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Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the

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,

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

We're using 2.2.13 with pigeonhole 0.4.3, in a clustered environment (maildir on netapp, dual dovecot instances where each server is both a proxy and a backend). Every now and then (once a month per user, maybe?), users will see a blank email in their inbox. Investigating further, and we will see that the only information recorded in the maildir file for the message is the Return-Path,

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

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

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

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

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)

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

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

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 >

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to

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 >

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 ! > > > >

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

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

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