similar to: T100P Timing Was:T100P/ ZAP / PRI errors

So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local & long distance for free. It also does 'internet'. 'Just give me

All, When calling an invalid number using, I expect to hear: "dooh-deeh-daah We're sorry you have reached a number which has been disconnected ..." And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas

All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing

My PRI is being reset at least once a day with the following errors in the logs. zaptel, zapata, libpri, and asterisk are from CVS this morning.. This has been happening for weeks on all versions (including -stable). the T100P card appears to NOT be sharing an IRQ. xenon# cat /proc/interupts CPU0 0: 1203977 XT-PIC timer 1: 3 XT-PIC keyboard 2:

To the list ... I got the IAX2 stuff simplified & working (for now). See my earlier posting to the list. Now, here's a question for you all. I found a posting by J Todd where he gives BW utilization for various IAX2 codecs with trunking on. Now, the number of calls I can sustain over an IAX channel, obviously is going to be determined by the capacity and state of the physical pipe.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I'm getting the following error at random intervals on my TE410P with Asterisk CVS-03/30/04-11:49:01-CEST. I have two spans active, one connected to my Telco, the other to a Siemens PABX. Both spans display this behavior at random intervals. All calls are dropped when this happens. Spans are not necessarily in use when this happens.

Here is what you should write in extensions.conf: exten => _5.,1,Dial(IAX2/iax-a2:secret@a1.mystrx.com /${EXTEN}@inbound-calls So when you will dial anything beginning with 5, the call will be dialed in the context inbound-calls of a1.mystrx.com -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: asterisk-users@lists.digium.com De: willy@yponeinc.com Envoy? par:

Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them

Hi All! I have two [*]s, and both work OK as a simple local PBX. Now, I try to link them using IAX. Let's call those babies a1 and a2. From a1, I want to dial a phone connected to a2. Both boxes have a fixed IP address, and use standard port 5036, say a1.mystrx.com and a2.mystrx.com. Where I (obviously) get confused, is when it comes to inbound, outbound, registration, etc. Taking a hint

Hi all .. Maybe I am just missing something, but when I press '0' and then '3' to record my name, it gives me an 'after the tone .. '. Then I say my name and press #. It says: your message has been saved. So HOW do I listen to my recording, make sure it sounds OK, and then CONFIRM that is what I want the mailsystem to use OR CANCEL out and leave things the way they are?

To recap: T100P card wouldn't sync with the telco using line side clocking ( span=1,1,0.........) Had to use internal clocking (span=1,0,0.......) zttool showed Tx/Rx Levels as 0/ 1 For the grins of it I replaced the T100P card with another newer card from inventory. This newer card has the same rev on the ASIC / FPGA but doesn't have any of the various jumper headers installed

I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten => 1004,1,Dial(Zap/g1/NPANXXXXXX) I see the following on the asterisk console: -- Executing Dial("SIP/sbruton-b8ce", "Zap/g1/NPANXXXXXX") in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type

Hi List, a number of our customers are reporting dropped calls. here is the config. 1 T100P T1 Card 1 Asterisk (Mid Nov build) T1 is signalled as a PRI(National) The card will only sync up if we clock, if we line side clock the card goes into yellow alarm and won't sync up. the only errors we see are framing slips. Around 2500 slips over a 18 hour period. (this was reported from

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a

In the process of turning up a new pri. Zttool indicates the T1 is ready with no alarms. asterisk*CLI> pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3

I'm experience echo on outgoing calls: Snom 200 ----> Asterisk ----> T100P ----> PRI ----> called party I am getting echo on the Snom 200 phone. The called party does not hear the echo. Since the PRI is digital, I don't really understand where the echo is coming from. I turned on echotraining, echocancellation=yes (128), and echowhenbridged in zapata.conf, to no avail.

I have seen this posted around the list a few times, but have never found my exact error or even a solution to this. I was getting this error every few seconds...... After moving the card to a different PCI slot and turning off some devices that are not needed. I was able to get the messages down to about every 10 minutes and the T1 card now is sitting in it's own intterupt. Apr 25 23:49:15

(to mods: sorry for any extra work I've put you through. Posting from my work account, as proper) Hello all, I'm back. I've got that lucky feeling that this will all work out very soon, and I hope that that is true. The issue: 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