similar to: zaptel installation

I have what my SBC/ATT representative calls a "Supertrunk," but he can't tell me the specifics I need to know to configure Asterisk to work with it. By fiddling about, I've come up with the "almost working" configuration below. This works except that it takes about 4 seconds from when the console says the line is answered until it plays my prompt. Watching the

> I've never bothered to check to see if cat5 cables use the appropriate > mating twisted pairs or not. Since the pinouts are different for cat5 vs > T1 cables, I'd have to guess a single strand is used from two different > twisted pair groups. That wouldn't be cool, but in short runs it > probably doesn't have much of an impact. > IIRC, standard Ethernet uses

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

I'm not 'overly' new to asterisk, but I am very new to the Digium card / Telco aspect of it. I was able to get a TE110p card, installed.. and hooked a t1 up to it. I'm using zttool to just look at 'some' output. Here is what it looks like when I just load it up: Current Alarms: No alarms. ? ? ? ? Sync Source:

> Close. 10/100mbps Ethernet uses wires 1,2,3,6 but that is pair 2 & 3. > Pair > one is the pair up the dead center (pins 4&5), pair 2 is pins 1&2, pair 3 > is > 3&6 and pair 4 is 7&8. A T1 uses pairs 1&2, which is why you can't use a > regular crossover cable for a T1 crossover, but you can use a regular > ethernet patch cable as a T1 patch

Hi, It seems like my ATA is making a ringing noise... (as it used to), but now (After the upgrade from 1.0.7 to 1.2) asterisk also is either making the ringing, or passing the PRI ringing from the telco on to me. Any suggestions on how to fix this?

Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn signalling=fxs_ks usecallerid=yes hidecallerid=no

Hello everyone, I bought 2 HFC-ISDN Cards and want to run the first card in NT-Mode an the second one in TE-Mode. Everything looks ok under SuSE 9.1, but I can't dial out. I removed one card, for testing purposes and want to run this one card in TE-Mode. I only want to make a call with my Grandstream BT-101 over Asterisk via ISDN. When I try to make a call I get: ---------------------

Hi, I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and I've ran into an issue. After a call placed any DTMF tone causes the server to lock up entirely. Calls placed work just fine (except for a problem with echo cancellation). The phone registered to the server is a Linksys SPA-942. I am seeing in zttool some IRQ misses, but it never seems to go above 74 (below).

Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = hu > defaultzone = hu > dahdi_tool > OK Wildcard X100P

Hi Group, I have a problem with my FXO interface. It seems that asterisk cannot see any configured channels, only pseudo. Pls see the error and my config below. Rgrds Markus ----- May 2 15:09:00 NOTICE[17327]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel

We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in the following manner: Current Setup: Telco-> T1->PBX Desired Setup: Telco-> T1-> Asterisk-> T1-> PBX. I am first trying to setup the Asterisk -> T1->PBX part without disturbing existing setup so I can get asterisk to forward calls to PBX and once that is done, I would try to move the telco

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

I need some help getting a system running for one of my company's plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and FreePBX 2.8.0.2. My D-Channel keeps bouncing. The telecom tech told me he thought that I might be using the wrong sync source, and I think I might have been, but I changed DAHDI system.conf to "span=1,1,0,ESF,B8ZS" (from

Basically, have an old IVR application running under Apex's Omnivox software on a box with 4 old intel dialogic D240SCs, and would like to allow remote clients to gain access to aforementioned IVR application via softphone, 7960, ata, etc via asterisk with a TE410P. Unfortunately, all I know about the dialogics is that they're configured for D4/AMI, which I believe I have configured right

When I send a call from my TE410P using E&M, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information