search for: baggen

..." I get the following error while booting: %C542-1-UNKNOWN_VIC: VNM(1), vic daughter card has an unknown id of FF Is the VIC-2BRI compatible with the 2610XM? What IOS needs to be loaded? http://www.cisco.com/en/US/products/hw/modules/ps2641/products_tech_note0918 6a0080111b16.shtml --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get: Got SIP response 302 "Moved Temporarily" When forwarding the call to other SIP server. This is a "bug": http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

...-07-2-00.sbn -rw-r--r-- 1 root wheel 124716 Aug 13 19:23 P0S3-07-2-00.bin -rw-r--r-- 1 root wheel 461 Aug 13 19:44 P0S3-07-2-00.loads -rw-r--r-- 1 root wheel 587122 Aug 13 19:40 P0S3-07-2-00.sb2 -rw-r--r-- 1 root wheel 571489 Aug 16 22:28 P0S3-07-2-00.zip thank you! --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

...and do not load ALSA ; noload => chan_alsa.so ;noload => chan_oss.so ; ; Module names listed in "global" section will have symbols globally ; exported to modules loaded after them. ; [global] chan_modem.so=yes /etc/make.conf CPUTYPE=i686 CFLAGS= -O2 -pipe -funroll-loops --- Jan Baggen - jbaggen@ip2.nl IP2 Internet BV / http://www.ip2.nl

...will load an image and you will get the protocol invalid error as noted earlier in this thread. 4. Change your SIPDefauls.cnf to P0S3-07-2-00. 5. Reboot. The phone will look for the loads file and load the proper sb2 image bringing you to version 7.2. Dan On Mon, 20 Sep 2004 10:58:54 +0200, Jan Baggen <jbaggen@ip2.nl> wrote: > > >I have about 10 of these phones. All of them started with sccp firmware on > >them. I used the XMLDefault.xml file to upgrade them all to SIP 3.7. The > >phones "should" look for a CTL<mac> file, then a SEP<mac>, then SI...

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

My SIP INVITE messages generated from Cisco 2610XM to Asterisk PBX is not in 'application/sdp' -- MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary -- Howto turn this off on my Cisco? I read something about it on cisco website but I can't find it anymore.