[Freeswitch-users] Problem while playing more than 10 voice files using playback

Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: = Set the playback_delimiter variable = Set the playback_sleep_val

[Freeswitch-users] Using odbc in FS core

Hello Folks, I'm interesting in completely moving away from sqlite and use postgresql everywhere including core ( switch_core.c ) All other applications can use odbc without issues (sofia, limit, fifo etc), but as I see in core only sqlite3 supported. I correctly set 'core-db-dsn'

Re: [Freeswitch-users] Using odbc in FS core

Hello, Looks like the issue is not in multi statements in one request. Manually creating DB schema helped and everything started up. I will continue testing Also in code I see such construction: switch_cache_db_execute_sql(dbh, begin;delete from channels where hostname='';delete from

Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated

Sorry for the extended forum thread on this subject - This really IS the last post ! I have now got the ATA to work without the dialplan fix provided by Michael. After I'd implemented the fix, I had more of an idea of what the problem was and was better able to go through the Polycom VPA-11

Re: [Freeswitch-users] Problems with Voicemail

I installed all sounds from SVN, but usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA isn't there. I checked another, older installation and couldn't this file either. I think that freeswitch tries to build a sound path for the file to be played, and some parts of the path are missing. I

Re: [Freeswitch-users] tcp call misses sip message

Attached is graphical representation of SIP message flow. You can see that for some reason FS doesn't resend to callee an ACK message recieved from caller. Regards, RobertT attachment: TCP FS SIP msgs.PNG___ FreeSWITCH-users mailing list

Re: [Freeswitch-users] tcp call misses sip message

Well since we aren't a proxy we wouldn't resend the one we receive... what svn rev and are you using proxy media? /b On Nov 21, 2009, at 7:28 AM, RobertT wrote: Attached is graphical representation of SIP message flow. You can see that for some reason FS doesn't resend to callee an ACK

Re: [Freeswitch-users] Using odbc in FS core

we had the code slightly out of order, you should update to latest trunk for the right version. The test of 2 deletes is to see if your odbc driver will fail when trying to execute 2 statements at once so I can properly fail over to sqlite because transactions are mandatory for a database running

Re: [Freeswitch-users] Problem while playing more than 10 voice files using playback

cant you use the execute_complete events to tell when your playback is done or var is set? On Sat, Nov 21, 2009 at 3:22 AM, Thangappan.M thangappan...@gmail.comwrote: Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have

[Freeswitch-users] Help Freeswitch with Voipuser Gateway

I need help as I cannot receive calls through VOIPUSER. This is a learning setup Attached are my conf files. What is wrong with them ? When I dial from a landline I get a continuous beep. Attached are my gateway and the conf file to transfer. Sopfia Status is my screen message. I can see a

Re: [Freeswitch-users] tcp call misses sip message

Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org

Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated

On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson steve...@primrosebank.netwrote: Sorry for the extended forum thread on this subject - This really IS the last post ! I have now got the ATA to work without the dialplan fix provided by Michael. After I'd implemented the fix, I had more of an

[Freeswitch-users] IP1001 Setup

-Original Message- From: Michael Collins Sent: Nov 21, 2009 4:51 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Analog phone with ATA. Phone Won't Dial Out, but can receive calls - Updated On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson

[Freeswitch-users] ATA that supports TLS/SRTP w FS

HI All, Has anyone got some recommendations on which ATA to buy that supports TLS and SRTP? Thanks! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] How do I know the destination profile name?

Thanks Mike! However, this doesn't fully solve my problem. When using sofia_contact() indeed it works ok with finding the destination's profile. However, it breaks the BLFs... When calling *sofia/sip_profile/local-user%local-do**main* the BLF works ok. When calling