similar to: How to Busy signals on DAHDI

2014-02-06 11:09 GMT+01:00 giovanni.v <iax at keybits.org>: > Il 05/02/2014 8.42, Olivier ha scritto: > > channel then it depends upon what you have the priindication option >> set to. With >> priindication=outofband then a busy cause code is sent to the >> network and the call >> is hung up. With priindication=inband then a busy tone

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

I'm using a TE405P and stable version of Zaptel. When I call a BUSY number on my E1 PRI, I don't get a busy status. The caller hears a busy tone, but the CDR logs a NO ANSWER when the caller hangs up. Is this normal for this version of Zaptel?

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

I've been beating my head a bit against the 1.0.6 Debian builds of Asterisk, using an E100P (E1, single span) board. In machines I've built in the past (back in 1.0.0 days), config I'm using and that card and 1.0.0 driver combo worked fine. ztcfg reports no problems: SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. And zttool sees the card, and

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

hello: thanks for Tzafrir Cohen for dahdi testing. I installed dahdi-2.1-r3c svn code and asterisk1-6 for testing OpenVox B400P and junghans card. i fund that there is bug (i think) to dectect NT or TE mode. actually on the board, i set it as TE mode, but after start wcb4xxp, but it show the port is NT mode. to detect the TE mode, I modefy the code in base.c

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Hello gang, We just got MaBell to turn on our callerid. I tested the capability with a southwest bell box and a plain phone, so I know the line is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a TDM400P card. Here is my dahdi_cfg -vv output: dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.3 Echo Canceller(s): MG2

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the