[Asterisk-Users] Problems with ZAP dial timeout

2006-05-31 Thread Ryan Laginski
Hi, I'm having a problem with the timeout option when dialing a ZAP channel. The goal is to ring a number for 15 seconds, if no one picks up, go to voicemail. The dial command is: exten = s,1,Dial(ZAP/1/613555,15) exten = s,2,VoiceMail(u1) exten = s,102,VoiceMail(b1) The call will continue

Re: [Asterisk-Users] seg fault when skinny phone answers

2006-03-06 Thread Ryan Laginski
receiving a call from external zap. I changed the earlyrtp=ringout as per a mailing list thread, and viewed the debug output on setting 10. Nothing obvious stood out.Regards,-RyanOn 3/5/06, Michiel van Baak [EMAIL PROTECTED] wrote: On 20:19, Sat 04 Mar 06, Ryan Laginski wrote: Downgrade to 1.0.10

Re: [Asterisk-Users] seg fault when skinny phone answers

2006-03-04 Thread Ryan Laginski
Downgrade to 1.0.10. I was unable to get the 12sp+ to work reliably in 1.2.0-1.2.4 and had the same problem.On 2/20/06, btb [EMAIL PROTECTED] wrote:hello-i'm having trouble completing a connection between an older skinny phone (12sp+) and a soft sip phone (x-lite).the skinny phone appears to

[Asterisk-Users] Core dumps since 1.2.0

2005-12-08 Thread Ryan Laginski
Hi,Ever since upgrading to 1.2.0, Asterisk occasionally core dumps. I'm currently on 1.2.1 with the same problem.It crashes when an incoming call (zap) dials an extension. It will ring the extension short, then crash. Here is the backtrace:#0 0x40187e06 in mallopt () from /lib/libc.so.6(gdb)

[Asterisk-Users] Restart DISA from the beginning

2005-07-06 Thread Ryan Laginski
Hi, Is there a way to restart the DISA to the enter phone number? For instance, Bell Calling Cards let you hit # at any point which lets you enter another number to call. This is useful to reduce the number of digits dialed and to utilize per-minute calls. I was not able to find anything on the

[Asterisk-Users] DVG-1120S does not show callerid Name and resets time

2005-05-27 Thread Ryan Laginski
Hi, I am having problems with callerid name and the time with my dvg-1120S. Every time I receive a call, it reverts the phone to January 1st 12:00am. I've looked everywhere in the browser and telnet configuration to change this. Also, it never shows the name of the caller. I've even tried forcing

[Asterisk-Users] DVG-1120S no call display name and time

2005-03-19 Thread Ryan Laginski
Hi, I am having problems with callerid name and the time with my dvg-1120S. Every time I receive a call, it reverts the phone to January 1st 12:00am. I've looked everywhere in the browser and telnet configuration to change this. Also, it never shows the name of the caller. I've even tried forcing

Re: [Asterisk-Users] DVG-1120 questions

2005-03-16 Thread Ryan Laginski
Hi, I'm able to see the callerid number, but not the name. I've tried your suggestion by removing the quotes, but it doesn't help. Any suggestions? Thanks, -Ryan On Fri, 11 Mar 2005 17:53:17 -0600, Eric Wieling [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I upgraded a DVG-1120M to a

Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

2005-03-04 Thread Ryan Laginski
not be that interested in your business. -mark On Mar 2, 2005, at 11:06 PM, Ryan Laginski wrote: Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves

Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

2005-03-02 Thread Ryan Laginski
Hi, I am experiencing the same problem as you. Ringback works great with the pstn or any other voip provider, but not with livevoip. I've just upgraded to 1.0.6 to see if that resolves the problem, but it has not. Please post back if you find a solution, I'll do the same. Thanks, -Ryan On Wed, 2

Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

2005-02-01 Thread Ryan Laginski
I sent the same email to [EMAIL PROTECTED], and got a response within minutes. It's now working. Previously, I sent it to [EMAIL PROTECTED] they don't check that address, eventhough it's listed on their site. -ry On Sun, 30 Jan 2005 10:15:41 -0500, Ryan Laginski [EMAIL PROTECTED] wrote: Hi, I

Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

2005-01-30 Thread Ryan Laginski
Hi, I made the mistake of ordering an 800 number as well. I have the same problem you have, asterisk registers, but I get a fast busy when dialing the number. Checking the cdr, and it shows the call has been placed, but for 0.0 minutes. I have yet to hear from them as well. Reading other posts,

[Asterisk-Users] Restart in the DISA to the beginning

2005-01-26 Thread Ryan Laginski
Hi, Is there a way to restart the DISA to the enter phone number? For instance, Bell Calling Cards let you hit # at any point which lets you enter another number to call. This is useful to reduce the number of digits dialed and to utilize per-minute calls. I was not able to find anything on the

Re: [Asterisk-Users] Cisco VIP30

2005-01-14 Thread Ryan Laginski
Hi, I've never used those instructions, this is my skinny.conf, and I was able to connect 3 12 sp+ (apparently the exact same as a VIP30 minus the extra buttons) with different firmwares. ; ; Skinny Configuration for Asterisk ; [general] port = 2000 ; Port to bind to, default

Re: [Asterisk-Users] IAX2 provider in Montreal, Canada

2005-01-12 Thread Ryan Laginski
I've heard videotron is testing a mgcp device that has the cable modem, wireless router, and 12hour battery, with the public. Although it's not iax2, it's probably still worth looking into it. I use iax.cc (sixtel) and talknet.ca for an Ottawa dids. -ry On Mon, 10 Jan 2005 13:03:54 -0500,

[Asterisk-Users] DISA restart from begining

2004-12-23 Thread Ryan Laginski
Hi, Is there a way to restart the DISA to the enter phone number? For instance, Bell Calling Cards let you hit # at any point which lets you enter another number to call. This is useful to reduce the number of digits dialed and to utilize per-minute calls. I was not able to find anything on the

RE: [Asterisk-Users] Callerid is recieved by fxo, but sometimes not passed to extensions

2004-11-11 Thread Ryan Laginski
Hi Jim, Thanks for your response. I do wait after the second ring. Regards, -Ryan On Thu, 2004-11-11 at 12:35, Jim Van Meggelen wrote: Are you waiting until the start of the second ring cycle before answering the phone? CLID information is sent in-band between the first and second ring

[Asterisk-Users] Callerid is recieved by fxo, but sometimes not passed to extensions

2004-11-10 Thread Ryan Laginski
Hi, I'm having a problem with callerid. It is recieved fine by the fxo (it appears in the cdr, and voicemail app gets it fine), but it is passed to the internal phones works about 25% of the time. The internal phones are all analog, a dvg-1120M (mgcp firmware) and a quicknet phonejack. There

[Asterisk-Users] Corrupt Callerid Data

2004-05-17 Thread Ryan Laginski
Hi, The incoming caller id on the X101P always comes up scrambled except when there is no name, just a number. Usually a cellphone would do this, and the number is perfect. I was reading posts about using ztmonitor to capture the spill and listening to it. The resulting file is alway 0 bytes...

Re: [Asterisk-Users] Cisco 12SP+

2004-05-14 Thread Ryan Laginski
exactly. The P002 is a cisco model, then the F204 is the firmware on my phone. When you boot the phone, you'll see the firmware version (mine says: F2.04, hence the F204). Hopes this help. -Ryan On Fri, 2004-05-07 at 09:44, Jakub Klausa wrote: On Tue, May 04, 2004 at 07:32:20PM -0400, Ryan Laginski

Re: [Asterisk-Users] Cisco 12SP+

2004-05-04 Thread Ryan Laginski
Hi Paul, To my knowledge, you can't change the image on them. I recently bought 3 of them, and we help from this list, I was able to connect them to my asterisk server. However, they are not fully functional. I can make calls and hear calls, but I'm muted. I'm looking for a solution. The protocol