June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problems with make current
by Nik Middleton 10 Jun '09

10 Jun '09

Hi Guys, Ran make current today, and am getting the following errors. I ran bootstrap and configure, but still get these messages. Any ideas ? Looks like I'm now missing some libraries Regards, configure: configuring in libs/pcre configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch --cache-file=/dev/null --srcdir=. configure: error: cannot find sources (pcre.h.in) in . configure: error: /bin/sh './configure.gnu' failed for libs/pcre

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ClueCon 2009 - Important Reminders
by Michael Collins 10 Jun '09

10 Jun '09

Friends, ClueCon 2009 is fast approaching! We are definitely looking forward to seeing everyone in Chicago this August. If you haven't finalized your plans to attend, please do so right away. Time is running out! The early-bird registration special of $499 per person will expire at the end of June - only 20 days away. Also, registration as a whole ends on July 21st, so don't delay. Lastly, be sure to book your room at the Chicago Wyndham. Try expedia.com to see what kind of deals are still available. Please call 877.742.CLUE to get registered today! -The ClueCon Team

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Clarification on start_dtmf_generate?
by Ben Jones 10 Jun '09

10 Jun '09

Hello all, The documentation has this to say on start_dtmf_generate: As an example, Placing this in the Dialplan prior to bridging a call, will allow a phone set to rfc2833 ( info ) to send DTMF tones (in-band ) out to the recipient (IVR) or Auto Attendants. Thus changing the outgoing routing from (info) to (in-band). From what I understand rfc2833 != info. Does this mean if a user is set for rfc2833 OR info that FS will generate inband tones to send out? Does it only generate tones for info? If I can get any clarification on this I'll be more than happy to update the documentation. Thanks, Ben J -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com

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Recording time limit?
by freeswitch-users＠digitaldan.com 10 Jun '09

10 Jun '09

Hi guys, I'm noticing that when recording calls (using session.recordFile in both lua and javascript) the call is disconnecting after 12 hours (exactly 12 hours). I'm still looking into our cisco gateways to see if they are doing the call clearing, but wanted to know if there were any timers on calls or specically on recording calls. I'm using svn rev 13471 on debian, the calls originate from a cisco 53xx media gateway and are using sip / g.711 ulaw. Thanks. Dan-

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Newbie Question wrt Originating calls
by Max Bridgewater 10 Jun '09

10 Jun '09

Hi, Getting to learn Freeswitch. So, please bear with me. Can somebody tells me how to do dial extension 1003 from the command line. I tried the following and the all failed originate 1003%192.168.10.103 originate 1003(a)192.168.10.103 originate 1003%192.168.10.103 & park() originate 1003(a)192.168.10.103 & park() Thanks, Max.

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mod_skypiax (Skype Endpoint and Trunk) on centos problems
by Giovanni Maruzzelli 10 Jun '09

10 Jun '09

Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

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how to determine who is hangup the call
by zigurds 10 Jun '09

10 Jun '09

Hi, How to determine from channel events, that are sent through event_socket, what party have terminated the call? If I call for example from 103 to 104 and in first time 104 hang up, but in second time 103 hang up, in both cases CHANNEL_HANGUP is sent at first to 104: Event-Name: CHANNEL_HANGUP Core-UUID: 8729f181-3325-4373-8030-537bc090e9d3 ... Event-Date-Timestamp: 1244634602734687 Event-Calling-File: switch_core_state_machine.c Event-Calling-Function: switch_core_session_run Event-Calling-Line-Number: 469 Hangup-Cause: NORMAL_CLEARING Channel-State: CS_HANGUP Channel-State-Number: 10 Channel-Name: sofia/Test/104 Unique-ID: 17cdd5b7-8949-44d4-bbb9-5d17a713811c Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered ... Thanks, Zigurds -- View this message in context: http://www.nabble.com/how-to-determine-who-is-hangup-the-call-tp23961888p23… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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mod_lcr for a-z rates
by Diego Viola 10 Jun '09

10 Jun '09

Hi everyone, I'm trying to write a calling card app with A-Z rates, and I plan to use mod_lcr for this case... the only thing I need mod_lcr to do for me is get the rate based on the destination number / prefix. Is there a way I could achieve this with mod_lcr? I seen the wiki page and the SQL examples, but the SQL examples does a lot more, so I was thinking if I could use a custom SQL query to only do what I need. Thanks, Diego

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Documentation error in dialplan XML?
by Lars Zeb 10 Jun '09

10 Jun '09

Is the closing of the condition element correct? I'm new at XML. <extension name="Test1_2"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="dialed_number=$1"/> </condition> <condition field="network_addr" expression="^192\.168\.1\.1$"/> <!-should the slash at the end of the element be there? --> <action application="bridge" data="sofia/profilename/${dialed_number}(a)192.168.2.2"/> </condition> </extension> Lars

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Few questions
by freeswitch list 10 Jun '09

10 Jun '09

1) How do you setup a gateway where the provider doesn't require a user name and password? For freeswitch gateways username and password are mandatory. 2) Is there anyway to have xml_curl send the password for directory entry requests. 3) What is the best way to monitor for failed sip register attempts? 4) Is there a way to increase the volume of voicemail message sent by email? They are fine in freeswitch but when sent by email as mp3 (mod_shout) they are really low. 5) Is there any future plans to make the voicemail module more customizable?

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