similar to: check_asterisk commands

Thanks Gene. Here is my localnet: localnet=172.16.64.0/255.255.240.0 Which matches our subnets network address and subnet mask. Are you recommending that I make it more restrictive? Thanks, Geoff > -----Original Message----- > From: Gene Willingham [mailto:gwillingham@comcast.net] > Sent: Tuesday, June 14, 2005 9:13 PM > To: asterisk-users@lists.digium.com > Cc:

I think I found the source of this. Been tracing it for a week. Look in sip.conf. It appears the definition of localnet has a bearing on how some sip devices handle invites and NAT. I had changed the localnet to 192.168.3.0, but did not change the netmask. localnet=192.168.3.0/255.255.0.0; All RFC 1918 addresses are local networks When I changed the netmask to 255.255.255.0 the problem

brett@websmyths.com wrote: > Also - an outside chance - make sure Tip and Ring > are correct. You could be getting ground loops - depends on the noise. > I am having noise and slip errors between my TE110P and a legacy PBX T1 card. Could this be the same symptom? The connection is made using a 15 pin serial on the T1 Card side to RJ48 on the TE110P side. I can't determine what the

Say I had a crossover cable that connected a Mitel SX200 to a TE110P and the pinout was done as such: 1 - 4 2 - 5 5 - 1 4 - 2 (the 5 and 4 are transposed on the left side) Instead of the proper way of: 1 - 4 2 - 5 4 - 1 5 - 2 What would the results be? We have had the former as our cabling for a few months and the connection has been fine. Slip errors here and there. But we have had major

do you have the manager interface turned on? You need to make sure your /etc/asterisk/manager.conf file looks something like this: ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [testuser] secret = test ;deny=0.0.0.0/0.0.0.0 ;permit=192.168.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write =

I have spent some time digging through the archives for comments concerning Asterisk and monitoring systems, and I have found few results. check_asterisk.pl.gz (http://www.dynx.net/ASTERISK/misc-progs/) which gives an error on download, and has no further Google references astping.tar (http://www.dynx.net/ASTERISK/misc-progs/ and also in the mailing list archives) supposedly sends a query to

Eric "ManxPower" Wieling wrote: >> >>> span=1,1,0,d4,ami >>> e&m=1-24 >>> > > Looks like you have told Asterisk to get it's timing from the Mitel. > I'll bet the Mitel is trying to get it's timing from Asterisk. > > Try span=1,0,0,d4,ami and run ztcfg -vvv > I just set this back. It was originally set to your

Correction: The hardware is a Wildcard T100P (not a TE110P) Thanks! > -----Original Message----- > From: Geoff Manning [mailto:gmanning@zoom.com] > Sent: Wednesday, May 18, 2005 9:07 AM > To: Asterisk Users (E-mail) > Subject: [Asterisk-Users] Integrating Asterisk into our Legacy PBX > <--Newb > > > I have been successful in setting up asterisk and making >

We use SetVar(TIMEOUT(digit)=8) In our dialplan to make sure that the user is done dialing before Asterisk executes the call. I just recently came across the piece I've copied below. It says for new incoming ZAP connections, the default digit timeout is 3 seconds and can only be configured in the source code. Is that true???? ============ How long will Asterisk wait?

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

Valued Colleagues, I am trying to configure and use asterisk manager API. The /etc/asterisk/manager.conf and the output of "netstat -nl" are appended below. When I restart asterisk, I believe I should be able to see the asterisk listening on port 5038 using netstat. But when I type netstat, I don't see any applications listening on port 5038. When I telnet to port

I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? TIA Manager.conf: [general] displaysystemname = yes

I'm not sure how much more simple I can make this but I just cannot seem to get my Asterisk 13 to accept a connection on the manager interface: --- manager.conf --- [general] enabled = yes port = 5038 bindaddr = 127.0.0.1 [myasterisk] secret=a permit=0.0.0.0/0.0.0.0 read = all write = all So, couldn't be any more wide open and simpler to connect yet: # echo -e "Action:

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

I have an AGI script written in PHP that worked great with Asterisk 13. I'm porting it to an Asterisk 20 site and have a strange problem. I tried running the script from the command line and it works fine; I see the script commands written to stdout like VERBOSE "SmartScreen v1" But when run from asterisk the CLI shows: [2023-06-30 15:50:47]

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

I am having problems using the manager even though I am following the instructions from the Manager.rtf doc. In manager.conf I have the following [general] enabled=yes port=5038 [fred] username=fred secret=fred read=system,call,log,verbose,command,agent write=system,call,log,verbose,command,agent I do the following: System prompt # telnet localhost 5038 Trying 127.0.0.1... Connected

Hey all, I'm sorry to bother you with something so trivial, but I seem to be having an issue with the Asterisk::AGI module. I am a relative newbie with Perl so it could be a stupid syntax mistake that I missed. It seems when I try to execute either the stream_file or the get_data subs nothing is actually done. It doesn't seem to stream the files, but on the console it says it played the

hi I've written, or upgraded a little more, a plugin for asterisk/nagios, just in case someone should be interested. it uses the manager interface to connect and checks staus. it's a dirty hack, but it works. see https://sourceforge.net/tracker/? func=detail&aid=746083&group_id=29880&atid=541465 for more info roy

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax