similar to: Zaptel - incoming delay

I've installed successfully freePBX with Asterisk, and got various internal extensions working, however. recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

am a NEW user to flac.installed flac on eac.tried to rip a cd using eac with flac as external compression.tried the 'wav' icontried the 'mp3' iconboth options saves the file as 'wav'thought eac will rip the cd in the 'flac' file format.is this a 2-step procedure?wherein eac to 'wav' and re-ripped on flac frontend?or is there a ripper with flac in it that

I'm getting the following errors in my error logs: ERROR! vuid 100 did not map to a valid vuser struct! At the time this error was being created I was trying to add a domain user as a local admin. I was trying to get the browse list of domain users up, but being denied. Cheers Jeff -- Jeff Gardiner [ gardiner@nospam.imaging.robarts.ca ] System Administrator - Imaging Research

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

Hello list, I am looking to enable samba aio on my Wheezy test box (I have version 4.1.17 from back ports). The output of smbd -b shows that the stable packages for my distro are not compiled using --with-aio-support. It appears that without recompiling I can use vfs objects = aio_pthread to achieve a similar effect? I'm aware that this somewhat dips into Debian specifics, but I figured

I have a TDM400P with 2 FXO cards and I'm using Asterisk@Home 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I see... - After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call -

This link: http://us3.samba.org/samba/docs/man/smb.conf.5.html#LOGONPATH States that: "Windows clients can sometimes maintain a connection to the [homes] share, even though there is no user logged in. Therefore, it is vital that the logon path does not include a reference to the homes share (i.e. setting this parameter to \%N\%U\profile_path will cause problems)." I am still

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)

We are trying to configure a Sangoma A101 card to allow both incoming and outgoing calls on a UK (BT) ISDN30e line with only 24 channels enabled. At present incoming calls work fine. We can't call out -- we get a BUSY/CONGESTED error. Do we need another context in our zapata.conf? In other words, do we need to reserve, say, channels 17-24 for outgoing calls? I also wonder if the signalling

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hi Johnny, Thanks for this - particularly the tip about cfe-dev. I'm currently trying to coerce lldb to debug these type of architectures (our current toolchain already outputs good dwarf info). However, I'm struggling since lldb has just assumes that the size of a byte is universally 8-bits. At some stage, I *think* at some stage we'd like to derive a compiler, from the "same

Hi, I was wondering if I might be able to ask some advice about doing residual plots for the lmer function in the lme4 package. Our group's aim is to find if the expression staining of a particular gene in a sample (or "core") is related to the pathology of the core. To do this, we used the lmer function to perform a logistic mixed model below. I apologise in advance

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If anyone has Samba setup to run as a PDC with Solaris 8 or 9 could you drop me a line? I have a few questions I would like ask. Our shop is looking at trying to implement this option with our SMB master. I didnt want to clutter up the list with more than one request. -- Thank you Todd Johnson ============================================== = Steven ?Todd? Johnson = =

Hi Patrik, Thanks for this note. It's encouraging to read there has been some provision made for non-8-bit bytes. I'm not a compiler/backend expert, (although maybe I'll need to be soon!), so I won't look at the patches right now, however may at some stage in the future myself or colleague may request these patches from yourself. Yes, our 24-bit architectures have non-power-of-2

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.