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Hi All; To simplify the the login and logout for the agent, I am looking for the variable that can be used for the AddQueueMember (in the place of the ?????? as following: exten => 100,1,AddQueueMember(CustomerSupport,${????????},1) exten => 100,2,Playback(agent-loginok) exten => 101,1,RemoveQueueMember(CustomerSupport,${??????}) exten => 102,2,Playback(agent-loggedoff) In other

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 All; I noticed in the queues.conf the configuration for recording the calls in the queuing, and regarding to the filename (or any other parameter), it is written that I can determine the filename using the command: Set(MONITOR_FILENAME=foo) But it should be called from the dialing plan, but really i did not understand how to call it from the dialing plan. Well, for example this is my

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

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 >

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

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,

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

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

Dear List; For each call (in specific case), I need to do a record and save in a spearated file, so I am thinking the best thing is to save based on the time. Monitor(wav,Record1,m) So, how can I make the file name to be based on the current time (which is changed always, or based on the some unique paramter (related to the call it self). Any advise? Regards Bilal

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

Regards Bilal ---------------------- On Thu, 2008-06-12 at 01:23 -0700, bilal ghayyad wrote: > Where did u find a good IAX IP Phone? I've had good success with my Allnet IP-7960 phones. They have the ability in the firmware to either do SIP or IAX, and they even have a mode where you dial one prefix to send the call out using the SIP protocol, and another prefix to send the call out over

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

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

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

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

your sip communication altogether. Have you tried changing IP address of your asterisk server? If changing IP works, then probably your provider has blocked you sip communication by IP only. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-23 7:22 AM, "bilal ghayyad" <bilmar_gh at yahoo.com> wrote: Hi All; I have my friend that use his mobile (Nimbuz) to connect for the

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 ! > > > >