similar to: FXO registration and VegaStream

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have

Any have any success using a patton smartnode 4118/js/eiu fxs gateway with asterisk? We we're able to get the unit to register with asterisk, but when trying to place a call, no codec was compatible, even though I had all of the following enabled on the patton ... # G.711 A-Law/?-Law (64kbps) # G.726 (ADPCM 40, 32, 24, 16 kpbs) # G.723.1 (5.3 or 6.3 kbps) # G.729ab (8kbps) the link to this

I have a problem where an Asterisk server is sending a premature DRQ... Not sure why.. Here's the setup - Asterisk using inAccess networks H323 replacement channel driver Connecting to a Lucent iMerge... The call connects fine - I get the out of the box greeting - but after exactly one Minute - the call terminates. I have had this problem on multiple different Asterisk configs... I'm

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Hello, I have started having a strange problem. Asterisk is connected via 4 analog lines to PSTN and we have SIP phones internally. All was working fine but recently each time a user calls from PSTN and when he is leaving a voicemail for someone, the caller gets disconnected after 30 secs. We have AMP installed. This is reproducible and is happening always. It seems that Asterisk is disconnecting

Hi, To solve echo problems, I'm considering 2 alternatives. 1> Sangoma A104d - I can't find support for asterisk 1.2.1 2> Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command "reload" can change rx_gain and

Hi, a partner, who exchanges voip traffic with my asterisk box, complains, that asterisk ignores hints about ports to use. Hints about ports to use, seem to be a feature of H323. (I'm not firm enough with H323 to verify this.) The remote party opens the media-in channel: remote-ip:port-A -> local-ip:port-B My local Asterisk-box uses the same channel for media-out: local-ip:port-B ->

I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin

Sir, Here is the working Audiocodes MP-11X FXO configurations to work with Asterisk. ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 905371 ;Slot Number: 1 ;Software Version: 5.00A.024 ;DSP Software Version: 204IM => 209.13 ;Board IP Address: 192.168.0.195 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.0.1 ;Ram size: 32M Flash size:

I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from