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Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

Hello All, I always used a compiled version for a x86 system >From http://kvin.lv/pub/Linux/Asterisk/ , and Works fine. At this time , I'm using a Pentium IV Dual core, Running Centos 4.3. I tried to install the 64 bits Compiled version but has a translation time > 20ms. Is it correct ? I also tried to make my self compilation and apply The patch but the patch doesn't work on ICC

Hello, >From: "Hermann Wecke" <hermann@wecke.com> >Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern => >extensions.conf >Date: 8 May 2004 22:03:57 +0000 > >Is it possible to strip some numbers from the *end* of a number? > >I know that ${EXTEN:1} will remove 1 positio...

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM.

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

I'm trying to add 2 FXO cards but when the second is added, asterisk stop to respond (zap channel): app_dial.c:545 dial_exec: Unable to create channel of type 'Zap' modprobe zaptel and modprobe wcfxo didn't return any error. ztcfg -vv is reporting only 1 card: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels

Most of my outbound calls are handled by macro contexts: [macro-pstn] exten => s,1,Wait(.5) exten => s,2,PlayBack(beep) exten => s,3,PlayBack(silence/1) exten => s,4,Dial(${ARG1}/${ARG2}${ARG3}) exten => s,5,Playtones(Congestion) exten => s,6,Wait(3) exten => s,7,Hangup which is called like exten => _920[1289]XXXX,1,Macro(pstn,Zap/g1,9,${EXTEN:${TRUNKMSD}}) Because of

How I can retrieve the original ${EXTEN} value when falling into the exten => i,1,whatever "context"?? I'm trying to implement this extension rule: exten => i,1,NoOp(${EXTEN}) exten => i,2,Wait(1) exten => i,3,Playback(vm-extension) exten => i,4,SayDigits(${EXTEN}) exten => i,5,Playback(is-invalid) exten => i,6,Playback(pls-try-again) exten => i,7,Goto(s,12)

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help?

Hi Friends, Is Apple IPhone mobile is released in India? Is Apple IPhone mobile is released in USA? If IPhone is released in India, Can you tell me any Apple authorized showroom in Hyderabad (Andhrapradesh, India)? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

I followed the instructions at http://www.opencall.org/instructions.html and http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html I was able to compile spandsp (./configure ; make ; make install), "manually" patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the supplied patch does not fit the actual CVS apps/Makefile After make clean ; make install, I

I found some old messages regarding a possible pkt8 DTA "bypass". Anyone is using Packet8 with Asterisk? ========== http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat Got Softphone Working with Packet8 Friendly name: {Anything you'd like.} SIP domain: packet8.net SIP proxy: packet8.net Leave everything else at the default. When you login, it will ask for a username

Hello, I have an Asterisk with one local Cisco ATA and one remote Cisco ATA connected to the Asterisk, the remore connection is a satellite link with an 900ms delay. I can make calls from the remote site to the local site, but when try to call from local to remote it doesn't work. The Asterisk timesout, it sais no one answered and can?t establish the connection. Can anybody help me with

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

...ng error - Unable to specify channel > 2: No such device (Todd Lieberman) > 6. RE: vm e-mail notification stopped (Uriel > Carrasquilla) > 7. Re: Booting error - Unable to specify channel > 2: No such device (Anon) > 8. Re: Adding two FXO cards - not working > (Hermann Wecke) > 9. editing errors/typos in rev 2 of The Asterisk > Handbook (current > version on digium's site) (Andrew D Kirch) > 10. Re: Hot plug PCI? (Yury Bokhoncovich) > 11. Re: Cisco 7940G/7960G SIP phones local echo on > * > box. (Mitchell S. Sharp) > 1...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

Is anyone using a 3com 3CNJPSE to power a 7960G? I have a couple of 7960Gs and 3CNJPSEs but no combination appears to work. Both phones work fine with a cisco power cube. I get a 47.6V reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE. The network still works through the 3CNJPSE, just no power to the phone until I connect the power cube. Using two straight through wired