="answer"/> > > <action application="sleep" data="1000"/> > > <action application="playback" data="ivr/ivr-sample_submenu.wav"/> > <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action > > application="playback" data="ivr/ivr-sample_submenu.wav"/> > > </condition> > > </extension> > > ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it > can > > be replaced by any other wav files. > > after setup, make a call to your skype user xxxxxxxxx. > > in the mean time use fs_cli to connect FS and execute /event plain > MESSAGE > > to listen esl events. > > sent any chat messages from caller, then you are very likely to find only > > the short version events. > > > > > > On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli <gmaruzz(a)celliax.org > > > > wrote: > >> > >> Xuyan, > >> > >> please. > >> > >> Can you give a clear defined procedure to reproduce the problem? > >> > >> Please, take the time to write it, complete of all relevant info > >> (dialplan, script, etc etc) so I can reproduce it from a freshly > >> installed FreeSWITCH. > >> > >> If you don't give this information I cannot fix the problem. And I > >> cannot neither read your mind, nor finding the time to try every > >> possible combination. > >> > >> Please, post something someone can cut and paste and a clear step by > >> step procedure to replicate. > >> > >> -giovanni > >> > >> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni(a)gmail.com> wrote: > >> > I tried it again. When the line is parked and idle. the raw events can > >> > always be generated. > >> > but if the line is kept busy such as the following case this events is > >> > trend > >> > to have problem > >> > <extension name="skype-in" break="on-false"> > >> > <condition field="destination_number" expression="^5655$"> > >> > <action application="answer"/> > >> > <action application="sleep" data="1000"/> > >> >  > >> > <action application="playback" data="test.wav"/> > >> > <action application="playback" data="test.wav"/> > >> > <action application="playback" data="test.wav"/> > >> > <action application="playback" > >> > data="test.wav"/> > >> > <action application="playback" data="test.wav"/> > >> > <action application="playback" data="test.wav"/> > >> > <action application="playback" > >> > data="test.wav"/> > >> > <action application="playback" data="test.wav"/> > >> > <action application="playback" data="test.wav"/> > >> > </condition> > >> > </extension> > >> > > >> > > >> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli > >> > <gmaruzz(a)celliax.org> > >> > wrote: > >> >> > >> >> I repeat, if you subscribe to "message" events you get only those. > >> >> Also, how I can replicate your problem? > >> >> I've tested events and I had no problem at all with spurious or > >> >> unreliable events in mod-skypopen. > >> >> Please, can you indicate a detailed way to reproduce your problem? > >> >> -giovanni > >> >> > >> >> On 12/2/10, xuyan yang <xyangni(a)gmail.com> wrote: > >> >> > I got it. Then the problem should be the unstable behavior of raw > >> >> > events > >> >> > which has about 25% chances of being missed. > >> >> > > >> >> > Fortunately, I have found a way to avoid this issue. Ignore all > >> >> > message > >> >> > events which contains Unique-ID field. > >> >> > > >> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli > >> >> > <gmaruzz(a)celliax.org>wrote: > >> >> > > >> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli > >> >> >> <gmaruzz(a)celliax.org> wrote: > >> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get > only > >> >> >> > those. > >> >> >> > > >> >> >> > The other events are "raw" events, that other users have > requested > >> >> >> > for > >> >> >> > other purposes. > >> >> >> > >> >> >> or at least that is the expected behavior, please let me know if > I > >> >> >> introduced some regression in integrating that "raw event" thingy. > >> >> >> > >> >> >> -giovanni > >> >> >> > >> >> >> > > >> >> >> > -giovanni > >> >> >> > > >> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni(a)gmail.com> > >> >> >> > wrote: > >> >> >> >> Thanks for your reply. I have read this page before. but the > >> >> >> >> random > >> >> >> emerging > >> >> >> >> of the second verbose esl events is causing trouble. And I > guess > >> >> >> >> it > >> >> >> >> may > >> >> >> be a > >> >> >> >> bug, or it should be predictable. > >> >> >> >> Before reporting to jira, I just want to check whether I have > >> >> >> >> made > >> >> >> >> any > >> >> >> >> mistake. > >> >> >> >> > >> >> >> >> > >> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli > >> >> >> >> <gmaruzz(a)gmail.com> > >> >> >> >> wrote: > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> > >> >> >> > >> >> >> > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSA… > >> >> >> >>> > >> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang < > xyangni(a)gmail.com> > >> >> >> >>> wrote: > >> >> >> >>> > hi, > >> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head > >> >> >> >>> > (git-8825b6e > >> >> >> >>> > 2010-11-28 17-15-39 -0500) > >> >> >> >>> > I need to handle skype chat message with a inbound ESL > >> >> >> >>> > connection. > >> >> >> But > >> >> >> >>> > there > >> >> >> >>> > are 1 or 2 esl events received randomly for each message. > >> >> >> >>> > the first short one is alway generated, but the second one > is > >> >> >> >>> > random. > >> >> >> So > >> >> >> >>> > when trying to filter according to UUID, nothing is caught > in > >> >> >> >>> > many > >> >> >> >>> > cases. > >> >> >> >>> > If I make filter based on skype ID, duplicated messages are > >> >> >> >>> > often > >> >> >> heard. > >> >> >> >>> > I do not why the behavior of the second event is random. > >> >> >> >>> > How should I setup the filter to get 1 and only 1 event for > >> >> >> >>> > each > >> >> >> >>> > chat > >> >> >> >>> > message? Thanks. > >> >> >> >>> > > >> >> >> >>> > the first is a short one with the following header: > >> >> >> >>> > [Event-Name] = [MESSAGE] > >> >> >> >>> > [Event-Calling-Function] = [incoming_chatmessage] > >> >> >> >>> > [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT] > >> >> >> >>> > [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500] > >> >> >> >>> > [Content-Length] = [1] > >> >> >> >>> > [subject] = [SIMPLE MESSAGE] > >> >> >> >>> > [FreeSWITCH-IPv4] = [192.168.0.3] > >> >> >> >>> > [hint] = [niqizhi] > >> >> >> >>> > [from] = [niqizhi] > >> >> >> >>> > [Event-Date-Local] = [2010-12-01 21:02:37] > >> >> >> >>> > [proto] = [skype] > >> >> >> >>> > [FreeSWITCH-IPv6] = [::1] > >> >> >> >>> > [id] = [5334] > >> >> >> >>> > [Event-Calling-File] = [mod_skypopen.c] > >> >> >> >>> > [Event-Date-Timestamp] = [1291237357051788] > >> >> >> >>> > [FreeSWITCH-Hostname] = [EYSRV] > >> >> >> >>> > [login] = [interface1] > >> >> >> >>> > [during-call] = [true] > >> >> >> >>> > [Event-Calling-Line-Number] = [2915] > >> >> >> >>> > [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a] > >> >> >> >>> > the second is much longer: > >> >> >> >>> > [Caller-Source] = [mod_skypopen] > >> >> >> >>> > [Event-Calling-Function] = [incoming_chatmessage] > >> >> >> >>> > [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT] > >> >> >> >>> > [Answer-State] = [answered] > >> >> >> >>> > [FreeSWITCH-IPv4] = [192.168.0.3] > >> >> >> >>> > [Channel-State] = [CS_EXECUTE] > >> >> >> >>> > [Channel-Read-Codec-Bit-Rate] = [256000] > >> >> >> >>> > [FreeSWITCH-IPv6] = [::1] > >> >> >> >>> > [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500] > >> >> >> >>> > [Channel-Read-Codec-Rate] = [16000] > >> >> >> >>> > [Caller-Destination-Number] = [5655] > >> >> >> >>> > [Caller-Channel-Transfer-Time] = [0] > >> >> >> >>> > [Channel-Call-State] = [ACTIVE] > >> >> >> >>> > [Caller-Channel-Progress-Media-Time] = [0] > >> >> >> >>> > [FreeSWITCH-Hostname] = [EYSRV] > >> >> >> >>> > [Caller-Channel-Answered-Time] = [1291237326697085] > >> >> >> >>> > [login] = [interface1] > >> >> >> >>> > [during-call] = [true] > >> >> >> >>> > [Channel-Name] = [skypopen/interface1] > >> >> >> >>> > [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500] > >> >> >> >>> > [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500] > >> >> >> >>> > [Channel-Read-Codec-Name] = [L16] > >> >> >> >>> > [Caller-Channel-Name] = [skypopen/interface1] > >> >> >> >>> > [Caller-Caller-ID-Number] = [niqizhi] > >> >> >> >>> > [Event-Date-Timestamp] = [1291237357051788] > >> >> >> >>> > [Channel-State-Number] = [4] > >> >> >> >>> > [Event-Calling-Line-Number] = [2888] > >> >> >> >>> > [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a] > >> >> >> >>> > [Event-Name] = [MESSAGE] > >> >> >> >>> > [Content-Length] = [1] > >> >> >> >>> > [subject] = [SIMPLE MESSAGE] > >> >> >> >>> > [Caller-Caller-ID-Name] = [niqizhi] > >> >> >> >>> > [from] = [niqizhi] > >> >> >> >>> > [Caller-Dialplan] = [XML] > >> >> >> >>> > [Caller-Channel-Hangup-Time] = [0] > >> >> >> >>> > [id] = [5334] > >> >> >> >>> > [Caller-Profile-Index] = [1] > >> >> >> >>> > [Caller-Direction] = [inbound] > >> >> >> >>> > [Caller-Username] = [skypopen] > >> >> >> >>> > [Channel-Write-Codec-Name] = [L16] > >> >> >> >>> > [Call-Direction] = [inbound] > >> >> >> >>> > [Caller-Screen-Bit] = [true] > >> >> >> >>> > [hint] = [niqizhi] > >> >> >> >>> > [Caller-Privacy-Hide-Number] = [false] > >> >> >> >>> > [Event-Date-Local] = [2010-12-01 21:02:37] > >> >> >> >>> > [proto] = [skype] > >> >> >> >>> > [Caller-Channel-Created-Time] = [1291237326468855] > >> >> >> >>> > [Event-Calling-File] = [mod_skypopen.c] > >> >> >> >>> > [Caller-Channel-Progress-Time] = [0] > >> >> >> >>> > [Caller-Privacy-Hide-Name] = [false] > >> >> >> >>> > [Channel-Write-Codec-Rate] = [16000] > >> >> >> >>> > [Caller-Context] = [default] > >> >> >> >>> > [Channel-Write-Codec-Bit-Rate] = [256000] > >> >> >> >>> > [Presence-Call-Direction] = [inbound] > >> >> >> >>> > [Caller-Profile-Created-Time] = [1291237326468855] > >> >> >> >>> > _______________________________________________ > >> >> >> >>> > FreeSWITCH-users mailing list > >> >> >> >>> > FreeSWITCH-users(a)lists.freeswitch.org > >> >> >> >>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> >>> > UNSUBSCRIBE: > >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> >>> > http://www.freeswitch.org > >> >> >> >>> > > >> >> >> >>> > > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> -- > >> >> >> >>> Sincerely, > >> >> >> >>> > >> >> >> >>> Giovanni Maruzzelli > >> >> >> >>> Cell : +39-347-2665618 > >> >> >> >>> > >> >> >> >>> _______________________________________________ > >> >> >> >>> FreeSWITCH-users mailing list > >> >> >> >>> FreeSWITCH-users(a)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 mailing list > >> >> >> >> FreeSWITCH-users(a)lists.freeswitch.org > >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> >> UNSUBSCRIBE: > >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> >> http://www.freeswitch.org > >> >> >> >> > >> >> >> >> > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > -- > >> >> >> > Sincerely, > >> >> >> > > >> >> >> > Giovanni Maruzzelli > >> >> >> > Cell : +39-347-2665618 > >> >> >> > > >> >> >> > >> >> >> > >> >> >> > >> >> >> -- > >> >> >> Sincerely, > >> >> >> > >> >> >> Giovanni Maruzzelli > >> >> >> Cell : +39-347-2665618 > >> >> >> > >> >> >> _______________________________________________ > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users(a)lists.freeswitch.org > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > >> >> >> > >> >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> http://www.freeswitch.org > >> >> >> > >> >> > > >> >> > >> >> -- > >> >> Sent from my mobile device > >> >> > >> >> Sincerely, > >> >> > >> >> Giovanni Maruzzelli > >> >> Cell : +39-347-2665618 > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users(a)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 mailing list > >> > FreeSWITCH-users(a)lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)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 mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

