search for: at320

Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan

Hi, I have a few of Atcom AT320 phones. I can choose if install SIP or IAX2 firmware. Any suggestion? Mimmus

Hi, I flashed my ATCom AT320 phone (PA1888S based) with IAX firmware instead of SIP but now call transfer doesn't work neither using phone buttons nor using Asterisk features. I heard that it can be a real problem. Any help? Mimmus

...combination of this options does not produce anything Anyway, apart from FreePBX, what I have to check ? And moreover, what are the correct actions to do to record a call ? Let's say extension 555 calls extension 567, 567 answers the call and then press *1 and no other key ? I am trying with at320 sip phones and snom 320 sip phones thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does anyone have any recommendations on an IAX Desktop Telephone or ATA Device. I currently have 2 of the SIPURA-841's on my local network and now I am wanting to try an IAX Device at my remote office since I think that it would be easier to configure through various routers than a SIP Device. I just started to look at the Digium IAXy Single FXS

I installed an asterisk server with oh323 channel driver support. Then I uploaded the H323 firmware on a AT320 phone (Usually I use it as a sip phone, but I am using it just for test) Let's say that I assigned 945 as phone number, account and password to this phone, and its ip address were 192.168.1.88 Which are the right entries to add in /etc/asterisk/oh323.conf ? I tried (with no chance..) [945]...

...n preference out-of-band DTMF , and if the remote system does not support it they default back to inband. I would like to use DTMF as out of band , and I defined dtmfmode=rfc2833 in the peer configuration. Nope, I am no able to access any ouside services using DTMF; Another kind of phones, ATCOM AT320, can be configured as inbond or out of band; Again, If I use out of band (rfc2833) I am not able to use DTMF If I use dtmfmode=inband in the peer definition and in the phone configuration, it works ( I am using g711 codec) Anyway I wouldn't like to use inband, I would prefer to use gsm codec...

I'm trying to set-up H.323 support under Asterisk. I built a recent CVS release and the ooh323c code from the asterisk-addons. Everything built and installed and the H.323 stuff loads OK when asterisk starts. What is the easiest way to check if the H.323 code is working? I've edited the h323.conf and extensions.conf files but I'm sure that things aren't right. I've

Hi All, I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the pricing for these in Europe, so I'd like to hear from people here whether that is a reasonable price for them? TIA & BRgds -- Francesco Peeters ---- GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to