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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

...all 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, 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, Someone have used dialogic card D/600JCT-2E1? Does it work fine with asterisk? thanks Eduardo Gon?alves AceNet do Brasil

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 list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

Hi list, There's any E1 card I can use with Asterisk that support R2 signaling? Digium's E1400P doen's support it, right? thanks in advance Eduardo

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

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

Hi list, What paremeter can I change to control interdigit timing? Because my PSTN provider aren't receiving all the digits I dialed on Zap/g1. My Zap/g1 are an E1 (E400P) using E&M immediate sigalling. thanks in advance Eduardo

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'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

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 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 list, I'm trying to connect a cisco 2620 to my asterisk box using ISDN PRI. But I got some problems. zttool shows "no alarms" and "Internally clocked". Asterisk starts fine, but doesn't bring up d-channel. And when I try to make a call, asterisk shows: NOTICE[16401]: File app_dial.c, Line 516 (dial_exec): Unable to create channel of type 'Zap' At

Hi list, This morning, my asterisk box (PRI trunk) just stops to make calls. NOTICE[16401]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of type 'Zap without any reason. Now, asterisk doesn't brings up d-channel DEBUG[11276]: File chan_zap.c, Line 6410 (pri_dchannel): Got event No event (0) on D-channel for span 1 anyone could help? thanks Eduardo

Hi, We'he bought 4 g729 licenses. By the first time i installed, a ran the Registration program out of the /usr/src/asterisk directory. This worked fine, but after a few minutes, I read some docs, and reliaze that I need to run Registration from /usr/src/asterisk directory or might have problens after asterisk upgrade. So, a run registration again, this time from /usr/src/asterisk. But

Hi list, I have an asterisk box with a TE140P using two span's (ISDN PRI euroisdn). Today, the box was running ok, but twice it's just stops to make calls and the CLI got flooded with the message "D-channel on span 1 up" and "D-channel on span 2 up". So I stopped asterisk, unload the module, load the module, ant them started asterisk again. After this, asterisk

Hi list, Does anyone use the .deb package of asterisk? Is it stable? woks fine? thanks Eduardo

Hi list, I have an X100P to place some outgoing calls. But sometimes zttool shows a red alarm and after I unplug and plug the line cable, the alarm is cleared. Sometimes dialing works and sometimes not. I suspect it's a timing problem. Could someone point me on how to configure timing parameters for an X100P? thanks in advance Eduardo