similar to: Dropping incompatible voice frame

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

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

Hi. I have newely installed vicidial now i am getting thise error anyone can hel me. NOTICE[31819]: channel.c:1972 ast_read: Dropping incompatible voice frame on Local/8600051 at default-ed7b,1 of format gsm since our native format has changed to slin Regard's Vijay Kumar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our

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

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

Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

Hi all, I have problems with rxfax application. It seems to be ok but I don't receive the fax in my directory. My extension.conf is as follow: [macro-fax] exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/incoming/${UNIQUEID}.tif) exten => s,2,rxfax(${FAXFILE}) [fax] exten => 100,1,macro(fax) [reception] exten =>s,1,Answer() exten =>s,2,Background(00) exten

I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2

I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5

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

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

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at