similar to: Adding entries on company directory

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hi, Can please someone help me. I have successfully installed Asteriskathome 2.5 on a server with a Intel Server Board SE7525RP2. May issue is after placing the TE405P in the server, it is not booting anymore. Has anyone in here have the same experience? Can someone please point me to the right direction. Thanks in advance, Leonimar __________________________________________________ Do You

Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

Hello, Can somebody please assist me with my problem. Currently I am using a Asterisk@HOme version 2.4 with a TE406P digium card. One the E1 is connected to a telco switch via an ISDN. May issue is that may incoming calls in the zap channels gets disconnected or release after 1 ring. I really dont know what setting should I change to increase the timeout of the ring. I have even tried upgrading

Hi Group, Is there anyway to force asterisk to use the ip address instead of the hostname in the sip via header. Our client's gateway is using a not FQDN as the hostname of their gateway. And I am suspecting that the asterisk is dropping the call because it could not resolve the hostname. I am also thinking to assigned it in the hosts file of my asterisk server so that in can resolve

0.8 appears to have been released. Start with that. It is very quick to update your current 0.6 or 0.7 iso. Just do this with rsync to do a differential copy: $ mv asteriskathome-0.7.iso asteriskathome-0.8.iso $ rsync -av --progress --partial \ prdownloads.sourceforge.net::sourceforge/a/as/asteriskathome/asteriskathome-0.8.iso . $ rsync -av \

I saw on here where there was an asteriskathome site where I could sign up for the mailing list. However when I bring up that site I just get a blank page. Is www.asteriskathome.org the correct address? Bob

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

i will be installing asterisk; before that i understood i need RHEL3 or CentOS and i downloaded asterisk@home1.0 already int this download page http://sourceforge.net/project/showfiles.php?group_id=123387 i have seen Download asteriskathome-1.0.iso and Download asteriskathome-1.0-md5sum.txt please anyone explain __________________________________________________ Do You Yahoo!? Tired of

?having problems with installing asterisk@home i downloaded the asteriskathome-1.5.iso file from asteriskathome.sourceforge.net link & burned it on a cd but it is not booting what seems to be the problem hoping for a quick reply -------------- next part -------------- An HTML attachment was scrubbed... URL:

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not

As far as I can make out the root password for the ISO download is supposed to be epping or EPPING depending upon which version you are using. I've downloaded an ISO image from the following link but neither passwords seem to work :( http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/asteriskathome-0.6.iso any one know the password for this one? -- Regards Phil -- This

Hi!I've 2 asteriskAtHome; How can I copy one database where are put all the sip authentificated registration to another one database on one other asteriskAthome so I've always the same Sip registrated and if one linux falls down I can run the other one without problems? Which files must I copy?then..I'll use a ssh scritp for this, I want only know which files I must copy...

I'm using asteriskathome 2.5. I'm using 2 spa3000 for dialing-out. I configured a trunk for each one with maximun channels=1 and an outbound route that includes both trunks. When a second outgoing call is placed, Asterisk tries to place it in the same that is already in use resulting in a busy tone. ?What can be the problem? -- Alejandro Vargas

How is working the automatic fax detection? I'm making tests in asteriskathome and the ivr plays, the fax sends little bips but asterisk don't detects it as a fax. (for testing I routed one caller id to the ivr). -- Alejandro Vargas

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home. We have also added new version of the usual Asterisk software AMP

Hi, I have cisco 2600 with fxo card, can we use it to connect to asteriskathome as SIP trunk..?, so we can use for incoming and outgoing trunk. Please need your help. -mendro-

Hi Group, I have an asterisk running as media gateway with a Digium TE405P 2nd Gen rev 2 with echo cancellation. It is interconnected to a telco carrier via ISDN Pri. The voice quality is clear except that sometimes a hear a beep sound that occure around 5 to 10 secs in the middle of the conversation. When I check the logs in the asterisk, I found this. Nov 30 00:48:38 DEBUG[27705]

Hi, Problem: When talking to someone (from pstn) and this person is not talking loud, the voice is cut by Asterisk. I tried increase RxGain but it changed nothing (was talking louder but voice still cut.) I use XLite as soft phone. I think this is probably a codec setting... but how do I check that on server side? I just don't know what to do. All works fine (asteriskathome) but I always