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Hi All; Anyone can advise for a good IP Phone that has the ability to support SIP firmware and IAX firmware? Ofcourse, SIP there is a lot, but we need also the ability to use IAX (as it is good for NAT). Any advise. Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo!

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

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Dears; To patch libpri: I just place the patch file in the libpri source directory and then I run make and make install? Or I need to compile the dahdi and asterisk also? If the problem stayed, do I have to go for previous libpri version? Or for previous dahdi version and asterisk version? Regards Bilal ----------- > bilal ghayyad wrote: > > But I am afraid it is a bug because I

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 ==== Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Olivier 334 Philipp Kempgen 251 Gordon Henderson 242 Atis Lezdins 239 Jay R. Ashworth 230 Doug Lytle 207

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

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)

Dears; I am looking for the card that does not need an electrical power, which one? Is the PCI express doing this? Regards Bilal -------------------------- > While we're at it, can someone please tell me whether I > should be using > vi or emacs? ;-) > > Many thanks, > > Tom > > PS: Bilal: You have asked a nearly unanswerable question. > Some prefer >

Hi Danny; Really I did not understand how I can determine if the IO will be DB or conf files? Is it from the Asterisk manager? Regards Bilal ------------------------- It depends on how you are configured. The gui interfaces using Asterisk Manager, so you get the Same IO from the gui that you would get from a native manager session. -----Original Message----- From: asterisk-users-bounces at

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

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

Dear; In the extensions. conf, I have the following: exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}@Internal) So, I am writing the arguements of the Voicemail ( ) wrong? Regards Bilal > > Dear; > > > > Where I can find a new documentation for Asterisk > 1.8? > > > > Where is the wrong in that line? I see it is as 1.8 > version ! > > > >

Dear Doug; But I am afraid it is a bug because I read something this in the below link: https://issues.asterisk.org/view.php?id=17270 But maybe this was for old driver .. again, I am afraid if it is a bug. DAHDI Version: 2.4.1 libpri-1.4.11.5 Any advise if the below message is a bug? [Jun 15 16:14:00] WARNING[2773]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using

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

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,