similar to: Music oh hold

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

Hi, I installed mpg123 v0.59r without error and defined as defaut folder /var/lib/asterisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8 *CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack -- Called 2391 -- SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for

The procedure that will do asterisk is very nice ;) but whe it was available ? Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. Here my sip.conf for the phone, can u say me if there is somethingh wrong ? [2391] type=friend username=2391 secret=2391 language=it host=dynamic context=intern dtmfmode=rfc2833

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

The wiki says this about the MusicOnHold command: "Plays hold music specified by class. If omitted, the default music source for the channel will be used." http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold How do I set the default music on hold class for the SIP channel ? I tried adding musiconhold=test to my sip.conf. musiconhold.conf looks like this:

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to

I have googled this to death, and all I get are reference to the MoH needing a Zaptel timing source, and then people saying no they don't any more. -- Set Response Timeout to 2 -- Executing BackGround("SIP/211-57ba", "my-greeting") in new stack -- Playing 'my-greeting' (language 'en') == CDR updated on SIP/211-57ba -- Executing

Hi all, I've compiled zaptel drivers and reconfigure asterisk afterwards from source --with-zaptel. Modules are loaded accordingly: asterisk-1.4.2 # lsmod |grep z Module Size Used by ztdummy 5472 0 zaptel 194504 5 ztdummy crc_ccitt 3521 1 zaptel my musiconhold.conf: asterisk-1.4.2 # grep -v '^;'

Ok, but is there a way to check if echo cancellation is active on a call in progress ? Thanks Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Steve Davies Inviato: marted? 28 marzo 2006 16.43 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Echo cancellation

I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 12 gennaio 2006 17.20 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users]

Hi all, i'm tryinf to install chan_capi but i get this error root@obelix:/usr/src/chan_capi-0.3.5# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO

Hi, just a question: anyone has never installed g729 codec on VIA motherboard with C3 processor ? I'm having problem with IPP libraries, and Intel said that it works only on Inter processor. Any suggestion? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading

Hi all, I'm looking for a PCI card which i could install on asterisk box, with purpose to use 15-20 cordless dect phone in a very "dect cell". Is there anyone that could help me pls ? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL:

I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Alexander Lopez Inviato: venerd? 16 settembre 2005 17.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: RE: [Asterisk-Users] direct sip call

Just the sort of newbie question we all hate ;-) I'm a bit stuck with MOH. I think all is done right and I've read everyhing I can find, but whenever * tries to do MOH, all that happens is '-z: No such file or directory' Yes, I am on redHat. Yes I have installed real mpg123. Yes, it does seem to work from the command line. Any suggestions would be greta, I'm sire

Hi group, anyone can explain me the exact difference between pri value in zapata.conf ? ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN If I use it, I also must use prilocaldialplan = local ? Thanks Giordano -------------- next

I've been trying to play the default music on hold file, but no luck yet. here is my configuration: extensions.conf [incoming] exten => s,1,Dial,Zap/2|10 exten => s,2,Voicemail,u34 exten => s,102,Voicemail,b34 exten => 34,1,SetMusicOnHold,default Musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random =>

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press