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Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

https://bugzilla.samba.org/show_bug.cgi?id=5299 Summary: 2.6.9 client cannot receive files from 3.0.0 server Product: rsync Version: 3.0.0 Platform: x86 OS/Version: Windows XP Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy:

...enior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > > ------------------------------ > > Message: 3 > Date: Tue, 28 Apr 2015 11:54:11 -0700 > From: Chad Wallace <cwallace at lodgingcompany.com> > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] adding area code > Message-ID: <20150428115411.71697421 at ws78.int.tlc> > Content-Type: text/plain; charset=US-ASCII > > On Tue, 28 Apr 2015 07:21:12 -0700 > Motty Cruz <motty.cruz at gm...

when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. thanks, Thufir

On 11 Mar 2015, at 17:53, Matthew Jordan <mjordan at digium.com> wrote: > On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes > <steve-lists at geekinter.net> wrote: >> Anyone know where it?s gone?.. Appears to have been down all day. > The hamsters should be running in their wheels again now. Cheers Matthew. Give them some food from me. Steve

On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

...2015, Bryant Zimmerman wrote: > > > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) > > Missing a colon? > > ${EXTEN:-1} > Does that work? I've always seen it like this: ${EXTEN:1} -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0

...tensions, one with priority 1 and one with priority 2. The first one will set a variable and hangup, and the second.... there's no priority 1 for that extension... I've never tried that... I'm assuming it just won't work. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0

Hello, I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3 Thanks, .Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150729/68b59f7c/attachment.html>

Anyone know how to suppress the -- Remote UNIX connection / disconnected messages. I have a monitoring application that calls asterisk from the command line to verify some uptime stats. I would like to not have the console log the connections.. Any ideas are appreciated. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two

Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1111 1112 1113. What i want to know

Hi all, I am configuring asterisk as a prepaid calling card. I am getting different local rate from my ISDN provider e.g 0.002 for landline and 0.13 for mobile etc. In this case I thing I have to say my asterisk/a2billing to bill based on local access number. so How can I retrieve called number (eg. 03-6832-1040 and 0120-272-060 is our ISDN PRI access number) to my asterisk server so i can

Hi, i need to match a number with like 03012345678*0 or 03012345*9 I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring that *X is required at the end. The interesting part is that like expected _X*X is matching only numbers like 1*1 and not 11 Regards Robert Wagner -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type:

It seems that for local channels (asterisk 1.4.33) the variable Variable: SIPADDHEADER="Alert-Info: Ring Answer" (call polycom phones and ring then auto answer) Is ignored, Is this just an oversite or is there some reason? It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration

Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full:

Hi, I'm trying to install voicemail-imap support but there seems to be a missing module: imap_tk checking for mandatory modules: IMAP_TK... fail configure: *** configure: *** The IMAP_TK installation appears to be missing or broken. configure: *** Either correct the installation, or run configure configure: *** including --without-imap. My configuration Ubuntu 14.04 LTS Asterisk

Hi all, I'm having issues with overwrite caller id, when I call someone my caller id should be "mycompanyinc" but instead my id shows up as my extension number 101. this is what i have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid="iuser 101" disallow=all allow=ulaw allow=alaw username=101 secret=Passwd dtmfmode=rfc2833 host=dynamic mailbox=101 at