search for: g723

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 se...

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080424/b442d5af/attachment.htm

Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've a Cisco here and it works fine with G723, but not with my asterisk. The bandwitdh is very important, since we will have our extensions at home. I know that I have what I pay, but the phone works with cisco. Trying to use G723 or G729 Asterisk sa...

Can anyone recommend a codec pack with G723 for use under Vista? I have G723 prompts (about 70 prompts totaling 1MB) needing to be converted to G711 uLaw. I tried Audacity but it doesn't have G723 codecs. I tired some google found adware free tools and websites with no success in converting G723. It does appear the old Cool E...

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to...

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 applications I have tested except for Echo. The critical a...

Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

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&quot...

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 3001 3002 6999 3001 3000 10999 - - 40994 8000...

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the P...

Hello list, i have a strange problem iam using the ulaw,alaw and g729 codecs in sip.conf i have like this [general] disallow=all disallow=g723 allow=g729 allow=alaw allow=ulaw even though i am disabling the g723 any UA could able to connect to the system and then suddenly asterisk stops working gives segmentation fault and closing the process. in logs i have this messages Feb 26 16:14:51 WARNING[1076266144]: Cannot disallow unknown f...

...92.168.0.187:5060> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=- 0 0 IN IP4 192.168.0.187 s=- c=IN IP4 192.168.0.187 t=0 0 m=audio 1400 RTP/AVP 0 8 4 18 0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 telephone-event 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.187 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 0 Peer RTP is at po...

...stix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - - - - - - - - - - - - - - gsm - - 2 2 4001 2 1 2 - - - 4001...

...ine. but once we are using the G729 codec, the asterisk is not responding to hash transfer, ie, when we press "#" it does not detect it and says "transfer..", is this a problem with G729 codec? (for testing purposes we have bought licenses for 2 chs) this also happens with the G723 codec, that is it doesnt detect "#" once we press it. please help us with this situation Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030523/48068de6/attachment.htm

Hello, I just CVS'd today and now I'm getting these errors when I call one grandstream phone to another both using 711U: NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from G723 to ULAW NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to...

...RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 38.114.20.207:28442 Found description format G729 Found description format telephone-event Capabilities: us - 0x4(ULAW), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x0(EMPTY) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp: No compatible codecs! What throws me off is the description format of G729. They said they used to be sending in ULaw and G729, but then I had him turn off G729 all together. But this sip debu...

...rdano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Juan Salas Inviato: luned? 3 ottobre 2005 15.41 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users] codec g723 on Via C3 I have a VIA Samuel 2, I use the Intel primitives (g729) setting the Makefile to a 586 processor. Maybe you can test with this. Regards. Jsalas. -----Mensaje original----- De: Giordano Grandis [mailto:g.grandis@invidea.it] Enviado el: Monday, October 03, 2005 7:06 AM Pa...

...f someone can throw more light on this for me. Goksie -----Original Message----- From: asterisk-ss7-bounces@lists.digium.com [mailto:asterisk-ss7-bounces@lists.digium.com] On Behalf Of Kai Militzer Sent: Monday, March 27, 2006 3:19 PM To: asterisk-ss7@lists.digium.com Subject: Re: [asterisk-ss7] g723 or g729 on ss7 link Hi! > Is it possible to set codecs on ss7 link? No. E1s channels (which chan_ss7 uses as voice channels) can only use G711 alaw. > Or receiving call with g723 or g729 and forward the call to pstn via the ss7 > link. You can do transcoding on the asterisk machine t...

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

...ndStream SIP phones: ***************[1005]**************** IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***************[1004]**************** IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:<empty> SIP User ID:1004 Authenticate ID:1004 Authe...