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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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frame per packet
by Kenneth Taylor 27 Apr '11

27 Apr '11

Hi, I'm building a voip client for android phones over gsm and consider using FS as sever. In order to save cellular costs I want to put more than one frame per Rtp packet. Is there any easy solution to do it in FS? Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go to lower levels? I don't think I should do it in the codec, it may cause problems in the rtp sequence number and probably a lot more. TNX, Ken

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mod_callcenter and uuid-standby
by Peter P GMX 27 Apr '11

27 Apr '11

Hello, I am trying to use the mod_callcenter functionality. This works nicely so far so thank you to everybody involved for programming this nice module! But I am stuck somehow with uuid-standby. Can anybody explain how uuid-standby works? Another question: In the Asterisk based callcenter solution named "Vicidial", an agent can be held permanently in a conference, waiting for calls who are bridged to his uuid in the conference. Can this be haviour be done with mod_callcenter? Best regards Peter

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FreeSWITCH Conference Call Today - Safi Systems IVR Builder
by Michael Collins 27 Apr '11

27 Apr '11

Hello all! We are happy to have Zac Wolfe from Safi Systems joining us today to talk about the IVR building system they have created. There is a gotomeeting.comlink on the agenda page here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_27 We look forward to this discussion! -Michael

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