similar to: How to write sound file with G723.1 codec or G729 codec

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.

Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec

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,

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

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

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 guys, This is my first post in this list. I'm newbie with Asterisk and VoIP technology. First, this is my scenario: - Asterisk server (asterisk-1.0.0) - Cisco router connected to PSTN I've got redirected incoming call from extension 532x from Cisco to Asterisk server. Asterisk server H.323 channel receive the routed call from Cisco, and it launch the following dial plan:

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 --

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hi there, I am using redhat 9.0 with asterisk 1.0.7. I created an user and was be able to leave voice messages to that user and retrieve the voice message. I looked the wiki and setup the voice message as the email attachment. However, I have never received email with the voice attachment. Here is the setting for voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats

Hi, I try to use Speex codec into Win32 platforms. However, I find the CPU bandwidth usage is very heavy on a Pentium 3 machine. Compare to Microsoft's G723.1 codec, speex 8k is using more than 20% cpu bandwidth. Does anyone know what is the best version of Speex to "beat" the Microsoft's G723.1's on CPU bandwidth usage? Does Speex have MMX-enabled codes? thanks very

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

Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" <sip:000BEA801CA6@hcs.net:5060>' failed for '204.194.36.138' The telephone LCD says "SIP registation

Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

Hi again, always latest CVS from 27/06/04. Calling to a SIP gateway I receive: Unable to find a path from G723 to ALAW Unable to find a path from ULAW to G723 Asked to transmit frame type 4, while native format is 1 (read/write = 8/4) Unable to forward voice [last messages repeated lot of times] Acked pending invite 102 <- My phone number ... No path to translate from SIP/... to SIP/... Had

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm Has anyone else seen this? Thanks,

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan