similar to: Cisco 2610XM and Asterisk

I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots

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 --

Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco

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

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

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

Hi! I've bought the EICON DIVA V-2BRI card an begin installation soon. Are there any issues with certain kernels? E.g. the standard debian kernel 2.4.27-2-386 and kernel-image-2.6.8-3-686. Further current kernel 2.6.16.14 would be an option. Thanks Klaus

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. ;

This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it

Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is

Hello, I am trying to update to dahdi 2.2.1. But somehow the xorcom (2BRI) device does not want to work. I pasted below the information I have. pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader reset pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[4937]: --------- FIRMWARE LOADING: (usb) [1 devices] ..Got all 1 devices 'xpp_fxloader'[5003]: ---------

Hi folks- I'm working on getting a test Call Manager Express system working with Asterisk. My plan is to have * support all the voicemail boxes for the CME/SCCP phones. Right now, I can call from a SIP phone to a SCCP phone and back fine. Calls go from Phone->CME->*->Phone and the reverse. Voicemail works for my SIP phones, but does not work for the SCCP phones. I tried to follow

Hello All, We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is

I have an asterisk box setup with a T1 card (hooked to a pri from the telco). Sometimes when people call a number that rings down that pri, the first ring is really really long, like 3 normal rings put together. My indications.conf is the default that comes out after you do a make install. Is this (first long ring) something I can control?

Hello, I'm in the process of building an Asterisk machine and need 5 or 6 BRI-Channels. I was looking for the beroNet and Junghans cards and stumbled upon the Xorcom Astribank xBRI products. Has anybody tried out the Astribank xBRI-Channel Banks? Are they production ready or should I go with a beroNet BN8S0 or JUNGHANNS.NET octoBRI ISDN? Thanks in advance, Lars -- We don't like their

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.

Hello I have the following structure SIP H323 (chan_h323) SIP Phone ------------------------> Asterisk/H323 -------------------------------------------> CISCO Gateway (CISCO 2610/NM2V-VIC-2BRI) -------------------------> ISDN SCCP Phone --------------------> CISCO CCM V3.3 --------------------------------------> SCCP H323 I have the

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway