similar to: g711 ulaw vs alaw

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

Hello, I'd like to use g729 pass-thru when I dial out to a sip provider from my IP phone but because I have no license for g729 I'd like to use g711 ulaw for asterisk voicemail, conference bridge and other services. When I set in [general] section of sip.conf the following: disalow=all allow=g729 allow=ulaw the g279 pass-thru works fine with my SIP provider but when I call the

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 Edit (now Adobe Audition 3.0 for $349USD) can do it -jason

Ok, I'm quite fond of CVS-Stable 10-26-04 as it's always been fine. One thing I noticed with this version and all versions prior, when I did a "sip show channels" it always displayed info in all caps. But sometime between 10-26-04 and 12-8-04 they changed this to all lower case. I believe this MAY be related to the latest problem I just fixed. My provider was sending me

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

> 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

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, 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

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-

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

In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'. "Use the NEW ulaw/alaw codec's (slower, but cleaner)" By slower does this mean more 'expensive', or does it instead mean that there will be more algorithmic latency? Both? Can anyone speak to the relative increases? With regard to accuracy, can anyone speak to what kind of situation might

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 *>*

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

Dear helpers I constructed a data frame with this structure > str(dados1) `data.frame': 485 obs. of 16 variables: $ Emissor : int 1 1 1 1 1 1 1 1 1 1 ... $ Marisca.Rio : int 1 1 1 1 1 1 1 1 1 1 ... $ Per?odo : int 1 1 1 1 1 1 1 1 1 1 ... $ Reproducao : int 3 3 3 3 3 3 3 3 3 3 ... $ Estacao : int 2 2 2 2 2 2 2 2 2 2 ... $ X30cm : int