similar to: SIP and ISDN

I have dealt with machines that have multiple network cards in them before, but never when they were on the same subnet so this issue has never come up before. My problem is that I can only access one IP address at a time. I started out using dhcp and found that if I went through the dhcp song-and-dance then that address became active and the other one was disabled, and vice versa. On our local

Hello all Here is the scenario: We have a mail server VM which currently has two virtual NICs attached to it. One NIC is has an IP on a subnet with a default gateway defined and the other NIC has an IP on a different subnet with a different gateway on a different VLAN defined. Now when I activate both NICs, and run an ifconfig -a, I see that both IP addresses are showing. Now here is the

I have a Centos 5 machine here that, up until about a year ago, was happily running Icecast and serving streaming audio through through three network connections, consisting of one "local" connection (local address 192.168.1.5) and two cable modems to talk to the outside world. We shut this down about a year ago, but now I am attempting to get it going again on one outside connection

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Hopefully this isn''t completely inappropriate, but lately I''ve been curious about what books people have found most helpful in their development as programmers. If one of the icons like Dave Thomas, DHH, Mike Clark, Andy Hunt et al could weigh in that''d be awesome :) Here are some of my favorites: Refactoring, Martin Fowler Mastering Regular Expressions, Friedl

Hello, I just received a couple SX440isdn phones and was wondering if they can be plugged into a Diva 4BRI port in NT mode and work with asterisk+chan_capi? I know they probably work fine with mutliHFC cards with either bristuff of chan_misdn but since I have some spare Divas, I was curious about their potential as phone ports. The Diva's 3 and 4 ports are already set to NT mode at boot

Hi, I am new to RSpec and BDD. I was wondering if RSpec specifications are a good solution for testing REST webservices (not implemented in Rails). If so, what would be a good way to test something like the API described below? =========== GET http://localhost/workflow text/xml Result: 200 OK <?xml version="1.0"?> <workflows>

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

I''m getting an error 500 and I''m at my wits end as to what could be causing it. I''m sure the path to the ruby executable is correct so what else could be causing it? -- Posted via http://www.ruby-forum.com/.

Ok I know this may have been covered and I did have a look back in the archives but didn't find anthing so I am asking it now.. Many moons ago the MySQL CDR functions and MySQL Voicemail functions had to be removed from the main asterisk code because of licensing issues.. Now there is new MySQL stuff like MySQL FRIENDS for SIP and IAX definitions.. So how is it that these options

Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial("SIP/2004-9634", "CAPI/g1/43XXXXXX") in new stack > data = g1/43XXXXXX

Hello, I'm new to this list. I'm trying to configure my Asterisk to have user access their email. SO far users can leave voicemail but they can't access voicemail. As you can see I had sip.conf and extensions.conf below. Please advice how to access configure extensions.conf to have users access their voicemail. Thanks in advance. -motty SIP.CONF [general] context=default

Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error "service not available". Out office is connected to he office

Hello everyone, I have been using an AVM Fritz! card with chan_capi and mISDN for quite a while in PTM mode and it was working finely. Now, I needed more DID/MSN, so I switched to PTP. But now nothing works anymore :( I am using Asterisk on Debian Sarge stable and installed Asterisk along with chan_capi from apt-get. I installed mISDN from the CVS of isdn4linux.de. It is : - Asterisk

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

Hello, I'm trying to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it doesn't work. I tried to set an extension to dial my from-pstn context and it works. So I think there's a problem with my capi.conf or something... Here's a debug when calling -- CONNECT_IND (PLCI=0x101,DID=9100,CID=46720XXXX,CIP=0x4,CONTROLLER=0x1)

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI> capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903xxxx,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903xxxx' -> '97' -- reventix: info element CALLED PARTY NUMBER --