similar to: Outgoing CallerID on PRI problems

We have a Mediatrix 1102 hooked into the network. Both of the attached analog phones and all of their features work, but in the CLI we keep getting "-- Got SIP response 481 "Transaction Does Not Exist" back from XXX.XXX.XXX.XXX " (Where XXX is the IP address of the Mediatrix ) every few minutes. I have changed most of the settings in the sip.conf multiple times and have done

Hello, Does anyone know how to set the outgoing CallerID properly when using Snom200/SIP/CAPI/BRI? Following doesn?t work: exten => _0.,1,SetCallerID,526910 exten => _0.,2,Dial,CAPI/526980:${EXTEN:1} Asterisk writes: *CLI> -- Executing SetCallerID("SIP/226-ada0", "526910") in new stack -- Executing Dial("SIP/226-ada0",

Is there any way to change the outgoing caller id on BroadVoice I have tried SetCallerID(Name <Number>) but that does not work Thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/bbd95fcf/attachment.htm

Hello all, Recently I purchased an QuadBRI card from junghanns.net after some playing around, reconfiguring dialplans etc with the exception of 1 thing everything seems to work: I seem to be unable to set the outbound callerid. The dutch telecom operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past years I'm more then used to setting the MSN without the leading 0, this

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

Hi, I have 10 DID numbers. Calls coming from the PSTN network are routed correctly to the SIP users based on the number that was called. But when sip users call the PSTN network, the CallerID should be set to correspondent with their DID number. At the moment I can set the CallerID to a global number, but I have no idea how to check who's making the call. All sip users start in the context

The situation is that I want to have certain extensions that can only call nationwide and others that can make international calls (with a different provider). I suppose, you could also see this as a scenario that could involve multiple internal extensions that get matched to specific outbound trunks. I have tried to find appropriate commands in the Asterisk documentation, but could find nothing

breakdown of what iam doing ok i have access to 6 ips and i want to run all of them through my firewall ifconfig eth0 209.159.32.162 netmask 255.255.255.0 up ifconfig eth0:1 209.159.32.163 netmask 255.255.255.0 up that sets up the network card to have 2 address well in shorewall i tried to add eth0:1 to my interfaces well it says that Determining Zones... Zones: inet inet2 loc cust vpn1

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

1 small question i have 4 network cards on my firewall eth0 inet eth1 internel network eth2 customer network eth3 freeswan vpn is there a way that i can connect the eth2 and eth1 network together so that i can access the servers off eth1 from eth2? Marshal McInnis Tech / Web Designs 1-205-344-4455 Ext 208

Is it possible to show CallerID names for dialplan applications? When I call from phone-to-phone, it shows the CallerID from sip.conf or iax.conf, but I don't know of any way to show CallerID Name when I call the extension for an application (voicemail for example): exten => 1000,1,Answer exten => 1000,n,VoicemailMain I'd like the display to read "VOICE MAIL" when I

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

on i have a small question we have a linux box with a windows 2003 server well we natted all the ports and mail is working remote desktop is working web is working the only thing that does not work is vpn how can i foward vpn traffic to this server i checked the site and searched for foward vpn and got nothen can somone please enlight me where to start Marshal McInnis Tech / Web

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

Here is the situation that I would like to create: Call comes in Receptionist sees that the caller ID is Jenny <8675309> Receptionist picks up phone and transfers call to Batman Batman looks at his phone and sees that the caller ID is Jenny <8675309> I can't seem to figure out how to forward the caller ID. Is this possible with Asterisk?

I have an outgoing line that allows me to set my caller id. I want to be able to change the caller id when ever I place a call using this line. I would like my user to press 7, then be prompted for their 10 digit caller id number, then be prompted for the number they would like to call in the form 1NXXNXXXXXX. How can I accomplish this??? Right now I am using this script that users have to

If receive a call on the IP Dialog SipTone II, and the other end hangs up first, the siptone immediately enters into the congestion tone. If I initiate the call from the siptone and the other end hangs up first, same thing -- congestion. The same thing happens if we make calls from the analog phones attached to the Mediatrix 1102. This does not happen on our Snom 200 phones, which have