similar to: "Cannot write OGG/Opus streams. Sorry" - any ideas?

hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]:

Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [*40 at liberado15:15] Record("SIP/1201-083453c8", "/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3") in new stack ??? -- <SIP/1201-083453c8> Playing 'beep'

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft

On 13.9.0 https://github.com/traud/asterisk-opus The README: Alternatively, you can use the Makefile of this repository to create just the shared libraries of the modules. That way, you do not have to (re-) make your whole Asterisk. The Makefile generates: codecs/codec_opus_open_source.so formats/format_ogg_opus_open_source.so formats/format_vp8.so res/res_format_attr_opus.so Without any of

Before I got an log a ticket, can I just check I'm not doing anything wrong? In 15.2, to install Opus: 1) run `make menuselect` 2) Highlight "Codec Translators" and press enter. 3) Scroll down to "codec_opus" in the section labeled "External" 4) Press enter to select the codec if it is not already selected. ... at this point, I see XXX codec_opus and a

The Asterisk Development Team has announced the release of Asterisk 14.0.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722),

On 27 January 2018 at 09:27, Ludovic Gasc <gmludo at gmail.com> wrote: > Hi Jonathan, > > If you put the cursor on the line XXX, you will see what are the > dependencies are missing to enable the option. > In this case, it's certainly curl that is missing on your system. Ah, OK! No, it wasn't curl that was missing, but I think the way it was phrased confused me. It

Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday).

Adding gluster-users. On Wed, Jan 31, 2018 at 3:55 PM, Misak Khachatryan <kmisak at gmail.com> wrote: > Hi, > > here is the output from virt3 - problematic host: > > [root at virt3 ~]# gluster volume status > Status of volume: data > Gluster process TCP Port RDMA Port Online > Pid >

Hi all, Sorry for the repost. If I should be asking somewhere else please tell me where. I woke up a few days ago to find this on one of my machines. Below is hopefully the revelent output from dmesg and /var/log/messages. I have had this b4 but this is the first time I have been able to get any useful information from it. The machine usually locks and all I see is the assertion failure on

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Hi, Consider the following situation : <SIP/asterisk-0000001d>AGI Rx << WAIT FOR DIGIT 3000 <SIP/asterisk-0000001d>AGI Tx >> 200 result=48 <SIP/asterisk-0000001d>AGI Rx << WAIT FOR DIGIT 3000 <SIP/asterisk-0000001d>AGI Tx >> 200 result=48 <SIP/asterisk-0000001d>AGI Rx << WAIT FOR DIGIT 3000 <SIP/asterisk-0000001d>AGI Tx >>