similar to: G729 and Meetme

I was playing with a Sprint Vision phone recently and noticed when viewing the low level ReadyLink configuration screens that there are references to SIP registrars and the like. Does anyone happen to know if Sprint's implementation of ReadyLink truly is SIP based, and if so, managed to get it to interoperate with Asterisk. If so, it would prove to be an interesting paging mechanism and

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code)

HI ALL; I found in the following page: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing 1-If I could record all IVR promts in G729 format 2-If I could record voicemail in g279 format with """format_g729.c""""" then I donot need any g729 license (I suppose all my clients have g729 ip phones) My question is, how

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for using g729 codec. The post was

I haven't seen anybody so far post a complex fix for the udev problems on FC3 with the latest kernel. On that note, I have a temporary fix to allow zaptel to load somewhat normally. I found that modifying the zaptel script to 1) load, unload, then load the driver modules and 2) insert a pause between modules seems to allow things to work. This assumes you have followed the instructions

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Thanks

Are their any issues with Asterisk and Redhat Enterprise? I have see one or two posts with issues concerning compiling zaptel drivers but that is about it. Just looking for some consensus to if any problems exist with it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/56d03b72/attachment.htm

Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten => 9,1,Dial(Zap/g1/<CELLPHONENUMBER> where <CELLPHONENUMBER> is the number it is calling out to. When option 9 is

does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk

Hi folks! Based upon your experience on the field what wifi sip phone would you reccomend ? A customer asked for a wireless * install and I'm looking for advice, tnx Alessio Focardi [[*] - Interconnessioni Italy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/05b2fb30/attachment.htm

Hello everyone, One more little problem with a %100 g729 setup. Native moh: musiconhold.conf: [default] mode=files directory=/mnt/kd/moh/default random=yes ; Play the files in a random order ls /mnt/kd/moh/default fpm-calm-river.g729 fpm-calm-river.ulaw fpm-sunshine.g729 fpm-sunshine.ulaw fpm-world-mix.g729 fpm-world-mix.ulaw Place a call on hold: Jun 1 14:55:30

Hi, How difficult is to setup and maintain an Asterisk PBX with phones from multiple vendors? Is it even worth considering or is it safer to pick one vendor for phones and stick with them? I am more concerned about proprietary DHCP extensions, firmware upgrades etc..If anyone has any thoughts or experiences they would like to share I would be more than happy to hear from them. Thanks -Ravi

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have

Greetings, I have seen a few postings in the past regarding the interop of Asterisk and the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to getting the phone working. Assuming someone has this actually working, can that person step up and answer these questions. 1) What Channel is it working with (chan_skinny or chan_sccp)? 2) If code was used that is not a part of a

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a