search for: mc3810

Hi Everyone, I'm new to asterisk and trying to get together the hardware to run a few POTS phone extentions and one or two POTS lines for starters. For these low port counts, I could just go with FXS and FXO cards, but... I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in ethernet port and I believe it does SIP too... Will this mean that I won't need a T1 card and dedicated channel bank? ie. Asterisk connected over ethernet with the MC3810 and the POTS lines and stations connected to the MC38...

Has anyone here successfully gotten a cisco MC3810 talking with asterisk? I am getting the dreaded - Got SIP response 400 "Bad Request - 'Malformed/Missing URL'" back from xxx.xxx.xxx.xxx If you've gotten it to work you can feel free to email me off list. If your willing to share config's that also is a definate plus. Than...

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the cal...

Hello, I'm trying to run an ATA behind a NAT device, and am confused on exactly what the qualify config option does, other than send NOTIFY packets. Outbound calls work fine, but inbound calls do not go through. With qualify=yes and nat=yes, my show sip peers looks like: 7771111001/7771111001 10.0.0.10 D N 255.255.255.255 1222 OK (36 ms) So, it has established a

...Date: Wed, 30 Jul 2003 15:42:43 -0500 (CDT) > From: Brian West <brian@bkw.org> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > I have done that but I think its the Asterisk => MC3810 via h323 thats > causing that. Does anyone have an example on how i can dump sip to and > from the MC3810 to my asterisk server? > > bkw > > On Wed, 30 Jul 2003, Patrick wrote: > > > > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband >...

...) but I get this errors. Because I can't find anything on Internet I'm hoping u can give me a hand. here are my logs: -- Attempting call on SIP/sip_provider/1234 for application txfax(/va r/spool/asterisk/fax/215690048.1145968036.383.tif|caller|debug) (Retry 1) > Channel SIP/mc3810-a20f was answered. > Launching txfax(/var/spool/asterisk/fax/215690048.1145968036.383.tif|ca ller|debug) on SIP/mc3810-a20f FLOW Slow carrier up FLOW Slow carrier down FLOW Slow carrier up FLOW <<< NSF: 20 00 00 0e 00 00 00 96 0f 41 07 00 10 00 02 95 80 18 01 49 02 53 2e 43 2e 4e...