similar to: Unknown signalling 896?

Setup: System Phones <--> PBX <--E1--> * <--> SIP Phones Calls work in both directions. However, ringing feedback to caller only works in the SIP->System direction. System callers to a SIP endpoint get silence, until call is picked up or dropped into something else. I've tried both Dial() with the r option, and Ringing() before Dial(), and neither works. With PRI

Hi all. The SPA941 and friends have pretty multicoloured LEDs, but there doesn't appear to be any support for SUBSCRIBE/NOTIFY as * as implemented for extension hinting. Has anyone managed to get the phone to support this? Thanks! -- David Zanetti <david.zanetti@catalyst.net.nz> Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 -------------- next

[Reposting, as was bounced for non-member, sorry if this is a dupe] Arrangement: { PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones] \__[PBX system phones] Normal calls between PBX system phones and SIP phones work, in both directions. The call logs look like (ignore the no answer, it did ring):

Hello my friends Today at FISL (International Forum of Free Software), I''ve had the opportunity to watch a Turbogears presentation. The framework, itself is really weak in comparison to rails, but... When the guy who was making the presentation show the NEW 0.9 TurboGears Toolbox I''ve though: - Isn''t something like that in Rails? Well, after a little googleing

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 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

THe wiki doesn't seem to have any user reports. If your using them, how are the working, better, worse about the same. Also what hardware seems to be stable with them installed. Alan

I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI> module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an

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

Hi, I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. I'm going to be using them in an office, 12 phones, on a LAN connected to an asterisk box. Thanks for any advice or

so here it is, the problem that's been nagging me for the past 2 days: connected a box to my telco's NTBA <-> zap/asterisk. which works: box:/etc/asterisk# cat /proc/zaptel/1 Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" HDB3/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In

Hi I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have started to notice the following messages when I recieve a call on my Zap channel :- [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my zapata.conf :- [channels] echocancel=no echocancelwhenbridged=no rxgain=-5.0

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

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

Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: *zaptel.conf:* span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be *zapata.conf:* [trunkgroups]

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

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

Hello, On a Asterisk 1.6.1 powered system, I've just discovered that using Busy() application in dialplan was no enough to send a Busy signal on incoming Dahdi channel. On this specific install, adding an Answer()) and a Playtone() statement in dialplan triggered sending of busy tone but I'm still surprised by my findings. Should I expect public switch to send a Busy tone to caller

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

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 =>