Re: [asterisk-users] how to configure zaptel for incoming call
Sanchal, You may want to make sure that you have "immediate=no" set for your E1 channels in zapata.conf. This makes asterisk wait for digits, rather than skipping to the "s" extension on incoming calls. --TS >>> <[EMAIL PROTECTED]> 7/30/2007 4:14 AM >>> Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming channel=1-15,17-31 What configuration changes is to be done for landing of call to asterisk PBX when dialled from E1 card running application. I was trying to dial out from E1 card running application with extension number 114 and added the following lines in extensions.conf of asterisk configuration files exten=>114,1,Answer but asterisk debugging console is giving the error message -- Extension 's' in context 'channelbank' from '' does not exist. Rejecting call on channel 0/1, span 1 Can anybody tell me how to handle the configuration files for extension number to be called from E1 card running application. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Inbound Voice Quality - Speed Change
Looks like output from the 'lsmod' command. >>> "Lacy Moore - Aspendora" <[EMAIL PROTECTED]> 3/27/2007 11:34 PM >>> On 3/27/07, Jim Duda <[EMAIL PROTECTED]> wrote: > ztdummy 4424 0 > rtc11156 1 ztdummy > zaptel178084 1 ztdummy > crc_ccitt 2016 1 zaptel > Ok, this is a dumb question, but what is that output from? What distribution of Linux are you using? I've never had to change anything related to the kernel. I use CentOS, though. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys
Somehow, I ended up with BootROM 3.2.3.0002 (which as far as I can tell hasn't been released yet...) and SIP version 2.1.0.2708. I do see the "sluggish buttons" from time to time. Rarely, but I do see it. --TS >>> Mike <[EMAIL PROTECTED]> 4/12/2007 9:59 AM >>> Exactly. It's a weird issue, and I can't imagine what the problem is, except maybe for bad phones (but then again, why would the phones be only bad with 2.x?) UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest have? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Thursday, April 12, 2007 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: > Jim King wrote: >> I've seen an issue like this from time to time on 601s, even with the >> latest firmware. Not just the softkeys, but also the dial keys. The >> phones seem to "run slow" sometimes, failing to respond to a key >> press right away but getting to it eventually. It usually clears up >> after a few seconds. >> Also, I've noticed that the 601s sometimes ignore key presses >> altogether, just as you describe. >> I have not yet found a solution for this problem... > > Try setting this in sip.cfg: dialplan.impossibleMatchHandling="1" > > I suspect it is either 0 or 2 now. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
In my case, I have a single PRI coming into a single port PRI card on my Asterisk box. My "old" fax numbers (prior to our switch to PRI) are DIDs on that PRI. Using IAXModem+HylaFAX, I can recieve faxes without having seperate POTS lines for faxes, or an external fax board, or a multiport T1 card. Adding new fax numbers involves assigning a new DID...so adding additional fax "lines" doesn't cost anything (again, IAXModem is free...) --TS >>> Olivier <[EMAIL PROTECTED]> 04/17/07 2:19 PM >>> 2007/4/16, Stephen Bosch <[EMAIL PROTECTED]>: > > > It's not entirely clear to me why people continue to cling to the idea > that Asterisk should handle faxing also. What's the benefit? Hylafax is > great, and you can even use it on the same machine. The benefit, I guess, is to save a dedicated line and not changing incoming fax numbers, as you cannot port them individually. But you're right to point it also has drawbacks ... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users