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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 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; 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

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 All; Did anyone try to use IAX IP Phone behind NAT, and let it receive calls from Asterisk without doing port mapping at the router existed at the site where the IAX IP Phone existed? Is the need just to let the IAX IP Phone that is NATed to register on the Asterisk and at asterisk I set nat=yes for the IAX client configuration? Or it is impossible to let the NATed IAX to receive calls

Hello everyone! Well i want to know, if there is a way to specify a physical nic (like eth0 or eth1) to a domU and how can i do it. The server has two nics, what i want to do is assign physical nic (eth0) to the dom0 and assign physical nic (eht1) to the domU. I appreciate your help. Regards Ivan ____________________________________________________________________________________ Be a

I've checked the samba code (sessetup) and found out that samba always send spnego packet when Extended Security capability is ON. (This can't be turned off/on ie. "use spnego = false") Does it mean I can "never" connect to smb server that doesn't support spnego if Extended Security is supported by server? Is my understanding correct?

Hi Jean-Marc I am using speex in my voip app (wideband mode). I have noticed that when VAD kicks in, CPU considerably increases (bitrate decreases to 4 kbps). It takes about 7 ms to encode one frame versus 0.5 ms (for speech). This happens regardless preprocessor is used. Although the issue seems to be driver dependent, it occurs on XP, Vista and Mac OS X. Besides complexity and quality, this

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!

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug.

Hi I have been using pygrub for most of the needs. Was trying out some stuff on SLES 10 SP1. It seems that it does not have pygrub!! On the other hand,they have domUloader.py which requires bootargs="--entry=hda:/boot/vmlinuz...., path to ramdisk. It is quite strange to be asking for the path of the kernel and the ramdisk to be supplied in the conf file!!. Am I missing something

Hi I have been using pygrub for most of the needs. Was trying out some stuff on SLES 10 SP1. It seems that it does not have pygrub!! On the other hand,they have domUloader.py which requires bootargs="--entry=hda:/boot/vmlinuz...., path to ramdisk. It is quite strange to be asking for the path of the kernel and the ramdisk to be supplied in the conf file!!. Am I missing something

Hi Jeff. I read that you use PCI Passthrough and works well, i was wonder if you can tell me how you do that ''cause i can''t find how to do it :( and i really need to. So if you can give some advice about how you do it would be really helpful. I appreciate your help Jeff. Regards Ivan ____________________________________________________________________________________

I have a game that is not working under wine (it just seem to hang up with a black screen, though when I try to close the program, confirmation dialog for quitting appears). Output solely says it's executing an interrupt with unimplemented parameters in wine: err:int21:INT21_Ioctl_Block int21: unknown/not implemented parameters: int21: AX 440d, BX 001a, CX 0848, DX fbac, SI fc10, DI fbec, DS

Hi, I'm doing an upgrade to the latest samba 3.0.28, and something's bugging me: on the old server I had this working OK: root preexec = /usr/local/bin/ntlogon.py --user=%U --os=%m --dir=/home/samba/netlogon/ && /usr/local/bin/pdc-log.sh %U %m %a %d On 3.0.28, I can only get the first script to run, everything after && is ignored. I have tried to put just the second

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

Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * AT&T/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

Are shorewall-shell and shorewall-common required at compile time even if one only wishes to use shorewall-perl (4.0.9)? ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