similar to: 7960 utilize all lines

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

2 new Cisco 7960 phones are requesting a CTLSEP file, seems like I triggered the universal application loader. I want to load the sip image 7.2 According to this Cisco information: http://www.cisco.com/en/US/customer/products/sw/voicesw/ps4967/products_upgr ade_guides09186a008022a968.html#wp1047292 If the CTLSEP MAC file is not present or is empty, the phone proceeds in nonsecure mode with the

Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb

I''ve set up Traffic Shaping on a Linux Router. Using HTB with SFQ, i''m trying to slow down heavy downloading for 20 subscribers over a 2048 kbit downlink. I''m classifying internet related traffic using iptables marking. bri0 is my local lan bridge, receiving egress traffic destined for subscribers. tc qdisc add dev bri0 root handle 1: htb default 2 tc class add dev

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Not normally a question for this group, but you guys are very bridge/router/firewall savvy, so I thought I''d toss it here. I have a bridge. On one side of the bridge is that fancy thing called the Internet. On the other side is my LAN. The bridge is the obvious demarcation line and a good place to put a firewall. Now, I have all my iptables stuff planned out, EXCEPT for nat. The

I know this is not a normal request, but I would like to connect two vlans into the same box that have the same subnet on them, and then NAT them out to the Internet. We don''t need to make any connections inbound to them, but only provide outbound access. Any suggestions on how to do this? If I can''t do it with routing then I could use something like user-mode Linux, or other

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

A little off-topic: I have the following hardware: 2610 XM NM-2V VIC-2BRI NT/TE IOS loaded: flash:c2600-ipvoice-mz.123-5d.bin" 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

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

Its been a few days, I was wondering if anyone else has any ideas on how to get this to work? If not, could I ask a the direct question as to if anyone here has (or knows someone who has) successfully got something like a Cisco Router to successfully establish an ISDN data/Internet connection through the Asterisk PBX?

Good Evening, The company I work for is attempting to connect an Cisco ISDN Router to an OpenVOX B200P BRI Card so that we can get it to dial out across an existing ISDN PRI Line also installed in the Asterisk PBX Everything has compiled successfully and the BRI card has been detected when mISDN is started: # misdnportinfo Port 1: NT-mode BRI S/T interface port (for phones) -> Interface

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

The place I currently work at has a Panasonic Key system with 9 extensions, and no voicemail. It services 2 PSTN lines. I am hoping to use Asterisk to host voicemail (I would like to use the IVR also, but I don't even know if or how it would work). Do I need to use a PRI between the two, or is there a simple solution? I would like people to be able to answer the phone and

Here are some notes I have about Linux bridging. I''ll try to separate what I know I know from what I think I know. Let''s say I want to bridge eth0, eth1, and eth2 together, all with an IP Address of, say, 1.2.3.2. This is how to do it: echo "Setting up br0 to bridge eth0 with eth1 and eth2" /usr/sbin/brctl addbr br0 /usr/sbin/brctl addif br0 eth0

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

Compiled Asterisk from FreeBSD port (0.9.0_2) When I start asterisk it uses 100% cpu. Searches on Google say to comment the noload => chan_oss.so in modules.conf But this is already commented. Make.conf contains some optimizations. modules.conf: ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; If you want, load the GTK console right away. ;

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

Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers.