similar to: Call waiting setup/Confenencing problems in AAH

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen. I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine. So I am

Hi On my FC3 box with asterisk 1.0.9....MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any

Hello, This is what what I see after a Yum install asterisk16 asterisk16-config freepbx: Use of uninitialized value in string ne at /var/www/html/panel/op_server.plline 4997. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5439. Use of uninitialized value in substitution (s///) at /var/www/html/panel/ op_server.pl line 5440. Use of uninitialized value

Hi there, We did a clean install the AsteriskNOW 1.7.0 64 bits ISO and configured it. On the main screen (Crtl-ALT-F1) we keep seeing the following lines when making a call 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

When I drag and drop a call from the PSTN to a SIP phone (the SIP phone's icon) using FOP .25 with * 1.0.9 with the intent of transferring it, the called party gets transferred rather than the calling party. This is controlled by the reverse_transfer parameter in op_server.cfg but the behavior is exactly the same whether the parameter is set to 0 or 1. This is after the call is picked up by

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

Howdy - I've been noticing a problem where I only receive a single call, before other calls go to voicemail. This only happens when the user is on the phone. I have the polycom 501's setup for 2 lines per key and 2 line keys for the first registration, which should allow for multiple calls. Anybody have any ideas what is going on? thanks Rolf Brusletto

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

Hi everybody, I've proxy server IP, user ID and password. Now I need to connect to a remote Asterisk server as a SIP using my Asterisk @ Home box. That Asterisk server will make PSTN calls for me. I think I am making mistake while setting up the Trunk because when trying to make calls, it give all circuits are busy error. When I setup Sipura adapter, which is relatively easier to setup,

hey all, a followup from yesterday, not only are my incoming faxes blank, but they are also EXTREMELY small (like .4in wide), I've seen several people mention this problem in my searches, but no definative answer that works with the fax>email setup. Is there any resource that explains how AAH handles this or some tips for troubleshooting this issue? I've gotten several replies to my

any pointers to where this list is? I dont see it on the sourceforge pages. Hans Witvliet wrote: > aah-handbook (version 1.6) doesn't spill a single character about bri > and "tfot" doesn't spill much paper of the subject either ;-( > > Any suggestions/pointers > > Hans > You may want to try the AAH mailing list.

I am running AAH 2.8. I have an IVR for our main phone number that allows users to dial an extension directly. I would like to have a "this call may be recorded" announcement played before the call gets transferred. There is not a built-in option for this in the IVR web interface, but one way I can do this is to create ring groups for each user with announcements and modify the

On 1/21/2013 2:01 AM, Óscar Fuentes wrote: > Ashok Nalkund <ashoknn at qti.qualcomm.com> writes: > >> I was using the find_package(LLVM llvm/share/llvm/cmake) and >> llvm_map_components_to_libraries(REQ_LLVM_LIBRARIES jit native) to get >> the libraries to link against. This works well for the libLLVM* >> libraries, but how do I implement similar find stuff for

Hello All, Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH? I keep getting 'Invalid Channel' but I cannot figure out why. Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050607/b18b3743/attachment.htm

Hi, Can anyone please point me in a direction as to how to set up these 2 pci cards with AAH 1.1? I have (am still) googling left, right & center - but haven't found a definitive guide yet. The centos based setup lacks any of the tools I know (insmod, modprobe ....) so it is time consuming just to even dig around the AAH box. There are no zaptel.conf files ....and on it goes. A

Is it possible to use the outbound routing features of AAH0.9 but also allow a user to dial a prefix to force the use of a certain route? Thanks, Wiely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050510/0c261294/attachment.htm

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3