similar to: Codec translator problem (g723.1,ilbc => alaw)

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing

Hello, Would you please suggest me, where can I buy g723.1 liscence in cheap. I might need a liscence for 10-50 channels. Thanks,

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones,

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -

Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Hi, First of all I am not sure that what I am trying to do is correct/supported, but here is what I'm trying to test: Some of my endpoints only have g723 codecs. Because of this I am only allowing g723.1 codec in sip.conf and h323.conf. Calls between endpoints work fine. I am trying to configure voicemail and meetme applications. I see that all voice files in asterisk are in gsm format and

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 [Jun 2 17:08:28] WARNING[21652]:

Hi, I have an * server on a routable (public) IP address and a sip client behind NAT using a Grandstream phone. He is connected through a bi-directional satellite so he has a bit of latency involved. Usually I can dial this extension and them to me. But I keep getting a registration failed message. I have other sip clients not on a satellite and they don?t get these time outs. So I assumed it

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --