similar to: ZAP bchan and dchan HELP!!

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good day eveyone, I'm hoping that someone can help me. Perhapps i'm overlooking the obvious, in truth, I hope that I am. I've scoured the mailing list and google, and haven't come up with much. I have a Digium T400P thats been connected to a channel bank for testing for some time now. all has been well. I've now just had

Can anyone explain to me the difference in zaptel.conf and zapata.conf? I'm trying to get a real clear understanding of them but its getting a little murky in places. I will be setting up a PBX running asterisk with 2 T100P cards. I will be bringing a 23 channel PRI into one card and connecting the other card to a Nortell 24 channel FXS channel bank. As I understand it zapata.conf is

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

Hi all I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them configured as an E1 PRI connected to PSTN and another one configured as a T1 E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1 lines and 2 T1 lines connecting to this server. The purpose of this server is as a TDM trunk gateway that gets call from E1/T1 and then forward to an IP-PBX

I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

Hi: I'm using a te110p E1 card in Spain. There is way to change the dchan channel number? Here in Spain there is a voice company that works with dchannel in channel number 31. I must to set channels from 1 to 30 and dchan to 31 but when asterisk loads the chan_zap.c it says that channel 16 is reserve for dchan. Can you help me? Cheers. -- Jos? Luis Garc?a Ext. Pineda 2002001 Ext.

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

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

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

I'm having difficulty getting a PRI working between asterisk and a Merlin Legend. I have been attempting to get it going as esf, b8zs & 5ess (although I have tried other switchtypes...). I found the following info in the Legend docs: If the switch type is 5ESS, the protocol MUST be set for ATT Custom. If switch type is DMS-100, the protocol MUST be set for NTNAPRI The README in

Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and "Advantex" the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the

Today we have been having some problems with the dchannel of out T1's. I was wondering if there is a way for asterisk to send out an email or page whenan error occurs. Not I know errors happen quite offen for many reasons, but I would like an email sent when there is a TI problem, or specefic other problems. I am using the safe_asterisk to run asterisk and it sends an email when asterisk

Hi guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

Hi! I have problems with the ToN configurations in chan_capi-cm. I understand how incoming calls are rewritten using national and international prefix. But for outgoing calls - what is the ToN? Further, is there any debug info of the Ton? "capi debug" and the divactrl dchannel both show the CLI and CDLI, but do not show the ToN. How can I set the ToN? I suspect chan_capi to use the

Hi (Armin?) ! Today I had a problem with Diva Server 4BRI-8M 2.0. Asterisk 1.2.12.1 chan_capi-cm-0.6.5 divas4linux-melware-3.0.f-106.622-1 Asterisk could not receive and make calls on the BRI ports, although the ports looked fine within Asterisk. I usually use "/usr/lib/divas/divactrl dchannel -c 1" to test line activity. This time there was no activity (cryptic log

Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to

I've read as much as I can on this error and still can't seem to figure out what's causing this error: Jun 23 20:54:58 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]: