similar to: No Sound via Sip Phone

Displaying 20 results from an estimated 11000 matches similar to: "No Sound via Sip Phone"

2004 May 26
2
SPAM MESSAGE - [Asterisk-Dev] warning message (sound card) - when I run asterisk!!!
All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing
2004 Aug 20
1
x100p won't answer
Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks
2004 Aug 21
3
zaptel config
Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and
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 Jun 27
1
Re: I never get to hear more than 5s of the demo channels
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. I'm new to this so please forgive my ignorance if I missed something obvious. I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not linux but that's all we have available at that stage). After some struggle to understand how everything works, I set up some SIP accounts for test purposes. I can log in,
2004 Sep 18
1
Asterisk stopped answering the calls
Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000>
2004 Sep 22
1
(euro)ISDN: complete silence / can't hear a word.
Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network
2005 Jul 03
0
no sound. "Failed to write frame"
Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make
2005 Jul 04
0
no sound. "Failed to write frame" (2nd post)
Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make
2004 Sep 17
5
Background() command
Folks, Apologies ahead of time if this has already been asked (read the list for the last month looking for something similar). I have been trying to get the Background command to work with no joy yet. Here is what I am trying to do: 1. Answer the call. 2. Play the message in the background, while waiting on DTMF from user. 3. If I get a "1", then interrupt the message and dial the
2003 Jul 11
7
ISDN PRI E1 configuration with E100P
<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --
2003 Mar 04
1
Failed to play audio data file!
The extension.conf: exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten => s,5,BackGround,demo-congrats ; Play a congratulatory message When call is coming, asterisk always failed to play the message,
2005 Jan 18
1
Asterisk and IAX softphone (firefly) problem/question
Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly
2004 Sep 07
1
Asterisk + NetJet (ISDN4Linux)
Well, it has been a long time since I used my NetJet S cards with asterisk (moved to the quad PRI card) but I am trying to get this working again for home use. Basically, what I am using is this: Linux 2.6.8.1: config lines: # ISDN subsystem CONFIG_ISDN=y # Old ISDN4Linux CONFIG_ISDN_I4L=y # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y # CONFIG_ISDN_TTY_FAX is not set CONFIG_ISDN_DRV_HISAX=y
2003 Jul 01
3
H.323 Gateway Connection
Hi, I'm trying to setup Asterisk to allow users to dial out to the PSTN using a remote box supporting h.323. I'm using chan_h323.so, and I'm able to make outbound calls to a client like netmeeting with a line like this: exten => 242,1,Dial(h323/xxx.xxx.xxx.xxx) And I'm able to receive incoming calls to asterisk. However I'm not sure how to route calls to the remote h.323
2005 Feb 12
2
Asterisk+GNOMEMeeting=No Sound.
Hi all! I'm newie to asterisk and I've been trying to make it work in order to use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none hardware phone. I'm using asterisk packages from Debian SID (my distribution), asterisk, asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried with any IAX softphone (gnophone?) but with linphone (SIP) I've
2005 Sep 12
1
Can't pickup inbound calls with TDM400P Fxo
Howdy, 1 x TDM400P card with 1 x fxo module. 1 x BT Pots line. Location - UK Calls work fine outbound but i'm unable to pickup the inbound calls. Asterisk debug: Asterisk -vvvvvvvvvvcg *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack
2004 May 19
1
Old sound in new call.
Hi, I have a problem that I just can't figure out how to solve. I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so
2005 Jan 17
4
Wait(n) -v- Background(silence/n) ?
Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a
2008 Sep 11
1
Probably very simple... call a number and play a sound?
Hey hey... I'd like to create a new feature code in asterisk so when a user dials... say... *00, it would then call some other extensions and play a sound file to them. So far, this is what I have... [testing-custom] exten => *00,1,Wait(1) exten => *00,2,Playback(beep) exten => *00,3,Playback(beep) exten => *00,4,AGI(festival-script.pl|I will now attempt the call) exten =>