similar to: ADPCM codec and IAXy device

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 everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

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 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 everyone, We are using the IAXy boxes and Asterisk over the internet and I was wondering if Asterisk can do a codec translation during a call in order to lower the bandwidth that the comunications consumes? I mean, the IAXy boxes only support the ADPCM and uLAW codecs, but for a certain number of calls our bandwidth runs out, then I think if Asterisk can convert the signal that comes in ADPCM

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

Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only

I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04:

iH i have an IAXy which i can make calls from but am unable to call. when i dial the extension assigned, i get the following from the console; -- Executing Dial("SIP/5801-b665", "IAX2/5899@192.168.0.5") in new stack -- Called 5899@192.168.0.5 -- Call accepted by 192.168.0.5 (format ULAW) Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected

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 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 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!

In the mailing list archives, I found a message that indicates that the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything official on Digium's site about it. The Installation Manual has an example iax.conf file that indicates the ulaw codec, so I know that one is good. But we are thinking about using the IAXy over a VPN, to replace our MultiVoip. alaw and ulaw are

Hi.. I have to setup an extension in a remote location that will use a cordless analog telephone.. I am looking at the IAXY to do this for me..Basically the data path will be as follows... [Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset Since there are two NAT boxes in the path I know SIP won't work.. I also don't want to move the Asterisk box to the internet side of the

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