similar to: send DTMF digits

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, 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

Hi list, I have an asterisk box with 8 zap channels (E400P, only one span, E&M siginaling). And sometimes on the console, these messages apear about some channels: -- Starting simple switch on 'Zap/4-1' -- Starting simple switch on 'Zap/5-1' And then: -- Hungup 'Zap/4-1' -- Hungup 'Zap/5-1' This ocurs with ramdom channels, sometimes all the

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, 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 have a situation where I am receiving DID calls using Immediate Start Pulse signalling on a Loop Start trunk. The line terminates on a Newbridge Mainstreet 3624 channel bank, which provides battery etc. The channel is converted and routed to Asterisk. The lines are configured as follows: /etc/asterisk/zapata.conf ; Channels 1-24 service MainStreet 3624 channel bank context=infrom-did group=1

To answer my own question for the record: The relevant timing parameters in zaptel.h are #define ZT_MINPULSETIME (15 * 8) /* 15 ms minimum */ #define ZT_MAXPULSETIME (100 * 8) /* 150 ms maximum default, lowered to 100ms */ #define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50) And the pulse detecion loop that consumes these parameters begins at line 4866 of zaptel.c The

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,

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

Greetings, According to the FAQ "When you load the module and have no circuit/channel bank the LED's should flash red" I get the "knight rider" lights before the module loads, but after the modules are loaded I don't get any lights, other equipment plugged in or not. The card has been on my shelf a while so I wonder if it needs a firmware upgrade ? Can the cards

Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7

Hello, Is there any configuration in zapata.conf for fax detection (or transmission)? When I try to send a fax trought asterisk, the line 'Fax Handled:' is always set to "no". The scenario is: [ata186]---sip---[asterisk]---e1 E&M---[pstn] Fax sometimes goes without problem and sometimes the fax machine can't send the fax. Thanks Eduardo

Hello, I have the fowling scenario: fxs[asterisk1]-----iax-----[asterisk2]e1----e&m---PSTN I want to know the steps to transmit fax from a machine connected to the fxs to a fax machine on the PSTN. The same for dial-up's. Is it possible only with a/ulaw ? What configs I need in asterisk1? Thanks in advance Eduardo

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

I just recently purchased an E400P card and am attempting to get it working. My requirements are for 2 ports to run Q.931 and the other 2 ports to run R2 signalling with DTMF inband. The card will operate in a drop and insert fashion with calls arriving on the ISDN side and being terminated on the R2 lines. There don't seem to be any examples out there of how to set up this card for R2

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

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answered a phone on the

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. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --