similar to: Cisco router/SIP gateway registration

I'm not sure if I am wording this correctly, but I'll try. I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap analog phones plugged into the FXS ports. I am able to get * to ring those phones when a call comes in, but I cannot get the phones to dial out. I guess it's all syntax that I'm doing wrong. Does someone have a couple small snip-its to accomplish

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

On Tue, 2005-05-31 at 13:38 -0400, Jared Mauch wrote: > On Tue, May 31, 2005 at 12:30:07PM -0500, John Lange wrote: > > I am not a Cisco person; so the question is, is it possible to have one > > of the following: > > > > 1) Have the Cisco authenticate (register) as a SIP client to the > > Asterisk server. This allows me to place the Cisco in its own context. >

We are using a Cisco router with a T1 card plugged into a PRI provided by a local telco (Allstream). This Cisco accepts calls and sends them to a couple of servers running Asterisk depending on which number was dialled. But there is a problem. When a call comes in to the Cisco from the PSTN it sends it to the Asterisk server something like this: FROM: 204XXXXXXX@<CISCO IP> TO:

hello everybody, i want to configure a sip trunk between my system which has asterisk 11.5.1 and a cisco router. this is my scenario: Freepbx-----my system-----cisco-router----Freepbx my system acts like a router. if i set just one codec in dial-peers on cisco router, every thing is ok and i can make a call. but if i set different codecs in a voice class codec and assign it to dial-peers in

I'm new to Asterisk and have been attempting various configurations. I'm having problems with the basics of SIP to SIP phone communications within my own network. I've configured two phones ( Xten X-Lite) and whenever I dial either one I get errors as follows: *auto-congestion SIP/Phone 1 *SIP/Phone 1 is circuit-busy *Everyone is busy at this time I usually don't post such

Hello All, I am trying to connect to a third party's SIP based Hosted Billing service that supports G729 as I have found it difficult to get hold of prepaid billing solution for Asterisk. I was able to get an IAX2 connection working fine with another provider but as I am unable to find a billing service that supports IAX/IAX2, I need to connect to a hosted billing service using SIP and

Hello asterisk community, I have successfully set up asterisk as a SIP PBX and now would like to connect to the outside world using a Cisco 2600 with VIC-BRI as an ISDN gateway. This works already in the lab, but I have security concerns before conecting the gateway to the internet. I currently don't know exactly what VoIP services the Cisco runs by default besides SIP (H.323, MGCP, ...)

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am hoping for a little help from the collective. Inbound I have 2 different numbers. I can call in on both

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

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Hey gang, I have a 7960G in SIP mode. When the phone powers on it registers fine with * and calls can be made to/from the phone. But after a while 'sip show peers' no longer shows an IP for the phone. I can still make outgoing calls but can no longer receive incoming calls (because * has no IP to route the call). Any ideas on this? One of my other phones (MGCP) does a 'keepAlive'

I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

Hi all, I have problem with my Asterisk. I'm using the softphone Xten-Lite. I've removed the SIP client information in sip.conf. The softphone can't register to Asterisk, but it can make outgoing calls. I've tried to add back the SIP client information into the sip.conf, but make a wrong password in the softphone. The registration and outgoing calls are failed as expected.

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside

hi All, Sorry to repost but................ I have a server setup that runs sip no problem. I want to try a cisco phone. how do I a) Tell if I have skinny support loaded b) Load it onto a debian system Many thanks

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