April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

question about the conference module: DTMF collection via XMPP?
by Brian Davis 24 Apr '08

24 Apr '08

I am actually not yet a user of FreeSWITCH, but I have been reading the online documentation to determine whether or not I could use it as part of a project I am currently working on that requires conferencing. I see that there are caller controls that can be used to play audio files to specific participants in the conference, using XMPP. I need to play an audio file and then be notified if/when the participant acknowledges the prompt in the audio file by pressing a DTMF key. I do not see a way to do this according to the documentation, but maybe there is something I am missing (or something not documented). Do any of you out there know of a way to collect DTMF over XMPP? Thanks, Brian

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Call for IVRs
by Brian Snipes 24 Apr '08

24 Apr '08

I would like to ask anyone that has working IVRs to please post them to the list or on the wiki ( http://wiki.freeswitch.org ). If you post to the list I will add them to the wiki for you. The few items on the wiki now were created no less than 11 months ago. There have been a lot of questions regarding IVRs and if you have benefited either from the list or from IRC please give back by sharing and helping others learn from your examples. Thanks, Brian

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Question about socket client
by Pete Kay 24 Apr '08

24 Apr '08

Hi, I have tried out the following command from Telnet: api originate sofia/default/1005 10000 &park Content-Type: api/response Content-Length: 41 +OK 8dc747d4-1261-11dd-92a8-2750a3aa164c SendMsg 8dc747d4-1261-11dd-92a8-2750a3aa164c Call-command: execute execute-app-name: fifo execute-app-arg:myqueue out nowait /tmp/announce.wav /usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav Content-Type: command/reply Reply-Text: -ERR invalid session id [8dc747d4-1261-11dd-92a8-2750a3aa164c] I tried to send a command to a uuid that is returned from the "originate" command, but it giving ERR. So, which uuid should I be using? Thank you in advance for your kind inputs. Regards, Pete

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Variable Manipulation
by Luis Jimenez 24 Apr '08

24 Apr '08

Hi, is there any way to manipulate variables in the dialplan? functions like cut or strip, replace, etc...

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recording callers separately
by jonathan augenstine 23 Apr '08

23 Apr '08

I have an application were I need to record the callers separately. I know this can be done on Aculab hardware but I would rather use Freeswitch. The requirements are that for 2 or more callers record each caller unmixed with the audio from the other callers. Can I do this on Freeswitch? Jonathan

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Calling Queue Implementation
by Pete Kay 23 Apr '08

23 Apr '08

Hi, I am in the processing of implementing a typical scenario for service call. So, in my scenario, I would have several agents who may log in and log out of the queue at different times of the day. People can call in and will be routed to one of the available agents. If no agent responses within 1 mins, then they would be asked to call back again. I checked the fifo mod, and I see how I can pack and unpark a user in the queue. However, how would I be able to route call to the agent automatically? Does it mean the agent have to initiate by pressing a # key and then have the dialplan to fetch the first user from the queue (using fifo out ) ? How is it possible for the agent's phone to ring automatically when a user is added to the queue? If someone can show me with an example, it will be greatly appreciated. Regards, Pete

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Wiki documentation (multiple-registrations)
by Luis Jimenez 23 Apr '08

23 Apr '08

Hi, i think there is a bug in the wiky docs, they say you have to specify the multiple-registrations param in sofia.conf, but it does not work that way, in the sample config it is set in sip_profiles/default.xml and that way it works. Luis Jimenez

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voicemail problem
by Pete Kay 23 Apr '08

23 Apr '08

Hi, I am having a voicemail problem: Problem 1 In the wiki, it says that with the below line, the user can check vm without needing to authenticate: <action application="voicemail" data="auth default 1005"/> Instead, it goes to the mailbox asking user to record message: [/usr/local/freeswitch/sounds/en/us/callie/voicemail/8000/vm-record_message.wav] rate 8000hz How come it is not going to the vm? Problem 2 If I change to <action application="voicemail" data="check default 1005"/> it asks me to enter user id follow by #. So, I enter 1005#, and it just hangs up. The curl xml created by fs shows: <document type="freeswitch/xml"> <section name="directory" description="Directory For FreeSwitch"> <domain name=""> <user id=""> <params> <param name="password" value="1234"/> </params> <variables> <variable name="user_context" value="agent"/> </variables> </user> </domain> </section> </document> Since I am using the user 1005 to dial in, so userid and password should work, otherwise, I won't be able to login to dial the number. Any inputs will be greatly appreciated. Regards, Pete

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Segmentation fault (core dumped)
by Luis Jimenez 22 Apr '08

22 Apr '08

If a start FS and reboot a snom that is not previously registered, but is configured, when the phone send the unregister message, FS shows "Segmentation fault (core dumped)" ane exit. it is reporducible. Luis Jimenez

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PDD about 2-3 seconds
by Luis Jimenez 22 Apr '08

22 Apr '08

Hi, i installed FS and configured 2 IP Phones 1000 and 1001, when i call from 1000 to 1001 or backwards, i get too high Post Dial Delay, about 2-3 seconds, any help/sugestion on this? PS: with openser and Asterisk there is not appreciable PDD. Luis Jimenez

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