similar to: Convert wav-file to alaw-file

> 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)"

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

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)

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 I am having some problem forcing my tdm410 to alaw over ulaw, I have 1.6.1.0 asterisk (debian i486) dahdi 1:2.2.0 built with the hardware echo canceller firmware /etc/asterisk/chan_dahdi.conf alaw=1-4 but I have this in the general section, before any channel definition dahdi show channel 1 Default law: ulaw even when i have a call going it is still ulaw Thanks Alex

I have installed G729 but I cannot make a outgoing call with it. SIP/dennis-2c23 is making progress passing it to SIP/1010-8b60 NOTICE[311316]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from G729A to ALAW NOTICE[311316]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[311316]: File codec_gsm.c, Line 136

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

Hi, We have a e405p with a external Euro-isdn PRI-ISDN net interface from Telco connected. We tried to send a fax to another machine with a TDM400P. We use IAX2 with G711-alaw codec. Both fax machines connect, but have error in transfer. We use asterisk CVS-02/01/04. Which can be the problem ?. What can I do to find the problem ? Thanks. Regards, Pedro

Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 -> asterisk -> DIAX :: sound is ok. When I perform an outbound call: DIAX -> Asterisk -> h323 :: sound is terrible and CPU load is 80% When I

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, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec

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

Does ooh323 from asterisk-addons 1.2.1 support alaw codec? This is what is written in h323.conf.sample that can be found in asterisk-addons dir. The codecs to be used for all clients.Only ulaw and gsm supported as of now. Default - ulaw ONLY ulaw, gsm, g729 and g7231 supported as of now disallow=all allow=gsm allow=ulaw So, it shouldn't support alaw, but I manage to establish calls with

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-

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

I'm having issues with T.38 on Asterisk 1.6.2.8. A few lines are received OK and then I get only garbage. I'm using ReceiveFAX() provided by app_fax to receive the faxes. After talking to the engineers on the telco, they said Asterisk is sending a REINVITE to alaw after the T.38 reception is complete, and that could be the cause of the problems. I personally am not totally convinced of

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