similar to: Incompatible voice frame ulaw/alaw

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-

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 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

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)

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

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

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

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 *>*

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

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)" and the call hangs up.  Why? Last minute thought: Is it possible that

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri

> 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

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! Could you help me with Asterisk 11.21.2 and AsteriskNow platform. The problem is: My Asterisk PBX has SIP (chan_sip) trunk to provider. Asterisk periodically loses trunk registratrion: *sip show registry:* /Host??????????????????????????????????? dnsmgr Username?????? Refresh State??????????????? Reg.Time???????????????? // //X.X.X.X:5060??????????????????? N????? <LOGIN>

I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

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, 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