similar to: Transcoding

Hi all Asterisk Gurus. I am really badly in need of help. Asterisk is very lovely software, but has one big disadvantage.. lack of documentation.But let's get to the point. 1. Is it normal that I get such a crappy quality with iax, some drops and clicks? Could anyone with some similar setup check my quality and say if this is what the people are so excited about? ( I used to work as a speech

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

Hi, I'm currently building an asterisk system which should work as gateway between SIP phones and ISDN. Most parts are working very fine, but one problem occurs and I am not able to solve or debug it. Telephony from ISDN to SIP (a Sipura Hardphone) is working very well, but if the SIP Phone initiates the call, the ISDN phone rings, and a connection can be established. But no one of the two

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]:

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN

Hi all, I''m playing with Xen for the first time, and encounter problems when installing a client. We basically have limited space on the /var/ partition, and Xen seems to use this as temporary storage during installation. Is there anyway of configuring it to use some other location? Cheers, Dean. _______________________________________________ Xen-users mailing list

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials. Here is part of the content of sip.conf. [general] port = 5061 bindaddr = *.IP

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

Hi, Want to do some experiments with the G.723 codecs - where can I download the 723 source code for Asterisk? I know there are some ongoing discussion regarding patents and license fees for the g.723 but I have some hardware on which I only have the 723 and need to test it privately. Thanks! Dan _________________________________________________________________ Use MSN Messenger to send

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 --

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No

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

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable