similar to: Musiconhold interrupted sound

Displaying 20 results from an estimated 1000 matches similar to: "Musiconhold interrupted sound"

2004 Jan 30
2
Music on Hold Warnings
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
2005 Sep 09
1
musiconhold errors in 1.2.0-beta1
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
2007 Apr 10
1
Maximum retries exceeded on transmission
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
2005 Mar 16
4
problem with musiconhold
Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]
2003 Dec 18
2
Zaprtc compile error - virtual device for conferencing
Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing
2003 Mar 02
0
Entering username/password (DTMF) from Cisco 7960/SIP in Voicemail touchy...
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
2003 Nov 27
4
RFC3389 support incomplete
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
2005 May 28
1
Quintum Tenor AXT800!
Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with
2006 May 23
1
Quintum Tenor DX 3020 problem to register on Asterisk
Hi, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ?
2003 May 22
0
MGCP NOTICE message and WARNING messsage
> 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,
2008 Jan 17
1
buffer-issue when piping live-streams into musiconhold
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
2008 Sep 14
9
Streaming MoH on 1.4
Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream =>
2005 May 30
0
asterisk integration with Quintum Tenor AXT800!
Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all just only IPphones connected through ehternet port and analog phones connected on FXS port.Is
2007 Feb 27
1
Quintum configuration ASM200 Analog 2 tenor port
Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with asterisk@home but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated.
2004 Jan 12
4
RFC3389 messages with ATA 186
I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!
2010 Oct 11
1
Quintum Tenor AX and Echo
Let's try this again. I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? <p style="margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif;
2003 Jul 04
1
IVR problem from PSTN phone
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 =>
2004 Jul 09
3
ATA 186, firmware SIP 3.1 and codec g.726
I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:
2007 Mar 08
4
Asterisk distributed deployment
Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:
2004 Oct 03
0
Tenor AS cancells calls through Asterisk
Hello, Maybe some of you tried the SIP support recently introduced by Quintum in their AS devices. I have one Asterisk machine connected to PSTN via E1. It works properly. On the other side I got an ADSL line, with NAT and few devices behind it, like computer with X-Lite client installed or mentioned Quintum device. It works great - calls initiated from there are OK, as well as PSTN originated