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--- Ralph Giles <giles@xiph.org> wrote: > On Fri, Sep 14, 2007 at 02:51:34PM -0700, Josh Coalson wrote: > > > checked in to CVS is what will be very close to the 1.2.1 release > > of flac scheduled for monday. if anyone can try building it and > > even better running the test suite, and reporting back any > problems, > > that will help me get things in

Dear all I have directory base dialing in asterisk but i want to improve it like Grouping, etc.... is there any more feature in it ? One more problem when i find number in diarectory by typing 3 digit number then it is possible it would say a digit like 5450 caz when i dialing by directory but next time i have to know that number it is long process to find name again

Hi List;

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

I got one email from eric asked me to Lower the rxgain and txgain on your Zap channels. But actually it is already the voice volume is low and I was looking to increase the gain (currently it is 0.0), so I do not know if eric was mean to reduce it less than 0.0, but I can not do that due to the low volume that is already existed, so any more reduce will make the voice not hearable well, even if

Hi List; I think my problem in Zaptel compilation is related to autoconf: no input file, anyone has an advise? Also, I did a change in the Makefile existed in the following path: /usr/src/kernels/2.6.20-1.2319.fc5-i686/ EXTRAVERSION = 2.6.20-1.2319.fc5 Now, if I run uname -r then I get output: 2.6.20-1.2319.fc5 But the directory under the kernels is: 2.6.20-1.2319.fc5-i686 So do I have to

about something else than the IP Trunk, he is talking about outbound (which is related to using an application to run an outside call, which is used usually in campaign in contact centers and so on), I think nthis case differs that placing a calls via IP Trunk or even outside call but the caller who will do it (and not the application). Lastly, Mr. Amit helped me when he gave me a configuration

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

Appreciate the kindly help and advise. Regards Bilal --------------------- > > Bilal, > > I suggest you turn on logging on your tftp server to see > what files are actually being requested, and if the the tftp > server is dishing them out... Try adding a few v's to your > tftp setup: > > File: /etc/xinetd.d/tftp > Line to change: server_args = -s /tftpboot -v

getting calls, but I can only send calls from my main machine IP address so I can't control where I am sending calls to. I am hoping to have this developped somehow (a per SIP peer bindaddr and bindport), even if it means some bounty. I can't imagine this being this difficult, so a few of us who need this putting a couple hundred dollar would probably do it. Mike > -----Original

Hi Guys, I need your recommendation which card to buy?The TE411P or TE406P do they have any difference?I check their brochures they are differ only when it comes to PCI slot voltages.Which card best to use?Btw,Il be using supermicro board with 3.3V and 5.5V?Will il be experiencing problem during the implementation later on if i choose the wrong PCI voltage options?Will it depends on

I discovered that when I need to do compilation, and when running make menuselect then no need to select add-on from the modules, the only this is to select chan_ooh323. Regards Bilal -------------- > > Hi; > > > > > > > > > > Those are not needed for ooh323 . > > > > > > In fact, chan_h323 won't build with them ATM. > There's a

Regards Bilal ------------------------- > El 18/07/11 18:03, bilal ghayyad escribi?: > > Dears; > > > > If I need to login using as agent using the > AddQueueMember(team,....) then what to be the second > paramter? How to be written? > > > > For example, if the agent id is 8000 then it will be: > > > > AddQueueMember(CustomerSupport,Agent/8000)

> -----Original Message----- > From: JP Carballo [mailto:lists@netfone2x.com] > Sent: Tuesday, January 24, 2006 7:32 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Anyone using verizon fios ftth for > analogvoice?Any echo? > > Chris Mason (Lists) wrote: > > > JP Carballo wrote: > > > >> I would

Thanks for answering my question guys :) Guess I'll have to try it and find out, but will keep everyone posted... Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Totaro Sent: Tuesday, January 24, 2006 7:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE:

replies we received but I did not receive from where I download it and how I compile it. Regards Bilal ____________________________________________________________________________________ Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php

retreive the variable and did not write it directly ${CONSOLE} as already CONSOLE is configured in the [global] or what is the storey :) - ? Regards, ---------- Bilal Ghayad ____________________________________________________________________________________ Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222

Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send

Howdi. I'm new to dovcot (just succeded in compiling on unixware 713 today) I'm noww configuring for tests on other port. The sample configuration file is pretty well documented, however, I don't understand how I can firstt authenticate against passwd/shadow if not found against pgsql. What's the syntax? Please cc responses as I'm subscribed to digest TIA Reghards --