similar to: Asterisk Sip Proxies and SIP persistence

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

All, Recently my area got slammed by very bad storms (something among the lines of 120 MPH winds and lots of hail) which knocked out a quarter of the power in my city. When I arrived back to my home office, I naturally found everything powered off and silent, but what I did notice is my UPS still had full battery. While booting the systems back up after power was restored, I didn't

All, I have a Cyberpower UPS that I have been working with for about a year. I have used NUT in the past with good results, but as of late have been seeing issues with nut talking to the UPS. I will start with the information on the server that is running NUT, and has the UPS connected VIA an RS232 cable. O/S : Fedora 9 Kernal (uname -a): Linux haruhi 2.6.26.5-45.fc9.x86_64 #1 SMP Sat

All, I have a Cyberpower UPS that I have been working with for about a year. I have used NUT in the past with good results, but as of late have been seeing issues with nut talking to the UPS. I will start with the information on the server that is running NUT, and has the UPS connected VIA an RS232 cable. O/S : Fedora 9 Kernal (uname -a): Linux haruhi 2.6.26.5-45.fc9.x86_64 #1 SMP Sat

Ok, First I want to start off with this isn't the first icecast I have cloned, but it is the first where I have not only jumped Fedora release versions, but arch types as well (i386 - x86_64) and I am having strange problems. The Icecast versions are the same between the old and new servers. The biggest being is that the server binds to whatever port it feels like instead of the bind port

Karl Heyes wrote: > Seann Clark wrote: > >> The biggest being is that the server binds to whatever port it feels >> like instead of the bind port specified. The rest of the issues I have > > a random port bind is a new issue. Can you show us the > netstat -tnlp | grep icecast > for the xml provided? Thu Jul 03-13:54:17-root at haruhi-new:~> netstat -tnlp | grep

All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in

All, I just had a power failure that caused me a little drama. This spurred me into trying to get NUT up on my cyberpower 1500 again. For about 6 months I tried to get it to work with the new power panel drive with no luck. Today I removed all the old files for NUT and installed the newest version I could find (2.4.3) and it works perfectly now with my UPS. Hopefully now I

All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and

All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=5555551234 secret=password defaultuser=5555551234 insecure=port,invite context=broadvoice

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

All, I am contemplating moving static SIP users to SIP realtime, and I am wondering if there is a nice simple tool to be able to do this with? I am not concerned with something that would do all the work for me, just something easier to use for a decent set of changes, than pure sql or phpmyadmin changes for the users. This is also because I am going to try the same trick with my dial

Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->

Ok, sorry about the top post. Compiled and installed 3.2.3, found a few stilled borked configs with my MPD, and fixed that, and both Icecast (after playing with ownerships that were deleted when I de-yum'ed the icecast RPM from the Fedora repo's) and icecast is working. I will clean up the config to remove some of the things that were mentioned though. Looking at the interface for the

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack

With the increased number of "brute force" login attempts against port 22, I am concerned that an intruder may actually stumble accross a valid user/pass combination. To combat this, I would like to request an sshd_config option that would cause the running sshd parent process to keep track of login failures by IP address. If there are more than X number of login failures for a

I am running into an issue with rsync that I need some help with. When syncing large files (e.g. 1GB), the rsync algorithm creates a temporary 1GB file and then renames it when the transfer is finished. The issue I am running into is if the two large files have very few differences between them, the bottleneck is creating the 1GB temporary file on the target box. This process takes several

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)