similar to: E1 card with R2 signaling

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo

Hi, Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? thanks Eduardo

Hi, I have an ansterisk and a cisco 827-4v registered to a Gatekeeper. asterisk has two extensions: exten => 223,1,Dial,OH323/BYEXTENSION@827PD exten => 730,1,Dial(IAX/eduardo@10.0.11.103) (IAX are working well) When I try to call each other, gnugk shows a ARJ: ARJ|10.0.11.112:1720|223:dialedDigits|730:dialedDigits|false|resourceUnavailable I think this could be a codec

Hi list, I'm trying to set up a TDM40B, but modprobe returns the fowling errors: asterisk:~# modprobe wcfxs ZT_CHANCONFIG failed on channel 1: Invalid argument (22) /lib/modules/2.4.18/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.18/misc/wcfxs.o: insmod wcfxs failed Could someone give me a help? Thanks in advance Eduardo

Does the Digium's E1 card work with CASR2 Signaling ??? or only PRI?

Hi, I'm abaut to install asterisk and I want to know if buying an E400P (Quad Span E-1 Interface) from digium my linux box will be ready (of course, after configure it) to work with PSTN and my already in use PBX (E1 interface), or I'll need to buy additional DSP's cards or whatever? thanks Eduardo Gon?alves AceNet do Brasil LTDA 55 11 3365-0500

I am experiencing this: 1 - A,B,C are SIP users logged on QUEUEA with ringall strategy 2 - I call QUEUEA 3 - A,B,C start ringing 4 - nobody answer.... 5 - D logs on the QUEUEA 6 - D doen's receive any call, but A,B,C are still ringing How can i avoid that? I'd like that when D joins the QUEUEA it will immediately receive the call that is still ringing on other users... Thanks in

Has anyone found a solution for asterisk and r2 signaling ? Steve Underwood had given some information saying he had a working asterisk working. I need it to work with Argentina R2 signaling Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040915/3289f694/attachment.htm

Hello, I am in Argentina configuring an E1 in R2 and have some inconveniences. When the call takes the salient line, I receive an error of signaling. In * I have loaded the following: - libr2 installed - The [ar] in zonedata.c and indications.conf is configured - zaptel.conf o: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,cas,hdb3 #span1 bchan=1-15,17-31 dchan=16 #span2 cas=32-62:1001

Hi, Can someone send me a sample on how to get an E400P working with ISDN PRI signaling? What conf. files must I edit? thanks in advance Eduardo

I heard that asterisk support r2 signaling. I'm try to test r2 using e100p. How should I configure zaptel.conf, zapata.conf? And if I want to modify source for customization, where should I start? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040608/6392facd/attachment.htm

Anyone configured a E1 PRI in Kuala Lampur, Malaysia with TM Malaysia? What signaling did they provide, framing, formatting? primary-4ess Lucent 4ESS switch type for the U.S. primary-5ess Lucent 5ESS switch type for the U.S. primary-dms100 Northern Telecom DMS-100 switch type for the U.S. primary-dpnss DPNSS switch type for Europe primary-net5 NET5 switch type for UK,

Hello everyone. I am looking for a E1 PRI card which supports network side signaling not CPE. The main idea is to connect an plain old E1 compliant PBX which doesn't have an VoIP module to the newly created VoIP infrastructure. Could we use a Digium TE122P or something other to resolve this situation? Thanks in advance. Dimitar -------------- next part -------------- An HTML attachment was

Hi friends, I'm testing a E400P with PRI signaling. When I start asterisk (-vvc), i get the fowlling messages in /var/log/asterisk/message: May 12 12:42:06 WARNING[1024]: File chan_iax2.c, Line 4931 (set_config): Ignoring port for now May 12 12:42:06 WARNING[1024]: File chan_zap.c, Line 6525 (load_module): Ignoring ocancel And 'show channels' returns no channels. Can anyone help

Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as

and a little more on fax config support ----- Original Message ----- From: "Tilghman Lesher" <tilghman@mail.jeffandtilghman.com> To: <asterisk-users@lists.digium.com> Sent: Monday, August 11, 2003 4:48 PM Subject: Re: [Asterisk-Users] Fax Handled > On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote: > > On Mon, 11 Aug 2003 15:15:08 -0500 > > >

Hullo everyone, Iam new to Asterix but am trying to get R2 signalling working with a Digium E100P E1/PRA Card. My Telecom doesn't have an implementation of ISDN PRI yet, so it's R2 that is supported. Could anyone have a patch to fix this for me. Am told it has something to do with Assembly Language programming. Please help me fix this. Regards, Nicholas

Hi Moy, Bottom line up front (yes I'm military):How do I now configure some of my spans as normal Euro ISDN and a few with R2? Do I use a combination of /etc/asterisk/unicall.conf and /etc/zaptel.conf or just unicall.conf for everything? See below for the rest of the details and thanks in advance for advice! I downloaded your astunical-1.4.16 and have been studying it. I have a machine

Hello, I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps. With asterisk, what's the bit rate used by speex? Is it possible to have asterisk using speex at less than 10kbps of bit rate? If not, Is it dificult to implement? thanks in advance Eduardo