similar to: no reply to our critical packet

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

I can't login anymore... used to be able to. Timing doesn't seem to be working well any ideas? Also what is this "NOTICE" I'm getting? *CLI> == Accepting call on 'SIP/lenny-b19c' ("Lenny Tropiano" <5555>) -- Executing VoiceMailMain("SIP/lenny-b19c", "") in new stack == Parsing

I'm getting a FLOOD of these types of messages on my MAC OS X box: Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:43

> Hello all, > Can someone help me on the problem which I have on MGCP phone test . I test mgcp - asterisk- zap. But I got several NOTICE message from rtp.c. > NOTICE[20501]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support > incomplete. Turn off on client if possible > > -- Endpoint 'aaln/1@VG101-1-1' observed '9' > NOTICE[20501]: File rtp.c,

Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold: -- Executing [youngandson-test at

Hello all ! I have a problem with my IVR with terminate connection from PSTN phone Here is my configuration extension.conf [ivri] ;exten => s,1,Wait(1) exten => s,1,Answer ;exten => s,2,DigitTimeout(5) ;exten => s,3,ResponseTimeout(10) exten => ivr,1,Background(demo-congrats) exten => 1,1,MP3player,/mnt/linux/mp3/song/04.mp3 exten =>

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk. Oh323 Module compiled without errors. But When I try to stary Asterisk with the Oh323.so file in the modules folder, Asterisk is dying with the following error. [chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) ==

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hello All, We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the subscriber end. For long distance we have iax2 connectivity with a ip carrier. For local calls we are routing out through a commercial VEGA voicestream pots unit to an adtran channel bank and then from there to our class 5 soft switch. The sip to sip calls and the long distance calls work great.

Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =>1,1,WaitMusicOnHold(30) exten =>1,2,Dial(SIP/mateo,18) exten =>1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly,

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

Hi Folks, I'm currently trying to configure musiconhold (on a asterisk-1.4.17) for replaying a live mp3-stream (Icecast2). after reading the related material on voip-info and several other pages, I've successfully tried out mpg132, madplay and mplayer to pipe a stream into moh. however, there is one major problem involving some kind of buffer-issue. let me try to explain this problem

Hi, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing

Hi there, i had installed on all my servers the codec_g729b which is the old voiceage, so a month ago i updated the codec to codec_g729a. After that i started to get this message on my asterisk console: Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:264 process_rfc3389: RFC3389

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten