similar to: E1 config help and guidance

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

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Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green -

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

Hi, How would one activate/deactivate hardware echo cancellation on the TE406 card? Can it be done per channel? I'm going to run TE406 in the following scenario: ISDN -> TE406 -> PABX I understand from Steve Underwood's site that echo cancellation is not good for faxes (and they do that themselves). So what I want to do and bypass echo cancellation for selected extensions before

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

Hi I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian )with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card which is An E1 card. But the main problem is the first stage that no sync occurs the * card never syncs with meridian card I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk And I am assuming that meridian is using same as it is

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

On how to even search for this "feature" as I have no idea on what it can be. I've got a meridian linked to * (by EuroISDN) which is linked to a ISDN30. I can make calls from the meridian, and receive calls into the meridian. Great stuff. However, if someone dials an invalid number, then instead of hearing a "three tone", the line just drops and goes dead. The console

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

Hi all- This question is for those familiar with EuroISDN setup. I have a customer in Europe where I'm going to install an asterisk based system with 4 E1's. The customer will configure them all in one large hunt group. My question is about the E1 channel configuration. I know that usually single E1 spans are set up with 30 channels (1-15 and 17-31) as the voice (bearer) channels and

Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except

Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN <---1---> TE - * - NT <--2--> Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable