similar to: codec selection based on call prefix

Hi all, I'm trying to set up a test case for an ISP featuring an asterisk server and a couple of linksys rt31p2-na routers registering on it. Instead of using dsl lines, i'm trying to plug the * server and the routers on a cisco switch, just to test their functionality. I have created a vlan and a subnet on the switch and set up the ip addresses of the routers in that subnet. When i plug

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

Hi all, I have installed a TDM22B on an IBM xSeries 220 with FedoraCore 3 and setup asterisk cvs head to work properly. A few days ago, I tried to update the system kernel to 2.6.12 from 2.6.9 and also to change to asterisk stable 1.0.9. After compiling zaptel and asterisk, i loaded zaptel and tried to load the wctdm module but this failed with the following message: Notice: Configuration file

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

(Note: resending with proper Subject) If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Can Meetme be made to work with G.729? (I gather not) If a call comes in (internally or externally), the call comes in as a G.729 call, which then re-negotiates to a G.711u call when if gets transferred to a MeetMe room. Is there a way to set up asterisk that will allow me to have internal phones renegotiate to G.711, with the external lines instead transcoding within asterisk. (runtime is more

Firstly, I'm running Asterisk 1.4.4 on Solaris 10. I have several different internal SIP phones all sharing a single IAX2 VoIP channel. PHONES |------------- <SIP/uLAW> --------------| ASTERISK |-------------- <IAX2/g729> ------------|VoIP/ISP The g729 codec has been registered successfully and appears to be detected by Asterisk (NOTE: I have changed what I thought might have

We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones. I have 2 1 sip phone that's outside in the "world", and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

Has anyone here ever had the occasion to setup a system that would dynamically alter it's codec preferences based on trafffic? That is, presuming that the system is on a limited bandwidth connection is would start to prefer a compressed codec as the call volume increased? Perhaps shifting from G.711 to G.729? Michael -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org

I have 2 Asterisk servers connected with IAX. It's working fine I can call an extension from one phone in an office to another phone in the other office. The only problem I have is lagging. What codec should I use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I configured it to disallow all and use GSM only. In my sip config of each phone I use disallow all and allow

Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that

What is the default multicast codec for multicast in Asterisk 13 ? G.729 or G.711 or other ? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200401/b83fc8ca/attachment.html>

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

Hello, All! We was upgrade our Asterisk from version 0.7.2 to 1.0.7. And have big problem. When asterisk starts: ------------------------------------------------------- *CLI> h.323 show codecs Allowed Codecs: Table: G.729A{sw} <1> G.729{sw} <2> G.723.1{sw} <3> G.711-uLaw-64k{sw} <4> Set: 0: 0: G.729A{sw} <1> G.729{sw}

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

Here is the scenario... SIP UA's can use either GSM or G.711 ( in that order of preference in the sip.conf ).. Asterisk Server1 is linked to Asterisk Server2 via IAX2 and also supports GSM and G.711 ( also in that order of preference).. 1. If a call comes in from the UA using GSM and then goes out over the IAX2 leg, Will Asterisk simply move the GSM encoded data from the SIP channel to