[asterisk-users] ChanSpy() and Spygroup

2011-12-07 Thread Jeremy.Hellstrom
I am running an Asterisk 1.4.34 installation. I am trying to separate several SIP phones into two separate spygroups. These phones are making external calls as opposed to receiving incoming calls. Is there a place to assign a phone to a Spygroup other than when the call is initiated. I am tryin

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-09-29 Thread Jeremy.Hellstrom
Thanks for the additional comments, I though the timing was odd as well as I had thought it would be the provider that set the timing but it is currently working here and apparently at other sites. I'll let everyone know if it turns out to be incorrect somehow. -Original Message- From: as

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-09-28 Thread Jeremy.Hellstrom
I'd just like to thank everyone that helped me get this running. I thought I had a FAS PRI but it turns out it was NFAS so there was no dchannel on the second PRI. When getting the 2nd PRI changed, I received an email containing the cheat sheet TELUS employees use which I thought I should shar

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-08-30 Thread Jeremy.Hellstrom
The specific error message is as follows. _ Changing signalling on channel 24 from Unused to Hardware assisted D-channel DAHDI_CHANCONFIG failed on channel 24: Invalid argument (22) Did you forget that FXS interfaces are configured with FX

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-08-30 Thread Jeremy.Hellstrom
I tried those as you said, deleting my failed attempt. I've found that using hardhdlc=24 generates an error and reminds me that FXO uses FXS signalling and vice versa when running dadhi_restart, which seems to indicate that it is the wrong variable name. I also notice that if I change that variab

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-08-30 Thread Jeremy.Hellstrom
It is a half turned up PRI, so 1-12 should be correct? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: August 28, 2010 12:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-use

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-08-28 Thread Jeremy.Hellstrom
I'm not surprised both the conf file and myself are confused. I've pared things down in chan_dahdi.conf to ... _ [channels] spanmap => 1,1,0,esf,b8zs #include dahdi-channels.conf switchtype => national signalling => pri_cpe context => default ___

Re: [asterisk-users] TELUS British Columbia PRI Settings

2010-08-27 Thread Jeremy.Hellstrom
I see ... Chan_dahdi.c 2796 pri_find_dchan No D-channel available using Primary channel X as D-channel anyway. With X being whichever number I assigned to the D-channel in chan_dahdi and system.conf. Then when dialling I get an error 0 - unknown, which occurs when Asterisk tries to open a cha

[asterisk-users] TELUS British Columbia PRI Settings

2010-08-27 Thread Jeremy.Hellstrom
I am having some difficulties getting my Asterisk box to find the d-channel from a TELUS PRI and am waiting to hear back from one of their techs. In the meantime I thought I would check with the brilliant people of the mailing list. As I understand it is a T1 connection, not an E1 and I am u

[asterisk-users] Reinstalling Asterisk due to hardware changes

2010-08-06 Thread Jeremy.Hellstrom
My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card to put into my Asterisk box. I know from the wonderful O'Reilly book that the proper installation is Zaptel à libpri à Asterisk. Is it possible to simply reinstall in that order once I have installed the card and hav

Re: [asterisk-users] Using SIP to dial extension that will give anoutside line

2010-08-03 Thread Jeremy.Hellstrom
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Carlos Chavez Sent: Tue 8/3/2010 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using SIP to dial extension that will give anoutside line On Tue, 2010-08-03

[asterisk-users] Using SIP to dial extension that will give an outside line

2010-08-03 Thread Jeremy.Hellstrom
I am trying to add an Asterisk box to an Iwatsu ECS (Software Version 7.0 R.01) hopefully without using a physical T1/E1 card. Internally the SIP works fine, it is dialling an outside line that is giving me difficulties. One way that I think it might be possible is for an outbound call to connect