similar to: Integration with skype

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 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 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

I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Heh, well, I guess it's dead: http://www.digium.com/en/products/software/skype-for-asterisk If I have a really bad connection, can I "downgrade" SIP somehow? I

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

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 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

Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW?

Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal

Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine

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 All; Any one advise for open source prepaid billing other than A2Billing that can work with Asterisk and it is rich by features (for large business)? Regards Bilal

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 All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal

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>

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; I need to use a gateway that converts from SIM to E1 to I can send and receive calls via the GSM, so did any one use a good gateway for this and reliable and stable and costly effective, so he can advise us to use it? Also, it will be a separate product if we need also to use it for SMS (send and receive), also we need a reliable and constly effective product for this. Thanks in

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>