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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, I'm really lost on the directory functionality of the asterisk. I am currently setting a ACD using asteriskathome. I want to use the # feature where in caller can access the companies directory. My issue is how to setup the entry on the directory. Sorry for my silly question Can somebody guide me please. Regards, Leonimar Cape __________________________________________________ Do You

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

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, 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

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 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