similar to: change moh during a call?

Hello. Is it possible to "mix" musiconhold music and playback voices? What i want to do is something like this: A person calls a number, gets a playback voice while in background music is playing. The configuration i use at the moment don't do what i want. Someone knows how to do it? Thanks in advance. exten => 18,1,Answer exten => 18,n,Background() exten =>

All, I am using Asterisk 1.4.4 and it is not playing any MOH. I think the underlying problem is the following error: [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/moh/asterisk' [Jun 11 20:25:37] WARNING[3207]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player Now it does not matter what I change in the

Hello. Someone knows what cause 127 mean. The phone that i'm calling rings once and than the connection interrupts: P[ 5] --> l3id:10040 P[ 5] --> cause:127 P[ 5] --> out_cause:127 P[ 5] --> state:ALERTING P[ 5] --> Channel: mISDN/5-1 hanguped new state:CLEANING P[ 5] $$$ CLEANUP CALLED pid:3 best regards -- Thomas Stein knowledgeTools? ....damit Sie sehen, was Sie

Hi. Asterisk 1.4.12.1. I get a lot of message like this. Someone knows what this message mean? Do i have to worry about it? [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Local/152 at local-f137,1' not posted [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Agent/152' not posted [Nov 7 15:24:25] NOTICE[31247]: cdr.c:434

Hello. I have a problem recieving fax without a callerid. Somehow the script i'm using fails and i don't know how to fix it. Does anyone have an idea how to solve this? Here an example of a working fax transmission: >fax2mail v2.0 > Triggered on Tuesday, May 29 2007, at 10:38 AM > $1 = CallerID number of fax sender = 02365207150 > $2 = CallerID name of fax sender = >

Hello. Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line. misdnportinfo gives (what does ":Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib" mean?): best regards and thanks t. asterix asterisk # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001

Hello. I have problems to integrate an isdn phone. I don't know why but the isdn phone rings only once and than it looses its connection to his base station. I can make a call from the isdn phone to an VoIP Phone inside my network but when i pick up the phone the isdn phone also crashes. misdn.conf: [ntport1] ports=5 context=isdn-telefon msns=* extensions.conf: exten =>

Hello. Does someone know what "order field" means in followme.conf? The Doku says: number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ] So an example would be: number=> 123&124&125,10,? It would be nice if someone could enlighten me. cheers t.

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music on hold isn't working like it used to. It seems that even though the correct MoH class is being set, the system still plays the "default" music. All of my call handling is done with an AGI script. When a call is made, the AGI script sets the MoH class before dialing. The log indicates that the correct

Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Thanks -- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files

I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? ~jay

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hi all, I'm trying to make some simple MOH (Music On Hold) working. So far I've failed miserably - so I turn here for help. Basically I've been using the wiki and all the sample confs I could from there and via google. The queue system seems to work fine with my limited setup. Just 2 IAX2 clients where I keep Client B busy (by making it listen to mp3 via ext. 777) but logged into

I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B

Thanks Jan. I've been trying to obtain an English copy of the Dovecot book for months, prior to starting this project. So far, I just can't find a copy. It's too bad that the author/publisher won't do a second printing or, if they're not interested in making any more money, then release it to the public domain as a PDF. Very frustrating. Michael > -----Original

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11