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...tdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options here...please advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com <BLOCKED::https://mail.microsoft.com/exchweb/bin/redir.asp?URL=http://ww w.identitymine.com/> chad.brown@identitymine.com <BLOCKED::mailto:mark.brown@identitymine.com> 253.927.7737 - Office 866.4ID.MINE (866.443.6463) - Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ---...

...l Bielicki Sent: Saturday, September 18, 2004 9:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system try zaprtc from www.junghanns.net. Works fine in my SMP systems ----- Original Message ----- From: Chad Brown <chad.brown@identitymine.com> Date: Sat, 18 Sep 2004 20:23:54 -0700 Subject: [Asterisk-Users] Timing source on SMP system To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I hav...

...ing source on SMP system - Disable RTCforzaprtc It was my understanding that you don't 'disable' rtc, but recompile it as a kernel module. Again, just my understanding as I can't try it until monday. Matthew ----- Original Message ----- From: "Chad Brown" <chad.brown@identitymine.com> To: "Michael Bielicki" <cypromis@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, September 19, 2004 1:13 PM Subject: RE: [Asterisk-Users] Timing source on SMP system - Disable RTC forzaprtc...

Can someone tell me how to do this...Given the following line: exten => *97,3,VoicemailMain(${CALLERIDNUM}@default) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home =

Notice Zap/g2 -- Executing Dial("SIP/chad.brown-d1ac", "Zap/g2/9528737") in new stack Aug 1 00:42:43 NOTICE[1200884528]: app_dial.c:714 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time Does anyone know why Asterisk wants to use group 2 regardless of how I am configured. Take a look at how I'm configured. Shouldn't

Does anyone know what ports need to be opened up when using TFTP to configure Cisco phones? The configuration below doesn't seem to work. However if I open everything up the files comes down just fine. iptables -A INPUT -p udp -s 0/0 --dport 69 -j ACCEPT Thanks for your help! Chad

...ers minus the include statement. Needless to say, I transfer my inbound callers into the [nooutbound] so they can call all my users but don't have a path to the outbound context. Works great! However, there must be a more eloquent solution without the duplication. Thoughts? Chad Brown - IdentityMine -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041219/ff08e72f/attachment.htm

For those that are interested, I was just out on the Cisco site and noticed that they had released firmware 7.4 as of March 11th for the 7940/7960 phones. Doug

I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmod...Any ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss 46201 0

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09 When call comes in and is sent to a Cisco 7960, I see: -- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack -- Called 3000 Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum retries exceeded on call 033f41c2187409b13ca364502ea9434e@206.222.193.101 for seqno 102

...ailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ------------------------------ > > Message: 8 > Date: Fri, 27 Aug 2004 10:51:05 -0700 > From: "Chad Brown" <chad.brown@identitymine.com> > Subject: RE: [Asterisk-Users] sip change? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <93D6FFFFE7DE7142962E7D4A331538E8034048@IMEXBE01.identitymine.com> > Content-Type: text/pla...

I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone

Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

I was having problems with conferencing when I found a list which suggested that ztdummy might not be loaded. I checked using lsmod and sure enough it was not loaded. When trying to load ztdummy I get an error saying "Can't locate module ztdummy". I am using Asterisk CVS-09/13/03-23:21:19 Any help would be appreciated. Thanks, Chad

We have 2 offices. One is in the US and the other in India. We are testing Asterisk as a possible solution. Does anyone have advice for the preferred architecture when dealing with the latency from the US to India? For example, although it is not a requirement to have an Asterisk server in India it may make sense to have one there and connect office to office vi IAX since I hear that it supports

Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? I'm thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/bdd57a91/attachment.htm