May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call for assistance: Spanish sound files, translations
by Michael Collins 05 May '10

05 May '10

¡Hola mis amigos! We could use some help with the Spanish sound files that Carlos Reyna has provided. What we need the most is someone to get the latest phrase_es.xml file and compare it to the list of sound files that we actually have recorded. We need a list of sounds that are missing. Also, we need someone to get the latest phrase_en.xml and compare it to phrase_es.xml. There are many new English sound prompts that aren't even in the other language phrase files. We'd like to rectify that. Additionally, if you are able to translate the phrase_en.xml file into other languages that would be very welcome. It would be good to get the existing non-English files updated. Please contact me off list if you are able to assist with any of the items listed above. Thanks, Michael

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Re: [Freeswitch-users] ** Feature Code
by Jerry Richards 05 May '10

05 May '10

Sorry there was a typo in my scenario. Here is the correction: 1) 5382 calls 5403. 2) 5403 answers and two-way talk path. 3) 5381 dials **5403 4) 5403 loses call. 5) 5381 and 5382 two-way talk path. Thanks, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Wednesday, May 05, 2010 8:17 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: ** Feature Code As you know, dialing **<number> picks up a ringing call (i.e. Directed Call Pickup). However, I noticed yesterday that it will also take a call away from another phone even after it is already connected. Here is the scenario: 1) 5382 calls 5403. 2) 5403 answers and two-way talk path. 3) 5381 dials **5403 4) 5403 loses call. 5) 5382 and 5382 two-way talk path. Is there a way to keep the Directed Call Pickup feature, but prevent taking over the call after connect? Thanks And Best Regards, Jerry

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DTMF events
by Madovsky 05 May '10

05 May '10

Hi, is it possible to use min dtmf at 300 ? it's only to avoid to record again some dtmf files I did for a phone keypad I currently trying to develop Thanks Franck

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How to monitor find out which events are currently monitored, set only a particular group of events?
by Frank Church 05 May '10

05 May '10

Is there a way of getting of list the event types that are being monitored by your connection to Freeswitch, other than those keeping a record of those you've added from your own end? I want to add some without deleting what is present with the first event plain XXXX or filter Event-Name XXXX command -- Frank Church ======================= http://devblog.brahmancreations.com

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FreeSWITCH Community Conference Call May 5
by Michael Collins 05 May '10

05 May '10

Hello all! Our weekly conference call will be starting in about 90 minutes. (1pm Eastern/10am Pacific) The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_05 I have only a few things to discuss so please bring your questions and thoughts. Multilingual community members are especially needed today so that we can talk about translating phrase files. If you have any other things to discuss please by all means bring them. Rupa is traveling this week and won't be available this week for one of his killer presentations. :) Thanks everyone and hope to talk to you in a few... -Michael

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** Feature Code
by Jerry Richards 05 May '10

05 May '10

As you know, dialing **<number> picks up a ringing call (i.e. Directed Call Pickup). However, I noticed yesterday that it will also take a call away from another phone even after it is already connected. Here is the scenario: 1) 5382 calls 5403. 2) 5403 answers and two-way talk path. 3) 5381 dials **5403 4) 5403 loses call. 5) 5382 and 5382 two-way talk path. Is there a way to keep the Directed Call Pickup feature, but prevent taking over the call after connect? Thanks And Best Regards, Jerry

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Different codecs for different devices?
by Ken Fulmer 05 May '10

05 May '10

This may be a far out question but here goes: Our fax machines that are connected to ATA's need to be g.711 on both Call Leg A and Call Leg B (in either direction). However, our phones need to be g.711 on the internal side that faces our PBX. On the side facing our SIP provider, the calls should be transcoded to g.729. Inbound the calls to phones (not ATA's) need to be g.729 on the inbound call leg and g.711 on the outbound call leg. I realize we can set inbound / outbound codec preferences in single or multiple profiles and we can set the "absolute_codec_string" value for the outbound call leg. However, I've yet to determine how to set the inbound call leg's codec differently depending on the device. Is this possible? Thanks, Ken Fulmer

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Method to force a transfer of A-Leg
by Steven Ward 05 May '10

05 May '10

Hello all, I have a lua script running that checks the state of a call between A and B - the call between A and B was set up through a Polycom's attended transfer feature, so A itself didn't necessarily execute the bridge to B. A gets early media from B. After a certain amount of time, if A is still getting early media, I want the script to end that call between A and B, and send A through some specific dialplan. How does uuid_transfer work for this goal? I'd like to kill B and move A to specific dialplan. I don't want B to be in the transfer at all. I know there may be many possibilities here; I'm just wondering if anyone can recommend something. Thanks. - Steve

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mod_xml_cdr error log is not working
by Frank Church 05 May '10

05 May '10

I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs are not working. The system is a windows system and I wonder if the defaults for windows are different. I have logs/xml_cdr in addition to the log/xml_cdr in the c:\freeswitch directory but Freeswitch can't find them. Logs snippet ========= 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1811(a)192.168.1.133) Running State Change CS_REPORTING 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/1811(a)192.168.1.133) State REPORTING 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0] posting to web server [http://192.168.1.20:8132/] 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with url [http://192.168.1.20:8132/] 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to web server, writing to file 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file or directory] -- Frank Church ======================= http://devblog.brahmancreations.com

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Annex B
by Ken Fulmer 05 May '10

05 May '10

When we send calls to an Adtran gateway, our calls using g.729 work properly. I noticed in the SDP the following field: a=fmtp:18 annexb=yes When we send calls to a Cisco gateway, the same call fails. Inbound g.729 calls coming from the Cisco gateway work properly and I noticed the following: a=fmtp:18 annexb=no Is there a way to set this parameter in the dial-plan or sip profiles so we can match what the Cisco gateway is looking for? Thanks, Ken Fulmer

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