similar to: IAX and NAT Transparency

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 All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

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 List; Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo!

Hi All; Is there any limitation on the number of users for MeetMe Conference? In other words, how many parties can join to the room and become a member of the room? Is there any limitation? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

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

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

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

Hi All; I was using since one year Asterisk 1.4.19.2 and zaptel 1.4.10.1 and they were working fine via SIP, IAX and Digium fxo and fxs ports. Suddenly just before 2 or 3 days, the voice become garbage like robot when I place a call from the SIP Phone (which is in a country and the Asterisk box in another country). I am surprise what is the reason that let rtp become like this ! The sound now

Hi All; I am using Background in my configuration, and I noticed the following so if any can help: 1) If I pressed 1 twice (11), so it runs the step related to first 1 and then it runs the step related to second 1, so it does buffering for my input and run two steps, how can I make it run only the step related to first entered digit "1" and does not do buffering (so ignoring the second

Hi All; I installed one digium card of 2 fxo and 2 fxs, but the following problems related to the voice are happening: 1) Sometimes when I call to the PBX, I hear like modem sound and after little it disapear. 2) There is a disturbance in the background (like the channel radio disturbance that might happen if the frequency was not captured well), and that disturbance appear much more when

Hi all; I have two asterisk boxes installed in two separated sites, the internet bandwidth between them is very good and I am using G729 codec to communicate with them and IAX. The problem that side A hears well side B while side B does not hear well side A !! I did one thing in side B that in iax.conf, I set the bandwidth=high and it helped, but still side B is complaining from the quality

Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying "trabajo mensjes" would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton

HI Michal Thank you for explaining. I have this situation in a number of production servers where we would always use static IPs for the host and VMs. In such case we have no requirement for NATed network in the future. And we we ever do, we can rely on a DHCP server within the LAN to provide IPs to the VMs. I'll look to remove both libivirt-daemon-driver-network,

Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost.

Hi All; Anyone knows an IAX IP Phone works fine and tested? Does polycom support IAX IP Phone? Regards Bilal