search for: transcode_via_sln

...STN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110628/d741ba6e/attachment.htm>

...memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.org/r/622/) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9 Thank you for your continued support of Asterisk!

...memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.org/r/622/) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9 Thank you for your continued support of Asterisk!

Hi, I have an Asterisk box which has an IVR that plays random gsm files. The box has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPU usage along with the load seems to jump around. With about 500 callers it hovers between 250-400% CPU (so 2.5 to 4 cores) which seems reasonable. Every so often the load average spikes. The idle never drops below 85%. When the load average

Try setting transcode_via_sln=no in /etc/asterisk/asterisk.conf and restart Asterisk. A reload will NOT apply the new value. Setting it to no seems to smooth out CPU usage on one of my servers. On 4/22/20 2:01 PM, Dovid Bender wrote: > Hi, > > I have an Asterisk box which has an IVR that plays random gsm files....

...-1 of format slin since our native format has changed to ulaw Because of that outside caller can't ear the callee on the IAXY. Found somewhere that disabling transcode in asterisk.conf would fix the problem, so I added, not sure of the syntax the following section in asterisk.conf: [options] transcode_via_sln=no That didn't work, and I am not sure I am using the wright syntax... I have revert back to stable release and everything is ok, but would like to give latest release a try... Regards, Francois Random Thought: --------------- Democracy is more cruel than wars or tyrants. - Seneca, Epistu...

I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as

...llowing channel combinations SIP -> SIP -> SIP... IAX -> SIP -> SIP... DAHDI -> SIP -> SIP.. Tested in different systems that I've upgraded from 1.4.22 to 1.4.36, tested with different codecs. g729 -> alaw ilbc -> alaw alaw -> ilbc alaw -> g729 ... Tried to set transcode_via_sln=no and restarting *, it has no effect. moh files in wav format, also tried moh files in multiple formats to avoid transcoding.. =============== Test Multiple call transfer =============== A with codec X calls B with codec Y B answers the call B attended Transfer to C -> Ok C attended Trans...

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

...ems to be a small gap of 2 or 3 not sent packets when playing several files one after another. At the moment I'm using Asterisk 1.8 Certified (Cert 5). In asterisk.conf I've set: highpriority = yes languageprefix = yes internal_timing = yes defaultlanguage = de transmit_silence = yes transcode_via_sln = yes documentation_language = en_US "core show settings" says: PBX Core settings ----------------- Version: 1.8.15-cert5 Build Options: DONT_OPTIMIZE, LOADABLE_MODULES, BUILD_NATIVE, G711_NEW_ALGORITHM, G711_REDUCED_BRANCHING, TEST_CODING_TABLES...

...|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xc (ulaw|alaw)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x38000c (ulaw|alaw|h263|h263p|h264) In asterisk.conf we even activate transcode_via_sln = yes ;Build transcode paths via SLINEAR,instead of directly. Why is Asterisk trying to read messages in slin format? Thanks for any hint. -- Daniel