similar to: E&M ?

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

How do I setup my T1 card as an E&M tie line? Any special configuration? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/8aac4dbf/attachment.htm

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

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

Attempting to talk to an Eagle Telephonics switch at a disaster exercise. Didn't think a plain old E&M wink start T1 would be this much of an issue. We finally got the Eagle to accept a call from *, but whilst I can hear the person on the Eagle, they can't hear me. When they initiate a dial out I only get the first 2 digits from their switch... Does anyone have decent

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

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

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

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

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

Hello, I have a T1 (not PRI) plugged into my Asterisk server. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. Thanks. Here are snippets from the relevant files: --

Hello, I'm trying to set the 3rd span of a new digium quad card as a E&M T1 for Faxes to a Hylafax server. The 1st and 2nd spans are working as PRIs. When I start asterisk, the logs show a signaling error and chan_zap.c dies. I also get an error that it can't read the gains but they are the standard shown below. 2.6 kernel, Debian Stable, * 1.2 svn from feb 2007 my procedure: make

We've been working for the past 2 weeks to get a new V400P working with our PRIs from the telephone company. We're trying to get the Asterisk server setup as a VoIP gateway for SIP and AIX. We can make SIP-SIP calls, but all calls from or to the PRI fail. This is the applicable entries from the Asterisk log (configuration files follow) for a call coming from the PSTN on the PRI. I

Hi All, I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to make outbound calls on the zap channels. The light on the card is green. Asterisk CLI shows all 24 channels when I give the command 'zap show channels'. I also noticed that Asterisk CLI shows an incoming call every few seconds on the 24th channel. This must be

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Hi, Can I make e&m wink start lines just wait for digits - instead of going to default? Someone else cleared a similar problem (as described below) on an fxo port with "usecallerid => no" but it is not doing the trick for me. In this case the line when straight to default which would be ok also. John I posted the stuff below about a week ago... I set up a t1 from my sys75 to