similar to: Use the NEW ulaw/alaw codecs (slower, but cleaner)

Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed

Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be doing any encoding or decoding, all codecs should be passing through. Any

> From: "John Hughes" <john at calva.com> > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Sent: Thursday, May 14, 2020 2:10:45 AM > Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and > alaw; asterisk wants to translate g729 -> alaw. WHY? > I am having a

Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via

On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 > or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw > right?  In fact from the sip debug it looks like it does, but then I > get the dreaded "channel.c:5630 set_format: Unable to find a codec > translation path: (g729) -> (alaw)"

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

On Thu, May 14, 2020 at 11:31 AM John Hughes <john at calva.com> wrote: > On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 or > alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? > In fact from the sip debug it looks like it does, but then I get the > dreaded "channel.c:5630

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

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

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Does anyone have a Dial script or a hint on how I can dial 10000 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me.

The other end is sending g729 even though it was not negotiated. The other end should not do this and it usually seems that the other ends that do send g729. This was recently fixed. See https://issues.asterisk.org/jira/browse/ASTERISK-28139 Richard On Thu, May 14, 2020 at 1:11 AM John Hughes <john at calva.com> wrote: > I am having a problem with one of my callers who is using

Hi list! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hi everybody, got it from svn: dtmf_2833_1.pcap */asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN *>*

Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could

Is there a digium twitter 'user' to follow that only tweets important announcements and release information? If there is not, I think there should be. It would be highly utilitarian to get an SMS when there is an update to Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time with asteriskpbx tweets like: "Anyone trying anything cool with Asterisk over the

QUESTION: Who's in the wrong: I recently saw an example of a u-law file with a metadata header on the file. The asterisk playback function 'PLAYED' the ascii header values as if they were audio data, creating an audible 'click'. After realizing the click was coming from metadata (and fixing it), I became curious: Which is 'correct? In other words: 1. Is it

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send