- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to make scheduled calls ?
by hung nguyen 05 May '10

05 May '10

Hi list. I want config FS to make automatic calls follow a fixed schedule. Example: In fixed time in the future, FS will call a fixed user and play a fixed music file for this user. I don't want to use crontab to do this because i want to deploy large schedules. Thank to suggestions.

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Sangoma prid and multiple PRIs with FreeSWITCH
by Bryan Vyhmeister 05 May '10

05 May '10

I setup a FreeSWITCH pbx with a Sangoma A102D card, the latest FreeSWITCH as of yesterday, and Wanpipe 3.5.11. I have found that I cannot use both PRI ports with Sangoma's smg_prid. The first one always works great but the second one never works. I get the following error: 2010-05-05 09:42:02.573743 [CRIT] ozmod_sangoma_boost.c:257 SPAN is not online. 2010-05-05 09:42:02.573743 [ERR] switch_ivr_originate.c:2480 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] When I start FreeSWITCH up I get this error on the console right after all the channels are activated. 2010-05-05 07:25:59.414444 [ERR] ozmod_sangoma_boost.c:1534 Error: Opening MCON Socket [-1] Address already in use The configs are below: openzap.conf [span wanpipe 1] name => 1 trunk_type =>t1 b-channel => 1:1-23 [span wanpipe 2] name => 2 trunk_type =>t1 b-channel => 2:1-23 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="1"> <param name="node" value="cpe"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> <span id="2"> <param name="node" value="cpe"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </boost_spans> <analog_spans> </analog_spans> </configuration> I'm dialing with this dial string. <action application="bridge" data="{origination_caller_id_number=${outbound_caller_id_number}}openzap/2/a/$1"/> I am certain the rest of this config works fine because the other PRI works great. I have also attempted to use FreeSWITCH without Sangoma's smg_prid by specifying a d-channel in openzap.conf. That seems to work but calls tend to drop frequently. I would rather just use Sangoma's smg_prid. I also attempted a hybrid config where FreeSWITCH worked without smg_prid on the first PRI and the second PRI was handled by smg_prid. That config worked except for the dropped calls that happen with using FreeSWITCH to manage the d-channel. My question is how can I get two PRI's working with Sangoma's smg_prid? Thank you. Bryan

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External SIP profile
by Kurt Ward 05 May '10

05 May '10

Maybe this is a bit of an uneducated noob question (or misunderstanding about SIP) but here goes: If I have an external profile set up and registering, does it essentially behave like a soft phone would behave? What I am running into is incoming calls work fine, but outgoing calls always produce a 404. If I use a soft phone using the same registration I can make calls with no problem (in both scenarios the calls are to/from 4 digit extensions on a Mitel PBX).

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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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