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Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> > Con...

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get

...gium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asteri...

....com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX =20 Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> = wrote: Show us your sip.conf =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subjec...

...gium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asteri...

...Re: [asterisk-users] Hash Dial Pattern Problems Your interpretation is right own....very weird problem. The problem is when i dial #551212 there is absolutely no activity in the CLI. It is almost like there is a conflict somewhere. On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas <danny at debsinc.com> wrote: Ok. I'm confused. I was interpreting what you wrote to say that you are doing this: 1. pick up sip phone attached to pbx1 (1.2 box) 2. dial #5551212 3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box 4. 1.4 box should fall into _XXXXXXX and do DAHDI dial?...

...gium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asteri...

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

...gium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asteri...

...gium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asteri...

Hi all, I'm using ConfBridge within Asterisk 1.6.20 and want to record the conference, so I'd like to start the recording when the second user joins, so in the example below, for example, how can I get the current user count in ConfBridge 3000? [conferences] ;authenticated conference (ext C-O-N-F = 2663) exten => 2663,1,Answer same => n,Wait(1) same => n,Authenticate(143382)

Hi everybody, I' ve been following this list for a while now. Is there a way to detect the individual and cumulative s/n ratio values for the incoming calls in Asterisk or any other Call Center solution?... -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111130/d0d53c1f/attachment.htm>

Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the

...gium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <danny at debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asteri...

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(start) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:48) <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(end) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011 12-16 18:34:48)