similar to: Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)

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)

Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed number]@context-5c3e,2 of format ulaw since our native format has changed to slin Can anyone provide an English translation of what this means? The extension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw

I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've encountered a problem playing back a .wav file to an Ekiga client: My dialplan looks like: exten => 730,1,answer exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign) exten => 730,n,hangup Sovereign.wav is a .wav file that plays nicely on my 1.4 server. Here is what the console displays:

I have a SIP phone calling an AGI application. It starts out this way: -- Executing [s at macro-Call-AGI:2] AGI("SIP/151-b414f0c8", "computer-temp.sh,darwin,") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame

Hi, I'm having problems with the rxFax app. One of the messages that appear in my console is: Executing Set("SIP/something", "FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif") in new stack -- Executing RxFAX("SIP/something", "/var/spool/asterisk-fax/1137692307.5.tif") in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please

(alternative title - what did I do wrong? or suggestions to make this work) Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb /usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48: error: ? does not name a type ) 1.6 did compile and almost works. 'cept it thinks the .gsm files are not played. from

Hi all, After upgrading to lates CVS head, I have problems using a IAXY device, having slin problems: Jun 15 18:59:31 NOTICE[8197]: channel.c:1475 ast_read: Dropping incompatible voice frame on IAX2/lise-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

Hello, I have the following setup: (*)<--->IP<--->Micronet 5012 H.323 box <---> POTS <---> PBX (Alcatel OmniPCX) Grand idea is to use the micronet's POTS interfaces to connect SIP phones to the PBX and to the PSTN. I think i even managed my way in the arcane and cryptic management interface of that appliance, but I am stuck against theese messages: -- Executing

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of

Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have

Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try to playback something I get the following error: **[WOOMERA]** HW DTMF supported s1c1- -- Executing [number at from-pstn:1] Answer("WOOMERA/g1/1-7b29", "") in new stack **[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29 -- Executing [number at from-pstn:2] Playback("WOOMERA/g1/1-7b29",

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

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 Does anyone have the same problem, or know the solution? Multiple warning messages on Asterisk 1.4.36: Dropping incompatible voice frame on Local/.... when receiving calls with codec A and doing multiple attended transfers to codec B Reproduced with the following channel combinations SIP -> SIP -> SIP... IAX -> SIP -> SIP... DAHDI -> SIP -> SIP.. Tested in different

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

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

Hi, When I call forward on PAP2, the incoming call will right the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI> produce continuous errors as following: Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible vo ice