similar to: CT1 and callerid / DNIS

I'm just double checking.. I was told it wasn't possible but i'm going to ask just in case. Can you set outbound callerid on a channelized T1? bkw

I've noticed a problem with calls to Hangup when talking to my Norstars over channelised T-1 E&M trunk lines - it's been present since I started to fiddle with Asterisk last December and it's still present in 'Asterisk CVS-HEAD-08/13/04-10:37:13'. Specifically, when a call is connected to Asterisk from the Norstar DTI card to my T100p I get the following conditions

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten => 8881111111,1,Dial(ZAP/g2) exten => 8881111111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? Thanks! --------------

I installed a Sangoma card with the default install. I'm getting five digits of DNIS with each call. The T1 is setup ESF/B8ZS wink start. Each of the digits of the DNIS are being used for extensions in the context. I need a single extension that let me start an AGI script that can use the dnis. Can anyone point me in the right direction to do this? Thanks, David Ruggles CCNA MCSE (NT) CNA

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Can someone in Asterisk'land subscribing to 800 service explain to me how to setup extension.conf to route calls based on the incoming DNIS/ANI. For example I want to route 3 incoming 800# coming across a trunk group to all land in the same queue. So I guess I am asking how to perform DNIS/ANI based call routing? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a reason the variable ${DNIS} does not get set with incoming calls via chan_capi ? Is it related to the MSN=X in capi.conf ? version = chan_capi-cm-0.6.3 example; exten => _95555555XX,1,NoOp, ${EXTEN}, ${DNIS} == ISDN1: Incoming call '0400000000' -> '95555555' -- Executing SetCDRUserField("CAPI/ISDN1/955555 55-135", "Incoming") in new

does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set("Zap/49-1", "DID=1231234*4812*") in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important

The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to

Hi, Is it to possible to route incomming call using dnis information to specific extension or section in the extensions.conf ?? like my users dial my asterisk box using different toll free numbers, but i can't offer them unique services. Plz suggest that if it is possible ? if possible then any example ... Regards Apna --------------------------------- Do you Yahoo!? The New Yahoo!

Hello everyone, Our company has an Asterisk server where one of the T1s is connected to an IVR. Asterisk is configured for FXO Loopstart, and the IVR is configured FXS. We can call into it, but our software doesn't expect DNIS digits on FXS, and the digits are being interpreted as user input. We can certainly change our software accordingly, but our test bed also includes other

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |

Not sure if this is off topic but I installed freeradius with yum on centos 5 and I'm hoping someone has some advice on getting DNIS proxy working In the acct_users file I have DEFAULT Called-Station-Id == "5500", Proxy-To-Realm := "xxx" Fall-Through = yes In the proxy file I have realm xxx { type = radius authhost = xxxx:1645

Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, "In what format do

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

I've got an E&M wink T1 into a Sangoma card, and it's dropping my DNIS digits. I'm supposed to get four digits from the CO, and I reliably get the first digit -- it's a crapshoot as to how many after the first I get. I have overlapdial=yes and immediate=no. Any other suggestions of things to tweak/try? Thanks! -Ken

Alexander, Thanks a million, you really saved me some time on this one. -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on a PRI(TelephonyNumbering Plan (E.164/E.163) (1)'*4105556654*8005550215*' ]) from the

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry