June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

playAndGetDigits incompatible with callback?
by Alex 20 Jun '12

20 Jun '12

I'm trying to use PlayAndGetDigits to get a sequence of digits, terminated with * (abort) or # (proceed). I thought I had it working by using an InputCallback, but it appears that PAGD turns off InputCallbacks. Is this really the case? Any other suggestions on how to get the result I need?

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Most Scalable Way To Control FreeSwitch.
by Joe Flemmings 20 Jun '12

20 Jun '12

What is the most scalable way to control FreeSwitch Routing dynamically (apart from XML configs) that has the least load on FreeSwitch and scales the best. This is to help achieve the most Call Setups Per Second (CPS) eg xml_rpc, lua, event socket etc Joe

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Bridge calls hang with GXW410x
by ocset 20 Jun '12

20 Jun '12

Hi I have asked this question before with no real solution and then I stumbled across this posting on the 3CX website, as they seem to have good support manuals for FXO gateways like the GXW410x. This is exactly the problem I am having. For those of you with a lot more in-depth knowledge, how would this translate into FreeSWITCH speak? * ----------------------------------- ADDITIONAL NOTES:* When receiving an inbound call on the GXW4104/4108 which is then diverted to an external number via the 3CX configuration, if the GXW4104/8 is used for the diverted call the GXW will stop sending SIP signalling to 3CX. This is a known issue with the GXW. As a workaround, in the 3CX User Interface, edit the Grandstream device, go to the "Advanced" tab, and enable the "PBX delivers audio" option. Please note that applying this workaround will no longer allow inbound or outbound faxes to be delivered over the Grandstream device -- so you will need a separate dedicated device/line for delivering faxes. *----------------------------------- * Thanks in advance O

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transfer/bridge in a conference
by michal 19 Jun '12

19 Jun '12

What's the best way of implementing this scenario: In conferenceA there are user1 and user2, in conferenceB there is user3. How to create 3-way conference with all 3 users and then "transfer?" user2 to confereceB. User1 has to stay in conferenceA after the "transfer" so next call can be taken in conference A while user2 and user3 still talk in conference B. I have thought of creating a new conference C where all parties could talk and "lock" Conferences A and B till user 1 comes back to conference A and user 3 comes back to conference B (user 3 will hangup the call and conference C will be closed/finished) Hope this makes sense.. I am using JavaScrip/Node.js to develop this. All hints and help will be appreciated

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channel variables in show calls api
by awais nazir 19 Jun '12

19 Jun '12

Hello Guys Is it possible to do some customization of "show calls" api to show some more channel variables like accountcode? --waisee

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HA setup completed but nothing in the database.
by Grygoriy Dobrovolskyy 19 Jun '12

19 Jun '12

Hello list, i've completed HA setup with two FS boxes, connected them to the ODBC(mysql) and replicated sip_recovery table between servers, the problem i face is simple, i have nothing in there when i have calls. External profile has this option set: <param name="track-calls" value="true"/> <param name="odbc-dsn" value="fsw-cnf:fs-usr:2>sdfjeslkjfd4<gH(^s"/> switch.conf also the required options: <param name="core-db-dsn" value="fsw-cnf:fs-usr:sdfjeslkjfd4<gH(^s"/> FS connects fine and have no problems with that, calls, channels are in the database, so i am thinking that FS do not "track calls" because of my dialplan which is the following: <condition field="${sip_to_user}" expression="(1234567789).*$" break="never"> <param name="track-calls" value="true"/> <action inline="true" application="eval" data="URI=${sip_to_user}"/> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="{absolute_codec_string=PCMA}sofia/internal/1234569787(a)83.164.189.13"/> </condition> So what i do is simple, match the call, and bridge to external URI. I even added <param name="track-calls" value="true"/> to test just before bridge. Dear all please help me to figure this out, how to make FS track this calls ?

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Call Recording file format(s)
by Adelia C. 19 Jun '12

19 Jun '12

Please help with my call recording question. If you know of a good link/doc with this info, please pass it my way. How does call recording work on Freeswitch? Is the recording saved by default in a .wav file? If so, is it always saved as .wav or can I configure in what format I want to save the recording? I noticed that .wav, .gsm, mp3 and .ogg formats are supported (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls) Are recordings saved in different formats in real time or converted late? Are they converted at the first request and saved for future use? Is this design decision left to the FreeSwitch user - i.e. if converted later, do I need to implement the conversion myself? I scanned through some older questions and sox and lame are mentioned. My recordings will be around 800K in wave format (raw data uLaw 8b/8kz stereo) so I am not looking to save space but I am getting the request from the front end guys to provide the recording in .mp3 format. Thank you, A.C.

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Multicast RTP collector
by Ron McCarthy 19 Jun '12

19 Jun '12

Can freeswitch subscribe to a multicast group and when it gets a stream connect that to another channel? Basically looking for a RTP to SIP bridge, I know a little out of the ordinary :) So far ive been able to find nothing, but figured it's worth checking. Thanks

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Gtalk does not ring local phones when making outbound calls
by mtle 19 Jun '12

19 Jun '12

Hi,I have successfully compiled and installed FS from the latest git as at today. Everything seems OK. The only issue is that I do not get any rings at all with gtalk. I wonder if anyone have had this problem lately and how to fix that? Here's my dialplan. <extension name="gv_out"> <condition field="destination_number" expression="^1?(\d{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="ring_ready"/>   <action application="bridge" data="dingaling/gtalk/+1$1(a)voice.google.com"/> </condition></extension> Thanks.

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mod_syslog
by Ádám Koleszár 19 Jun '12

19 Jun '12

Hi, A have a little problem with the mod_syslog module. Freeswitch generates multiline log messages (for example: SDP or register messages) which syslog-ng OSE can't handle. The firs part of the message is sent correctly but missing the facility and ident from the other part of the message. Because of this the messages can not be filtered by ident (or facility) on server side and goes to wrong destination. My question is, have you ever noticed this kind of problem? Thank you for your help in advance. Br, Adam

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