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Hi all, Here another guy working with ztdummy and having problems with music. MP3Player does not work for me both from a telephone entering through a passive ISDN-adapter as well as a SIP-client in the LAN. Ztdummy works with conferences. Here are my messages. I have seen several discussion threads in the mail-archive regarding this subject, but could not find an answer. Is this something I

Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer("IAX2/maxhome@maxhome/3", "") in new stack -- Executing

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Hi all, I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I could not use it to run a remote stream, if I use mpg123 in command line, I can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3 file could not be replay with asterisk. I would appreciate with any suggestion. Phuong Here are the log message: -- Starting simple switch on 'Zap/3-1'

Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten => 200,1,Answer() exten => 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten => 200,3,Dial(SIP/200|20|tTrR) exten => 200,4,Hangup() -- end -- here is debug from CLI: -- Executing

I am running CVS HEAD (on a Linux-PPC machine.) My current dialplan generates an error at the console in asterisk when I attempt to issue the MP3Player command -- I can't figure out why it's not playing the actual audio file? The rest of the dialplan works great. Here's what I see in the console: -- Executing MP3Player("IAX2/income-in-01@IP",

Hi, I'm running the latest Asterisk (built last Saturday) and can't get mp3's to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I don't hear any sound, and the following displays on the console:

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of

Hi there, app_mp3 still does not work with the latest bri-stuff patch and the zaphfc driver. Here in my place it only works with the patch attached. For me it seems the bri-stuff worsens the asterisk timing... has anybody else made experiences with it? Deti -------------- next part -------------- Index: app_mp3.c =================================================================== RCS file:

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Here is my Extensions.conf file (Default Context). When an individual calling in dials the extension, the response time seems very slow. It doesn't immediately go to the next step, but hangs out for a few seconds (silence)... Suggestions? Thanks in advance... /pj [default] exten => _XX.,1,Wait,2 ; Wait a second, just for fun exten => _XX.,n,Answer

Hi all, according to http://sip.fontventa.com/content/view/15/44/ i have compiled the mpeg4ip libries without problem. After copying the app_mp4.c file into de Asterisk apps directory and changing the Makefile like. [...] app_sql_odbc.so: app_sql_odbc.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc app_mp4.so : app_mp4.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $<

Hello, I have an XP (SP2) CPU that has just been reloaded with the OS (hard drive was reformatted). The NIC works and I can access the Internet with no problem. When I join the CPU to the domain (there are many other CPUs already on this domain), it joins. After the mandatory reboot, if I try to log in using a domain account, I get an error saying the username and password are incorrect. I

Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:

Hi, I have been struggling with streaming ogg vorbis audio to windows media player for a couple of days now with no success. I would really appreciate if someone could provide more insight into this. Please do let me know if there is any other mailing list which would be appropriate for a question like this. I posted this to vorbis mailing list. However this issue could not be resolved.

Hi, I have been struggling with streaming ogg vorbis audio to windows media player for a couple of days now with no success. I would really appreciate if someone could provide more insight into this. Please do let me know if there is any other mailing list which would be appropriate for a question like this. Apparently it does work. Simply go to the following website

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These

Hello, I am new to Asterisk, as of today. Installed on a RH9 box, with no problems. Built with 'make samples' as to get an understanding of how things work. Currently, I am utilizing SJPhone as a SIP client - not interested in shelling out cash for the IP phones, until I know I have a hold on things. I have run into a few problems I was hoping you could help me with : 1. When