similar to: Dial ZAP Problem

Displaying 20 results from an estimated 3000 matches similar to: "Dial ZAP Problem"

2007 Sep 24
1
DTMF dropping digits
We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI & DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are
2007 Jul 27
2
Unicall/Dont know how to handle Accepted
Hi, I've finally got running Asterisk 1.2.14 with UniCall & MFC/R2 patches, I can generate calls and all seems OK but I cannot receive any call, this is what I get: MFC/R2 UniCall/3 <- 0001 [1/IDLE /Idle /Idle ] MFC/R2 UniCall/3 Detected MFC/R2 UniCall/3 Creating a new call with CRN 32769 MFC/R2 UniCall/3 1101 -> [2/DETECTED/Seize ack/Seize ack ]
2006 Jun 17
0
T1 + E&M
Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system
2006 Jun 17
0
E&M + Dial tone
Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly
2005 Sep 08
1
can not make call with Unicall (MFC/R2)
Hi, ? I run the program testcall with one E1, it works fine; I receive DNIS and ANI for making calls and answering calls. ? When I start the Asterisk I receive call from outside correctly including DNIS and ANI, and receive the following messages: ? Sep? 7 10:29:59 WARNING[12167]: Answer Call Sep? 7 10:29:59 WARNING[12167]: MFC/R2 UniCall/2 Call control(5) Sep? 7 10:29:59 WARNING[12167]: MFC/R2
2007 Dec 06
2
astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR
Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling "E" and the TELCO says that asterisk would have to send signalling "F", as to make for asterisk to send signalling "F"? The TELCO says that the signalling "E" is
2005 Aug 22
0
Does Asterisk support T1 E&M Wink/Wink voicechannels on any Digium/Sangoma hardware?
I could only get *ANI*DNIS* working one way and that was setting my signalling type on the Asterisk side to 'featd' The Definity won't send *ANI*DNIS* information back to the asterisk as far as I can tell. Other than that, I've been running it with wink/wink E&M for a while now. TN464 circuit pack on the definity side, te110p on the asterisk side. Just a crossover cable
2005 Sep 01
0
Re: Asterisk-Users Digest, Vol 14, Issue 1
Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. (E10 and EWSD exchange store in two provinces difference. in Vietnam) this is my logfile of E10
2005 Aug 31
1
Sangoma card problem with EWSD Exchange
Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. (E10 and EWSD exchange store in two provinces difference. in Vietnam) this is my logfile of E10
2006 Nov 03
3
Problems Overwriting CallerID with True ANI
I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>
2008 Apr 01
1
Unicall + incomplete DNIS on international calls
Hello everybody, i'm from Mexico, at the time i?m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI and
2006 Apr 03
4
R2 protocol error
Hi, I have three R2 installation on different carriers, all shows the same inconsistency at varying degree. But, on most test calls we made, it reaches T3. The worst part of these, the carrier claims that it's my R2 box that is not responding in time. Please, check the attached file and take note of the timestamp, you'll find that in some call, it already contradict what the carrier
2005 Mar 19
1
ANI & DNIS sent to analog FXs Port Possible
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> | | | | | |
2005 Aug 04
1
no ring to callers?
OK, i've got asterisk @ home 1.3 up and running with Broadvoice. BUT I have one nagging problem to sort out. When you call my BV # the calling party gets no ring indication, just silence until either I answer the phone, or the call bounces over to voicemail. below is the console output when a call is recieved. what am i missing here? thanks Bernie -- Executing
2007 Dec 05
3
Adtran supervision problems
I am sending a call down a E&M wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=>799179,1,Dial(zap/g2,20,D(9179)) exten=>799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered.
2006 Apr 10
1
ANI and DNIS Seperation on a PRI (TelephonyNumbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])
Yes its a T3 split into seven trunk groups with one D channel and NFAS on each. Can you explain the cut function or point me somewhere please? Thanks, Steve -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on
2006 Jun 28
0
Dial Tone + E&M
Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail
2008 Jan 18
1
R2-Unicall Asterisk as CPE and as CO
Hi! Im having some troubles trying to configure * as a bridge between a telco and a pbx with R2, the scenario is this: ------------ E1/R2 --------- E1/R2 ------------ | Telco |-------------| * |-------------| PBX | | (Telmex) | --------- | | ------------ ------------ I can receive calls from the telco
2003 Aug 11
1
ANI/DNIS call routing
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:
2003 Aug 17
3
Monitor application temporary hack
[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian