similar to: Sending Set Asynchronous Balanced Mode Extended

Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI:

I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe E&M. I have done E&M wink but have no idea about E4 Superframe E&M and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

Hi, I hope someone can help me with this.... Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2 System is located in Australia, so as technologies go, I believe it is twist on the euro standard for the E1 signalling. Here is the situation. The TDA E1 card is set in cross over mode and I am using a functional

Hi, I have the following happen to me after the restart of one of my servers: out of my 3 PRIs (all configured with the same technical settings), the last one isn't coming back. It's underutilized (chances it didn't get a call since my reboot), if it makes a difference . The PRI goes from provisioned to unprovisioned, and I get this regularly: [Jun 22 09:03:48]

I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the

Hi Guys, Running the following with a Sangoma A101D PRI card: *Asterisk 1.4.21.2* *LibPRI version: 1.4.10* No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show any activity. Problem goes away on restart of the system or maybe asterisk. I see post about upgrading Libpri to 1.4.10.2 and then I see posts that even that didn't work. Anyone can weigh in this please?

All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: jweisman@ibell.net Thanks, Jon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to install an E1 PRI, and I need it working by Monday, but although everything seems ok, I get no response to calls. When I make a pri extense debug on span 1, I repeatedly get the following: Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 999 EA: 1 M3: 3

Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on

I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello List, since some days i run into the problem that one span on a TE407P is not comming up correctly. With intense debug on that span i get: < [ 02 01 7f ] < Unnumbered frame: < SAPI: 00 C/R: 1 EA: 0 < TEI: 000 EA: 1 < M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] < 0 bytes of data

Hi Guys, Anyone might know why this error keeps showing up and inbound/outbound is not working on a Bell PRI with Sangoma A101D? -- Got SABME from network peer. Sending Unnumbered Acknowledgement No calls can be made inbound/outbound. Keeps repeating. No alarms ON and no changes been made to the system. Stopped all a sudden. Asterisk CLI doesn't show anything with full verbose for both

Hello list, i've got some problems getting a E1 line with a E100P up and running (germany). # cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS/CRC4 YELLOW RED "YELLOW RED" sounds not so good. When launching asterisk, enabling pri debug on that span, i see outgoing attempts: ;---------- snip -------- > [00 01 7f ] > Unnumbered

On Fri, 2006-10-13 at 12:52:38 -0400, Jon Weisman <jweisman@ibell.net> wrote: > Hi All, Anyone know how to generate random numbers in the > dial plan? I've tried using the RAND function but it doesnt > work. Basically I need to generate a random 5 digit number > everytime a particular extension is dialed and then save that > into

Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point,

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

hot to asterisk multiport...??? example 5060, 5061, 5080 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070815/8a515309/attachment.htm

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others