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Dynamic Conference with voicemail fialover
by Ash 11 Apr '10

11 Apr '10

Hi All, I am trying to setup a system that will allow me to make an external call when somebody calls into my FS server. Once the two calls are connected I would like it to play file that says something like "you are connected". I have this component working by using a dynamic conference bridge. The next part is what I am having trouble with is making the caller go to a voicemail if there is no answer on the bridge line. This is the dialplan I am trying: <extension name="9000"> <condition field="destination_number" expression="^9000$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=true"/> <action application="answer"/> <action application="playback" data="/tmp/pleasewait.wav"/> <action application="set" data="call_timeout=5"/> <action application="ring_ready"/> <action application="conference" data="bridge:_uuid_@conf_1:sofia/gateway/primary/614xxxxxxxxxx"/> <action application="log" data="INFO Voicemail"/> <action application="voicemail" data="default v2e 1000"/> </condition> </extension> I am using FreeSWITCH Version 1.0.5-20100401-0400 as my build. Is it possible to send a call to the voicemail application or transfer the call to another dialplan if there is no answer on the bridge line? Cheers, Ash.

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Dynamic Conference with voicemail failover
by Ash 10 Apr '10

10 Apr '10

Hi All, I have exhausted every possible avenue with this issue and am not sure if I am approaching this the right way. I am trying to configure a system that allows me to join two callers together and announce to both parties that the call has been connected. I have this part working by using a dynamic bridge conference. The part that I am having an issue getting to work is to have it go to a voicemail if there is no answer on the external bridge. This is the dialplan I am using: <extension name="9000"> <condition field="destination_number" expression="^9000$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ignore_early_media=true"/> <action application="answer"/> <action application="set" data="call_timeout=5"/> <action application="ring_ready"/> <action application="conference" data="bridge:_uuid_@conf_1:sofia/gateway/primary/614xxxxxxxxx"/> <action application="voicemail" data="default v2e 1000"/> </condition> </extension> What I find is that the call is made via the above bridge statement however if I do no answer the call it will timeout as per the call_timeout variable but it does not continue on to the voicemail extension 1000, it will just hangup the call. Looking at the debugs it appears the mod_conference actually makes the call and therefore doesn't know what to do if there is no answer, is there a feature that would allow me to transfer the person calling into either a voicemail or another dialplan? Thanks in advance. Ash.

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An issue when attended transfer to fs
by CHU, XINGJUN (XINGJUN) 10 Apr '10

10 Apr '10

Hi, I have Freeswitch and other two SIP sets all registered with a SIP proxy/registry, and I am writing a control software to control the freeswitch via event socket. Baisically the control software tells freeswitch when someone calls it, what action to take, for example, acts as Auto attendant and play prompt. Etc. Now I got a problem when A calls B then attended transfer B to freeswitch, The prompt B hear is not from the beginning, it actually hears the trailing part of the prompt played to A when A called freeswitch in the beginning of the transfer. I understand that's because nobody tell freeswitch to stop and start from the beginning. I am looking for what event should the control software monitor for the replaced session (the transfer is done via "invite replaces" ) and how to cancel the current prompt and start from beginning. Any suggestions are greatly appreciated. Thanks Xingjun

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Getting git updates
by Lars Zeb 09 Apr '10

09 Apr '10

Once we pull down the full FreeSWITCH version using git, should we continue to use ‘make current’ to update thereafter? I started with this command and interrupted it just after it had pulled the source changes. I was surprised to see that “fs_cli” no longer worked. I went to freeswitch/bin and it was gone. Are the contents of this directory removed in ‘make current’? I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe to remove these? They are about 200M each. Thanks, Lars

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Getting git updates
by Lars Zeb 09 Apr '10

09 Apr '10

Once we pull down the full FreeSWITCH version using git, should we continue to use 'make current' to update thereafter? I started with this command and interrupted it just after it had pulled the source changes. I was surprised to see that "fs_cli" no longer worked. I went to freeswitch/bin and it was gone. Are the contents of this directory removed in 'make current'? I also saw core files (core.pid?) in the freeswitch/bin folder. Is it safe to remove these? They are about 200M each. Thanks, Lars

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move on from asterisk, the Random or rand function
by Krzysztof Drewicz 09 Apr '10

09 Apr '10

Hi, i'm slowly, but constantly moving from asterisk to fs. Stuck on translating this into a freeswitch: exten => s,1,Random(90:s,3) exten => s,2,Noop(his is hit 10% times) exten => s,3,Dial(SIP/8000) What i need to do is put a bucket on some calling_number + destination numer with some random percent to a voiceguide and hangup. i've found that memcache is very convinent way to put data into dialplan logic in runtime, so i plan to do: <extension name="bucket_part_01" continue="true"> <condition field="${do_bucket}" expression=".+" break="on-true"/> <condition field="caller_id_number" expression="^123456789$"> <action data="do_bucket=${memcache(get b_${caller_id_number})}" application="set"/> <action data="bucket_part_02" application="transfer"/> <action application="hangup"/> </condition> </extension> So above i check for caller_id_number in the memchache b_123456789 with option 1 Then i transfer to anoter extension: <extension name="bucket_part_02"> <condition field="destination_number" expression="bucket_part_02"/> <condition field="${czy_robic}" expression="^1$"> <action data="bypass_media=true" application="set"/> <action data="hits=${memcache(increment hit_bucket_test)}" application="set"/> AND HERE I do need some logic that does if Random() > 50% <action application="answer"/> <action data="/usr/local/freeswitch/sounds/pc/weRbusyRightNow.wav" application="playback"/> <action application="hangup"/> </condition> </extension> any cleaner approach or someone has done it better? note: don't want to use any lua, api or sth, just plain xml dialplan logic.

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git other question
by Jean-Yves F. Barbier 09 Apr '10

09 Apr '10

Hi list, About the contrib tree, shall I build it out FS tree or inside? -- Turn the other cheek. -- Jesus Christ

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git question
by Jean-Yves F. Barbier 09 Apr '10

09 Apr '10

Hi list, I now use git but I'm not sure about things I've got to do after an upgrade: is it mandatory to re-do the whole cycle (bootstrap + configure), or just configure, or nothing at all but a make install? -- The best cure for insomnia is to get a lot of sleep. -W. C. Fields

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transcoding
by Christian Hiller - Baig Tel LTD 09 Apr '10

09 Apr '10

Hello, i am using a TC400B transcoding PCI card and the module mod_dahdi_codec seems to be loaded correctly, even i am getting freeswitch@internal> dahdi_transcode Using 0 encoders of a total of 92 available. Using 0 decoders of a total of 92 available. twice the message that i have encoders available. Now how can i use them? I want G723 calls transcoded to PCMA. I have set these variables. But when it comes to a call, it negoiates the codecs and fails. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729,G723"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU,G729"/> Any help apreciated. Its my first contact with freeswitch and i find it to be a pretty straightforward software. good job. thx chris

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Re: [Freeswitch-users] compile errors
by Peter Olsson 09 Apr '10

09 Apr '10

A good start is to use a current version. Download 1.0.6 and try again. /Peter ________________________________ Från: srinivasula reddy <srinivas.ksvreddy(a)gmail.com> Skickat: den 9 april 2010 17:11 Till: freeswitch-users <FreeSWITCH-users(a)lists.freeswitch.org> Ämne: [Freeswitch-users] compile errors Hi, i am compiling the freeswitch1.0.2 its giving compilation errors in libsofia_sip_ua_static project. my enviroment is windows xp. any idea -- Srinivasula Reddy K !DSPAM:4bbf439232931558811689!

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