similar to: G729 codec problems

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Date: Fri, 10 Sep 2004 16:37:33 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4141AE1D.3020403@inaccessnetworks.com> > Content-Type: text/plain; charset=us-ascii;

Would someone kindly share some definition into the following? Openphone version 1.91 includes dual sets of Speex codec's starting with: SpeexNarrow-5.95k{sw} SpeexNarrow-5.95k{Xiph} Through SpeexNarrow-18.2k{sw} SpeexNarrow-18.2k{Xiph} I do not understand what the differences are between {sw} & {Xiph} given the same bit rate for both? Are all of these Narrow or Wide or Ultrawide

I'm trying to use a Howlink CL-100 ip phone with * It's h323 phone with very limited protocol support. But it's enough that I can use it to dial netmeeting client and artisoft pbx just fine. When I try to dial my * with it using either chan_h323 or oh323, it seems to fail on negotiating H245. Maybe this phone doesn't support it? I've used all different versions of

Hi guys, I got 2 licenses of g.729 and while running the asterisk with Monitor (for recording a channel) and using one channel for the call... I receive this error: WARNING[23826]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! many times.... it starts only when the call through the Zap channel takes place. while this error is being running on my screen I ran the cli command:

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not

how to make g.729 preferred, but failover to gsm I've purchased a few g.729 licences, and would like to set up iax.conf such that g.729 is used if they are available, but then it fails over to gsm. I'm not sure how to specify such a preference. I'll let the server transcode from ulaw (from the sip phones) to g.729. Got plenty of CPU for the number of phones we run off that

Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI> exit No such command 'exit' (type 'help' for help) *CLI> quit No such command 'quit' (type 'help' for help) *CLI> Any other ideas? I started asterisk with -cvvvvg option. Same problem if use asterisk -r to connect. Can not exit. Any

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hi, all I finally got myself one of those SIPURA boxes. It is labeled as Linksys, but this is actually a SP3000 box. Anyway, unit has lots of configuration parameters. Not all are obvious. At the moment it registers against my *, but all the calls I do from analog phone connected to it, go to VoIP channel. As this part is still in testing, I want all the outgoing calls got to PSTN by default

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

Hello, I've recently installed Asterisk@home, i'm following step by step the "new user guide" but I cannot get my X-Lite SIP phone see my asterisk@home proxy... I've installed in a viertual machine (vmware) and there's some problems with the Zaptel service and I think that this is why I cannot connect. Thanks in advance. Fabrizzio Valencia -------------- next part

>From: speex-dev@xiph.org >To: wolfkharl@hotmail.com >Subject: Please confirm your message >Date: Fri, 06 Jun 2003 07:08:06 -0400 > >Hello, this is the mailing list anti-spam filter at Xiph.Org. >We need you to confirm your e-mail message with the subject of >"Adaptivity". > >Please send a message to the following address, or simply use your >mailer's

Hi all, I am currently running asterisk (CVS HEAD) on a p4 machine with rh7.3 and using nufone's h323 channel driver. I was using the old voiceage g729 and have replaced it with the digiums g729. I used to have the message "Measured length exceeds frame length" but since I changed the g729 codec I have not encountered this problem. However, another peculiar problem has cropped

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23