similar to: sip add header

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

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

Hi All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: "Got SIP response 400 "In alert-info header: Empty value expected" Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I

I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and

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

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:

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

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

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:

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

I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. What would be the best way to handle this? I have been looking at the strategies and none of those seem to be right for this. My current thoughts are probably a macro that places a penalty on the user tell the next call is answered.

Hello, list. First of all i want to say sorry for my english. Long story short, on my future work i'll deal with asterisk and now i have a test case. But i'm very young to asterisk and don't have a lot of time so any help is appreciated. Test case: We have e1 trunk and multi-channel sip line. Clients waiting in the queue, which can handle 30 clients. They listen mellody and their