similar to: Odd occurrence

Greetings Listers, I'm running 1.4.21.2 on SUSE 11.0 with and zaptel 1.4.12.1 on a TDM400P. Most of my calls work great, but occasionally we try to connect to a customer or vendor external conference call and the call will drop after 60-65 seconds unless I have an Answer before the Dial in the dialplan. Isn't this solution a hack and what would be a better one?

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

Hi gang, I upgraded the E1000 driver on my machine from 7.3.20-k2-NAPI to 8.0.9-NAPI. This unfortunately did nothing to resolve the problem. The best workarounds I've come up with are: 1. use -l on scp and ftp 2. install wondershaper QOS and limit throughput to 32K. These are workarounds, but I'd really like a solution. -------------- next part -------------- An HTML

Hi Listers, I'm running 1.4.25-rc1 on opensuse 11.0 with dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. Incoming calls work fine. Outgoing calls made directly (exten => s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to let Asterisk make the call (exten => s,1,Dial(DAHDI/G1/5551212,,r). If I use "m" (moh) the

On Nov 4, 2006, at 8:30 AM, Wesley Shields <wxs@atarininja.org> wrote: > > On Fri, Nov 03, 2006 at 07:54:59AM -0800, Ricardo A. Reis wrote: > [...] >> In the II COLARIS - Joanna Rutkowska alert the possible >> new technology of Malware's using hardware virtualization, present >> in AMD and INTEL new processor. >> >> I've two questions ...

Greetings listers, I have been running 1.4.21 for about 7 months now, but have been told I have to move up the 1.4 food chain or into the 1.6 chain because 1.4.21 is too flaky for our POTS line handling (does funny things with echo, doesn't connect to external conference calls, etc.). Which release will give me the most joy/least headache/closest performance to

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

Greetings listers, I've got 1.4.21.2 using Polycom 501 phones and Zap lines. Most of my calls come in and go out fine with the exception of Mechanized answering devices. When I call my 401K plan (1-800-777-401K) the call will last exactly one minute. The call never bridges, so even though the connection is made, Asterisk hangs up at the end of the Dial command.

Hello I'm using the Read() function to play a message prompting for the user to type a number followed by the # key to validate, with a 30s time-out and 2 tries: ============== [test] exten => s,1,Wait(2) exten => s,n,Answer ;typed DTMF: prompt for number to dial: 2 tries, 30s time-tout exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,,,2,30) exten =>

Greetings, This may be a dumb question, but here goes. When I was on 1.4.21.2 using Zaptel, I had (at least as far as I could tell) access to line supervision on my POTS lines using a TDM400P/TDM410P. Since upgrading to the DAHDI branches of 1.4 (SVN and 1.4.26.1), I've only been able to duplicate the success of the 1.4.21 functionality once. To test what I'm talking

Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate "text-to-speech" in a "less mechanical" voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house

Hello asterisk users... We are having a little problem in our installation, we are using Asterisk 1.4.21.2 and zaptel 1.4.11 with a Digium TDM410P (3FXO + 1FXS), the problem is that when we disconnect the line from any of the fxo ports, we receive an incoming ghost call (using zap/x channel) it rings on the phone but we cant hear nothing...it's always doing the same everytime we disconnect

Hi, I'm trying to set up dovecot to do pop3 from Eudora, but it keeps breaking on authentication attempts using plain auth. When I snoop the connection with tcpflow, here's what I see: 128.174.246.068.00110-062.107.004.050.49653: +OK dovecot ready. 128.174.246.068.00110-062.107.004.050.49653: +OK dovecot ready. 128.174.246.068.00110-062.107.004.050.49653:

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Greetings listers, I've posted this at least once previously, but thought I'd try again. I've got a TDM410P card on Asterisk 1.4.21.2 and experience these two problems. 1. When placing an outgoing call, I get no audio until Asterisk bridges the connection (2-15 second delay). I can Answer before Dialing, but this give me a incorrect CDR and no way of

Hello my friends... We are having some problems with the fax in our asterisk server... We have: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.5 Digium Card TDM 410P This digium card has 3 FXO ports and 1 FXS port where we have a fax machine connected! The problem is that we can receive fax very good, but we can't make any outbound fax call, in fact, our asterisk get freezed

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username