1 0

: Skypopen error
by Jian Ren 02 Dec '10

02 Dec '10

Could anyone tell me what's this error about? It happened to me quite often. 2010-11-18 20:45:52.690158 [ERR] skypopen_protocol.c:259 [|] [ERRORA 259 ][interface00 ][UP,INPROGRS] Skype got ERROR: |||ERROR 21 Unknown/disallowed call prop||| 2010-11-18 20:45:52.690158 [ERR] skypopen_protocol.c:261 [|] [ERRORA 261 ][interface00 ][UP,FNSHED] skype_call now is DOWN Thanks! Jian

5 18

Lua not playing any WAV files
by Marc de Corny 02 Dec '10

02 Dec '10

Hi all, I have run into an issue on something so basic that I must be as simple as enabling a feature somewhere. I have been trying to get lua to play a message from a WAV file. I have tried session:execute("playback", main_msg) and session:streamFile(ivr_invalid_msg) but neither of them play any music to the caller. I tried both to answer and preAnswer the call first but it made no difference. However if I put the same file into the XML dialplan and play it with the commands below I hear the music fine. <action application="set" data="playback_terminators=#"/> <action application="playback" data="${401_sound_file_path}"/> The issue only seems to be from lua when playing any type of wav file and those files are definitelly there as can be read by the XML The error message is below for the execute(playback) command, but nothing can be seen for the 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playback Requires media! pre_answering channel sofia/external/2031701665@194.0.147.16:5060 EXECUTE sofia/external/2031701665@194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav) But there is no mention of the streamFile command. I have had similar issue with the PlayAndGetDigits command. Is there something that I need to enable in lua so that is can playback messages to the caller. Many thanks to anyone who can help. Marc below is the XML dialplan and lua script as well as the log at the very end. XML DIALPLAN: <include> <extension name="IVR_FROM_MYSQL"> <condition field="caller_id_number" expression="Anonymous" break="never"> <action application="set" data="effective_caller_id_number=0000000000"/> </condition> <condition field="destination_number" expression="^(404)"> <action inline="true" application="lua" data="ivr_mysql.lua ${destination_number:3}"/> <action application="set" data="effective_caller_id_name=${404_tag}"/> <action application="bridge" data="${404_dial}"/> </condition> </extension> </include> The LUA script ivr_mysql.lua is callsed and this is it. -- IVR : PLAY IVR WAV FILES -- Global Variables: local dialstr_prefix = "sofia/gateway/CS2k/" local dialstr_main = "" local breakoutcode = "184" local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/" local ddi = argv[1] -- answer the call session:preAnswer(); freeswitch.consoleLog("info", "All Answered\n"); ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav" main_msg = sound_file_folder .. "default_autoattendant.wav" -- Play with Execute session:execute("playback", main_msg) -- Play with StreamFile session:streamFile(ivr_invalid_msg); dialstr_main = dialstr_main .. dialstr_prefix .. breakoutcode .. "02031701665" session:setVariable("404_dial",dialstr_main) session:setVariable("404_tag","IVR") RELEVANT LOGS : Dialplan: sofia/external/2031701665@194.0.147.16:5060 Regex (PASS) [IVR_FROM_MYS QL] destination_number(4042031956241) =~ /^(404)/ break=on-false Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action lua(ivr_mysql.lua $ {destination_number:3}) INLINE EXECUTE sofia/external/2031701665@194.0.147.16:5060 lua(ivr_mysql.lua 2031956241 ) 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP [sofia/external/2 031701665@194.0.147.16:5060] 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c odec: 8 ms: 20 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft] 160 b ytes per 20ms 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send payload to 101 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive paylo ad to 101 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP: v=0 o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105 s=FreeSWITCH c=IN IP4 10.5.2.105 t=0 0 m=audio 29900 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer sofia/external/2 031701665@194.0.147.16:5060! 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544 (sofia/external/2031701 665@194.0.147.16:5060) Callstate Change RINGING -> EARLY 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel sofia/external/203170166 5@194.0.147.16:5060skipping state [early][183] 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal sofia/e xternal/2031701665@194.0.147.16:5060[BREAK] 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playba ck Requires media! pre_answering channel sofia/external/2031701665(a)194.0.147.16: 5060 EXECUTE sofia/external/2031701665@194.0.147.16:5060playback(/usr/local/freeswit ch/sounds/svc_sound_files/default_autoattendant.wav) 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972 sofia/external/2031701665@ 194.0.147.16:5060destroy/unlink session from object Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action set(effective_calle r_id_name=${404_tag}) Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action bridge(${404_dial}) 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119 (sofia/extern al/2031701665@194.0.147.16:5060) State Change CS_ROUTING -> CS_EXECUTE 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal sofia/ external/2031701665@194.0.147.16:5060[BREAK]

1 0

Sangoma D100 usage
by Javier Gallart 02 Dec '10

02 Dec '10

Hello we got a Sangoma D100 transcoding card, and we like to get it working with our freeswitch. Freeswitch and the Sangoma card are running on the same server. The module is properly loaded: 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading sangoma codec configuration 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma RTP IP x.x.x.x 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and activated 1 Sangoma codec vocallo modules 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs, register='all', noregister='' 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec PCMU ... 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec PCMA ... 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec G729 ... 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [mod_sangoma_codec] 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps ... 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps ... 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 Sangoma G729 8000hz 40ms 8000bps ... 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API Function 'sangoma_codec' The codecs look also good: freeswitch@internal> show codecs codec,Sangoma G723,mod_sangoma_codec codec,Sangoma G729,mod_sangoma_codec codec,Sangoma GSM,mod_sangoma_codec codec,Sangoma PCMA,mod_sangoma_codec codec,Sangoma PCMU,mod_sangoma_codec What we don't know is how to actually use the card. We have forced a call with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how to bridge them. I've tried unsuccessfully this config in the vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma PCMU,Sangoma PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma PCMA"/> Does anybody has any experience usgin this card? Thanks Javier

3 2