similar to: asteriskathome and cisco 2600

I'm using asteriskathome 2.5. I'm using 2 spa3000 for dialing-out. I configured a trunk for each one with maximun channels=1 and an outbound route that includes both trunks. When a second outgoing call is placed, Asterisk tries to place it in the same that is already in use resulting in a busy tone. ?What can be the problem? -- Alejandro Vargas

You have no dial-peer telling the router what to do with the outbound call. http://www.tape.net/~gerry/asterisk/cisco26x0.html At 12:50 PM 9/26/2003, you wrote: >Like Gerry wrote for callerid you need VIC-2FXO-M1 card. > >Right now I am stuck on making outgoing call. > >Could soembody help me with the configuration. > >On cisco I have soemthing like that: > >dial-peer

Hey, Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. Does anyone know how to configure this 2600 with Asterisk? Thanks, Erik

hello Dmitry thank you for your reply. Ok, you are right. i want to configure trunk h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: PBX(100)--->cisco--->asterisk11.13.1---->PBX(200) when i call from 100 to 200, everything is ok but when i call from 200 to 100, phone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is

I saw on here where there was an asteriskathome site where I could sign up for the mailing list. However when I bring up that site I just get a blank page. Is www.asteriskathome.org the correct address? Bob

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

i will be installing asterisk; before that i understood i need RHEL3 or CentOS and i downloaded asterisk@home1.0 already int this download page http://sourceforge.net/project/showfiles.php?group_id=123387 i have seen Download asteriskathome-1.0.iso and Download asteriskathome-1.0-md5sum.txt please anyone explain __________________________________________________ Do You Yahoo!? Tired of

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

I get echo on my X100P for about 15 seconds then it disappears. Believe me, when I get a chance I'm going to tweak it since my wife nags me about it. I converted my house over to *. My setup: Compaq Deskpro EN SFF P3 500 Dual NIC for real IP and internal IP X100P Card for PSTN. Cisco ATA186 (two cordless phones for the old lady) Cisco 7960 Phone Cisco 2621 w/ 2 FXO's and 2 FXS (Fax is

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

How is working the automatic fax detection? I'm making tests in asteriskathome and the ivr plays, the fax sends little bips but asterisk don't detects it as a fax. (for testing I routed one caller id to the ivr). -- Alejandro Vargas

Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.

Hey All, Our upstream provider requires the use of H323 and after several months (6!) of having problems with OH323 I've decided it might be worth biting the bullet and getting a cisco device that can gateway up to approximately 50 calls from SIP to H323. Would a 2500 or 2600 series do the job? Once we get to the point of 50 simultaneous calls hopefully we'll be able to get something

Hello list. I've been running a Cisco VOIP WAN for the last two years using CallManager V3.2. The remote 1760 routers are set up as H323 gateways. I've got a VG200 that is my PSTN gateway with a full T1 voice trunk to the local CO. I'm also running Unity v3.x for voicemail. We are currently using two ATA186 devices that are using the G.711 codec to pass incoming calls on the T1

I have couple of questions about the following. Currently I have 2 phone lines going into my house, and I would like to have both of those coming into asterisk. I also want to have a backup asterisk, so here are the main questions (I am knew to this so I apologize if I ask something stupid): - Is there a dual FXO card available from digium or do I need 2 x single FXO (if this is the case then

I upgraded from RC2 last night, but have a major call quality issue. Heres our setup: 1 FXS and 1 FXO card. Incoming/Outgoing calls via IAX trunking from our provider. G729 running between us and the VoIP provider. Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2 SIP firmware. Both these phones are using ULAW to the server, and we have plenty of G729 licenses on the server.

Can someone help me configure a Cisco 2600 offlist please? I have too little experience with these things. Thanks! Mike

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Asterisk Gurus: Our county is finally ready to begin implementing IP telephony. We intend to use a Cisco router as our PSTN gateway and Asterisk as our soft switch. The plan is to use SIP between the Cisco router and Asterisk. We will have a single PRI T1 connected to the Cisco router for PSTN access. My question is this: Are Cisco routers able to pass caller ID information (from PRI