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Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEP<MACADDRESSHERE>.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24

What have people found to be the ideal setup for recording asterisk prompts? I'm looking for both the ideal application to record them in, the ideal format, as well as hardware (do I need a fancy studio mic or will a headset mic work?). Thanks, Justin

This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone & was hoping to collect some real-world input. Thanks!!

Hi, I have an odd problem where when a call is made to any landline phone the callerid number shows up on the caller id device with no problems. However when I call a cell phone from Sprint or Verizon it always shows up as unavaliable. I've tried multiple cell phones from each carrier. Though today I called a t-mobile phone and the callerid came through fine. Is this an * level problem

I finally have my 7920 working though I'm seeing this bizarre behavior. As soon as the 7920 boots and authenticates with the AP my 7960 release's its ip. Aug 24 15:52:37 babel dhcpd: DHCPRELEASE of 192.168.30.133 from 00:07:eb:26:e3:22 (SIP0007EB26E322) via eth0 (found) Aug 24 15:52:37 babel dhcpd: DHCPDISCOVER from 00:0d:28:2e:95:f0 via eth0 Aug 24 15:52:37 babel dhcpd: DHCPOFFER on

Hi, I'm trying to setup Asterisk to allow users to dial out to the PSTN using a remote box supporting h.323. I'm using chan_h323.so, and I'm able to make outbound calls to a client like netmeeting with a line like this: exten => 242,1,Dial(h323/xxx.xxx.xxx.xxx) And I'm able to receive incoming calls to asterisk. However I'm not sure how to route calls to the remote h.323

Hi, I just setup a box with RH 9, and latest asterisk via CVS. The box as a T100P card in it that is currently hooked up to nothing. I did have the sample configs in place via make samples, and the only change I made was to add an entry to sip.conf for my Cisco 7960. When I dial 1000 to get to the main greeting I hear nothing, though the command line output looks fine to me. Any ideas? --

Hi, I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login & pw. This doesn't seem to work quiet right, where only the last phone to register seems to get the calls. What is the proper way to set this up? Thanks, Justin

Hi, Can I do variable substitution in the [globals] section of extensions.conf? For example something like this: [globals] EXT_BOB=4206 PHONE_BOB=SIP/${EXT_BOB} Thanks, Justin

I'm looking for toll-free #'s in: Germany Australia United Kingdom China Russia Singapore Netherlands that ring to a US based PSTN #. I've contacted people like QWest, XO, etc.. and their rates are extremely high ($1.74/min from the UK). Is there a better way to do this that involves VoIP? Thanks, Justin

Hi, I have been trying to get my 7960 & 7960G to register with two seperate * servers. Asterisk box 1 is on out on the Internet running: Asterisk CVS-10/30/03-20:08:15 Asterisk box 2 is on the Lan running: Asterisk CVS-12/19/03-19:28:30 7960 is on the LAN running: P0S3-04-4-00 7960G is on the LAN running: P0S3-06-0-00 In sip.conf I have nat=yes to get the phones to register properly.

Hi, Any recommendations for PRI providers in San Francisco? I'm not looking for the cheapest provider, but a reliable provider with reasonable prices. Thanks, Justin

Hi, I'm running CVS-v1-0-07/25/05-11:15:51 and when I dial into the Demo extension on my 7960 I get no audio. If I monitor it with tethereal it shows no audio being sent to my phone. I can make calls from my 7960 via the Asterisk box to SIP & IAX providers and it works fine. If I unload the wct1xxp kernel module, then restart asterisk, I DO get audio when dialing the Demo ext. No

Hi, I'm trying to put together an * system with extremely high up time. The system as it stands now is 1 dual p4 with raid, redundant power, etc.. and a T100P card. I would like to get a second similar or identical box with another T100P card. I have 1 PRI, how do I get the second box to take over communication with the PRI should the first box fail? Of course I'd like this to be

Hi, I'm having problems with transfer from an analog line via a X100p and Cisco 7960's running SIP. With an attended transfer the a call comes in, I transfer it to another 7960, they answer I announce the call, press transfer again, the two parties talk for 1-2 seconds then the analog line drops, though the Cisco phone is not aware of this, i.e. nothing on the screen changes. The