similar to: Asterisk, Zaptel and the Kernal Compatibility Matrix

Hi All; We have a telephony service provider that is asking what is new technology and services to be added with the telephony service that can be used for VoIP and PBX purposes. Any suggestion to be added that can really give new advantages and technologies specially in VoIP issues? Anyone interested? Regards Bilal

Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr => true)? 2) If there relation between setting the vbr => true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value?

Hi All; I am using IAX Trunk and I used ddns (dyndns.org) with the host (host=xyz.dyndns.org), when I restart the router who has the hostname xyz.dyndns.org then its IP address change and updated, but at asterisk level still it keeps sending for the old IP address and sometimes this problem does not resolve until I restart asterisk. Any one faced this and has idea how to resolve it so Asterisk

Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed

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!

Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like

Hi All; If one of my FXS port damaged at TDM22B because we connected the Telephone Line cable to the FXS port while it should be connected to the FXO port, then can I order S110M FXS Module and fix it instead of the damaged FXS? (This if we assume my problem that really the FXS port damaged). Rregards Bilal

Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

Hi All; I am trying now to compile h323 to be able to use it, I did the pwlib and openh323 successfully and I exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need to compile h323 as following: cd /usr/src/asterisk-1.4/channels/h323 When I type make, it gives me: make: Nothing to be done for 'degault' And when I type make opt, it

Hi All; I need a help as the voicemail need GSM codec while I am using G729 for the call, why Asterisk does not do codec translation from G729 to GSM, it does not support? Any need for settings, what I am missing? Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search.

Hi All; My Asterisk has a public IP address, how can we let two IP Phones in different site and both are behind NAT (each one has a private IP address) to call each other? In other words, Assuming Asterisk IP Address is 193.111.194.111 IP Phone (A): 192.168.0.1 and its default gateway is: 195.226.225.70 IP Phone (B): 192.168.0.1 and its default gateway is: 185.200.200.80 How can IP Phone at

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

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

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.

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

Hi All; Anyone knows if there is an Asterisk version that setup can be stored in Database instead of the configuration files (.conf)? Any advise? Regards Bilal

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

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 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>