similar to: Call Quality Issues

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

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

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

When I start Asterisk, I receive these errors: Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on peer 'gv_trunk' without zaptel timing Jun 13 16:26:05 WARNING[2870] chan_iax2.c: Unable to support trunking on user 'zoom_trunk' without zaptel

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

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

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

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

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

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

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

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

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a Digium

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

I'm trying to connect an Asterisk system to an Avaya Partner ACS R6 system. The problem I'm having is that I cannot get the partner system to get CallerID over the T1 modlue. The partner is using the T1 with E & M signalling (which I don't think can be changed), and whatever I tried didn't work. My only option right now is to get FXS ports on the Avaya side plugged into the

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

Hi all, So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my