similar to: autoanswer

Hello; Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131218/7fbbc3c8/attachment.html>

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,

Hello; If I have load up to 220 extensions with 50 concurrent calls. Can one hardware server carry all this load? What is the hardware server required for this? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130714/b63756d2/attachment.htm>

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

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

Hello; Is there a method "way" to be able to dial the phone number through asterisk from the outlook email contact? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140128/17174762/attachment.html>

Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

Hello; Is it possible to configure in the sip.conf for the Phone to be auto answer? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130717/b8e0dc7f/attachment.htm>

Hi All; Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal

Hi List; In the outbound, I read in the documents the Wildcard match "by using the . (period)", but I did not understand how Wildcard will work (like what)? As I know that Wildcard is a term used with the Diguim TDM card (FXO and FXS), so what is the relation between such cards and the matching in the dial plan? Any help? Regards Bilal

Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

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

Hello; I am looking for SIM adaptor to be connected with Asterisk to be able to send and receive calls from the mobile operator and if possible the same adapter to be used for SMS "sending and receiving". But what if anyone called this SIM card that is connected to this adapter and no one relied his call, how this miss call can reach for the use at the asterisk PBX? Regards Bilal

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

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

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

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed