April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Load mod_odbc_query error
by FERNANDO VILLARROEL 29 Apr '11

29 Apr '11

Hi all. I getting mod_odbc_query from contrib. My FS Version: FreeSWITCH Version 1.0.head (git-496fd5d 2011-04-10 23-16-56 +0200) But when i load the module i get the follow error: freeswitch@internal> load mod_odbc_query +OK Reloading XML -ERR [module load file routine returned an error] freeswitch@internal> 2011-04-28 18:41:22.665848 [INFO] mod_enum.c:755 ENUM Reloaded 2011-04-28 18:41:22.665848 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_odbc_query.so **/usr/local/freeswitch/mod/mod_odbc_query.so: invalid ELF header** 2011-04-28 18:41:22.665848 [INFO] switch_time.c:999 Timezone reloaded 530 definitions My steps are: mkdir /tmp/fs cd /tmp/fs git clone git://git.freeswitch.org/freeswitch-contrib.git cd freeswitch-contrib/ledr/c/ mv mod_odbc_query/ /usr/src/freeswitch/src/mod/xml_int cd /usr/src/freeswitch/src/mod/xml_in/mod_odbc_query make cp mod_odbc_query.so /usr/local/freeswitch/mod/ cp odbc_query.conf.xml /usr/local/freeswitch/conf/autoload_config What are doing wrong? Regards.

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Call Hold scenario, please help.
by Dave Bracken 28 Apr '11

28 Apr '11

This is my scenario, i have a caller call in, and i bridge them to an operator. The operator would need to put the caller on hold(with music) and then the operator freed up for other calls,etc. Then the operator can pull the caller back off hold. I need to know how to do this using the commands on an "outbound" socket connection to fs. What the commands are, the order, etc. Cant seem to find anyone who knows this yet, so i hope one of you fs gurus can point me in the right direction. Thanks in advance, Dave in irc aka FSnewbDelphi.

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[OT] SIP Trunk Service
by Rob Hutton 28 Apr '11

28 Apr '11

Sorry for the OT post.... I am having a hard time finding someone that can deliver reliable, business grade SIP trunks and port existing numbers at a service location (706-776) in Northern Georgia. If there is anyone that either can provide service, thinks they can provide service, or knows of someone that think they can provide service, please contact me off list... Thanks, Rob

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attended transfer to gateway
by elijah 28 Apr '11

28 Apr '11

Attended transfers to an external c-leg is breaking 3-way conference functionality for me. Like this: <action application="att_xfer" data="sofia/gateway/onesource/${attxfer_callthis}"/> The c-leg is established and hanging up the originator's leg will successfully bridge the other 2. However, pressing 0 on the originator, which should establish a 3-way conference, actually hangs up the c-leg. Is there another method by which I should specify the outbound leg for att_xfer?

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ESL & DTMF
by Antonio Teixeira 28 Apr '11

28 Apr '11

Good Morning. I'm currently in the process of converting my IVR's into ESL , freedom :D. >From my testing is extremely flexible until i have reached a wall. We use TTS to generate alot of information for our users then we need to collect some answers, in session i had GetDigists , yay :). In ESL i have the DTMF event the problem is that from my testing and wiki , DTMF event is only send down the socket using asycn , now i need "sync" because the IVRs need to understand the exact position of the user and blah blah blah , need to be dynamically extended in realtime and would be hard to implement in async , specially the timing for digit collection. So i tried capturing the event with no luck the only one i got is SERVER_DISCONNECT at the end. I think It is possible if for every thread i would raise a listener for that specific uuid that would share memory with the first one so i could read that DTMF events that arrives trough the other socket but i think this is way to much duct tape for a simple thing. So any help on this ? Thanks A/T

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transfer_ringback triggered by "183 Session progress"
by Irina Ivanova 28 Apr '11

28 Apr '11

Hi, I've noticed that if to set transfer_ringback (to any ringback tone) for already answered call and then to execute the bridge to some external number through the gateway, the ringing is triggered not only by "180 Ringing" SIP response, but also when "183 Session progress" is received. Does anybody know if there is a way to make transfer_ringback not to be triggered by 183? I need it because in the case when the destination number is busy and provider sends me 183 and then 486 (Busy here) the caller hears one ringback tone and then the busy tone which makes an impression that the called party rejected the call. Thanks, Irina -- ================================================================ Distinti saluti -- Irina Ivanova Settore Sviluppo MasterVoice tel: +39 0522 1590100 fax: +39 0522 331673 mob: +39 334 6449290 e-mail: i.ivanova(a)mastervoice.it web: www.mastertraining.it - www.registroelettronico.com Master Training S.r.l. Sede Legale: via Timolini, 18 - Correggio (RE) - Italy Sede Operativa: via Sani, 15 - Reggio Emilia - Italy Sede Commerciale: via Sani, 9 - Reggio Emilia - Italy ================================================================ Le informazioni contenute in questa e-mail sono da considerarsi confidenziali e esclusivamente per uso personale dei destinatari sopra indicati. Questo messaggio può includere dati personali o sensibili. Qualora questo messaggio fosse da Voi ricevuto per errore vogliate cortesemente darcene notizia a mezzo e-mail e distruggere il messaggio ricevuto erroneamente. Quanto precede ai fini del rispetto del Decreto Legislativo 196/2003 sulla tutela dei dati personali e sensibili. This e-mail and any file transmitted with it is intended only for the person or entity to which is addressed and may contain information that is privileged, confidential or otherwise protected from disclosure.Copying, dissemination or use of this e-mail or the information herein by anyone other than the intended recipient is prohibited. If you have received this e-mail by mistake, please notify us immediately by telephone or fax.

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bypass_media=true and socket
by Tim Meade 28 Apr '11

28 Apr '11

We are somewhat new to freeswitch, but have found it to be an excellent solution to some projects. With bypass_media=false before our socket command and the perl app returning the bridge everything works fine. Calls sound great. But if we set bypass_media=true immediately before our socket command we receive an almost immediate CANCEL back from ????? If we just hard code the bridge instead of the socket, and bypass_media=true then everything works fine. It's just the combination of bypass_media=false and socket. Here is the dialplan.... <include> <extension name="default"> <condition field="destination_number" expression="^\d\d\d\d\d\d\d+$"> <action application="set" data="bypass_media=true"/> <action application="socket" data="0.0.0.0.0:8040 full"/> </condition> </extension> </include> I have also tried: <action application="set" data="bypass_media_after_bridge=true"/> And the behavior is the same. Is there something simple we may be missing? Much thanks Tim

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Holding properly on "outbound" socket
by Dave Bracken 28 Apr '11

28 Apr '11

This is my scenario, i have a caller call in, and i bridge them to an operator. The operator would need to put the caller on hold(with music) and then the operator freed up for other calls,etc. Then the operator can pull the caller back off hold. I need to know how to do this using the commands on an "outbound" socket connection to fs. What the commands are, the order, etc. Cant seem to find anyone who knows this yet, so i hope one of you fs gurus can point me in the right direction. Thanks in advance, Dave in irc aka FSnewbDelphi.

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custom kernel
by Bojan Sukalo 27 Apr '11

27 Apr '11

Hello. Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To Code Repository" Where I can download this custom kernel and libc code?

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Instant message problem
by Kenneth Taylor 27 Apr '11

27 Apr '11

Hello, I have a strange problem when trying to send IM messages between two clients that connected to FS. When not during call, I'm sending a MESSAGE with the IM string and it works well. But when the two clients are during call, I see in wireshark that the message is coming to FS, but not forwarded to the second client. There is nothing in the log file regarding this message. Is this a known bug? Thank you for helping, Kenneth.

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