similar to: "Dropping Incompatible Voice Frame"

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

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, I have a new phone in our IP phone network: Planet VIP-101T. When calling from that Planet phone to anybody, everthing is fine. But when calling from anybody to that Planet phone, I get a mashine gun noise and the following msg in asterisk log: NOTICE[262161]: File channel.c, Line 1091 (ast_read): Dropping incompatible voice frame on H323:0 of format SLINR since our native format has

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

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906

I just bought a new IAXy box and am only achieving one way calling. Both iax.conf and the IAXy support ulaw and gsm. When I try to call, however i get this error: Feb 11 15:20:32 NOTICE[7963]: channel.c:1893 ast_read: Dropping incompatible voice frame on IAX2/iaxy-2 of format ilbc since our native format has changed to ulaw I added support for ilbc on the IAXy and in iax.conf and am still

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

Howdy, folks. I'm having a problem finding a way to auto-record calls (both incoming and outgoing). I know how to make it so either party can initiate recording, but I want it done as soon as both ends are connected (or prior to that if that's what it takes). It's probably right in front of me and I'm just missing it. Any help would be much appreciated. Thanks, Jay

Hey, Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related. [May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw) -------------- next part -------------- An HTML attachment

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

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 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

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",

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have no audio on chan_mobile but this message repeats continuously: Dropping incompatible voice frame on Mobile/DNA-54f4 of format slin since our native format has changed to 0x0 (nothing) Can somebody point me to the right direction? Asterisk SVN-branch-1.6.2-r313579 - -Stefan - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\

How do I include a file (not a context) in AEL? #include "filename" returns an error. Thanks, Jay

Greetings, List. I have my Asterisk box setup with 8 Centrex lines that were "left over" from our old PBX system. My boss is asking me to set up Asterisk so that he can flash hook and make an outgoing call on the same line to have a 3 way call. This is what he wants to do: 1) Incoming call on his Centrex line 2) Flash hook and dial a new number (goes out the same line) 3) Flash

I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay

Greetings, folks. I'm having a problem with blind transfers. It seems that, despite not having the T flag set, callers are able to use the blind transfer option. Scenario is this: - Asterisk 1.2.14 - Caller calls into our call center on one of our many phone numbers. - Call gets placed into queue. - Operator answers call. - Caller is able to hit our blind xfer key sequence (#0) and dial

Ok, I searched the lists and found no definitive answer. I'm assuming the IAXy has some primitive form of echo cancel, is there anyway to adjust this? Or any ideas on what to do instead. Here's the setup, this will not be a typical setup for our company however, well whatever. Anyway it looks like this: IAXy --> Netgear router --> Charter --> Level3 --> Asterisk --> VOIP or