[asterisk-users] Can't make outgoing calls (T100P)
Hi there, I have a new box setup using the latest version of FreePBX and the latest SVN of Asterisk 1.2 as of yesterday. Incoming calls from our PRI work fine. However, outgoing calls gives me the operator saying The call cannot be completed as dialed after two rings. Here's an outgoing call from extension 271: -- Executing Set(SIP/271-09f61dc0, OUTNUM=7883229) in new stack -- Executing Set(SIP/271-09f61dc0, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/271-09f61dc0, 0?customtrunk) in new stack -- Executing Dial(SIP/271-09f61dc0, ZAP/g0/7883229|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/7883229 -- Zap/1-1 is proceeding passing it to SIP/271-09f61dc0 -- PROGRESS with cause code 28 received -- Zap/1-1 is making progress passing it to SIP/271-09f61dc0 -- Hungup 'Zap/1-1' I've tried to find out what cause code 28 is with no luck. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us zapata.conf: [channels] language=en #include zapata_additional.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes callerid=asreceived echocancelwhenbridged=no echotraining=800 group=0 channel=1-10 We have 10 enabled lines from this PRI. Any help/suggestions are appreciated. Regards, -Darren ___ --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] SNMP Monitoring (Cacti)
Hello, Any luck with getting snmp monitoring of channel usage working with Cacti? Thanks, - Darren Callum McGillivray wrote: Hi, We use Cacti (an MRTG based monitoring tool), and I would also like to see how you set that up. Any chance you are willing to share ? Cheers, Callum Florian Overkamp wrote: Hi, -Original Message- I use MRTG to graph Active/Configured SIP channels and Active/Total PRI/ZAP channels, but I don't monitor the up/down status. You could probably Any chance you will share the mrtg setup you used for that ? How did you read out asterisk (via manager interface, tailing logfiles, or ... ?) How busy is your setup ? Thanks, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Manager
Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan management and assignment of scripts to extensions Voice Menu Manager - configuration of Auto Attendant and multi level voice menus Script Manager - creation of scripts for call handling (used by Extensions Manager) Conference Manager - configuration of conference rooms Configuration Editor - direct access to Asterisk configuration files Command Shell- web interface to Asterisk command line interface File Manager - intelligent upload and download for various configuration and support files ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Yet another faxing issue..
Hello, fax/ata(ht286) - asterisk/tdm04b - pstn fax machine I can fax out from the sip side, but I can't fax in from the PSTN side. When I try to send a fax, asterisk sees the call and show me this: Redirecting Zap/1-1 to fax extension Timeout on Zap/1-1 TCPDUMP doesn't show any activity to the extension that I configured to be the fax machine. Here's my config: extensions.conf ; ; Zap Fax ; exten = 8021,1,Dial(SIP/8021,20) exten = 8021,2,Hangup [incoming] exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,DigitTimeout(10) exten = s,4,ResponseTimeout(20) exten = s,5,Background(vm-extension) exten = fax,1,Goto(8021,30) exten = fax,2,Congestion exten = fax,102,Congestion exten = t,1,Hangup zapata.conf [channels] context=incoming switchtype=national signalling=fxs_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=1.5 txgain=1.5 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=4 faxdetect=incoming channel = 1-4 Yes, I realize faxing isn't reliable but I should at least get something. Any ideas? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MAX TNT SIP / Asterisk
Using Software version 10.1.0
Here's what I did:
1. Create a Media Profile (called voip)
name* = voip
active = yes
protocol-type = sip
[in MEDIA-GATEWAY/voip:voip-options]
packet-audio-mode = g711-ulaw
frames-per-packet = 2
silence-det-cng = no
ena-adap-jitter-buffer = yes
max-jitter-buffer-size = 19
initial-jitter-buffer-size = 2
voice-ann-dir = /current
voice-ann-enc = g711-ulaw
call-inter-digit-timeout = 6000
silence-threshold = 0
dtmf-tone-passing = inband
maxcalls = 672
rfc2833-payload-type = 96
g711-transparent-data = no
rtp-problem-reporting = { no 30 60 }
[in MEDIA-GATEWAY/voip:sip-options]
t1-timer = 500
t2-timer = 4000
invite-retries = 6
non-invite-retries = 10
primary-proxy = { x.x.x.x 5060 compact } (IP ADDRESS OF ASTERISK)
secondary-proxy = { 0.0.0.0 5060 compact }
registration-proxy = { x.x.x.x 5060 compact 1 } (IP ADDRESS OF
ASTERISK)
proxy-heartbeat = 0
proxy-failover-window = 60
reroute-on-proxy-failure = no
trusted-proxy =
unknown-ani =
blocked-ani =
privacy-proxy-require = disabled
cause-code-map = s
start-call-method = invite
trunk-group-options =
onhold-minutes = 0
support-100rel = disabled
internationalize = no
international-prefix = no
country-code =
national-destination-code =
local-number-ton = unknown-ton
call-transfer-method = ip-transfer
notify-timer = 0
invite-with-multiple-codecs = disabled
2. Configure Call Route for Digitam Modem card
admin get call-route {{{1 3 0}0}0}
[in CALL-ROUTE/{ { { shelf-1 slot-3 0 } 0 } 0 }]
index* = { { { shelf-1 slot-3 0 } 0 } 0 }
active = yes
trunk-group = 0
phone-number = 7299 (last 4 digits of your DID)
preferred-source = { { any-shelf any-slot 0 } 0 }
call-route-type = voice-call-type
cost = 0
3. Configure the T1 ports
default-call-type = dnis-or-voip
media-gateway = voip
I did this about 8 months ago and don't have my notes with me so I hope
I remembered everything. Give it a shot. Good luck
- Darren
On Tue, 2004-11-09 at 09:49, Tim Connolly wrote:
Do you have the TNT's config available? I'd love to see this work!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Bentley
Sent: Monday, November 08, 2004 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MAX TNT SIP / Asterisk
Have you attempted to use SIP? It's working quite well for me.
sip.conf
[maxtnt]
type=friend
host=xxx.xxx.xxx.xxx
dtmfmode=inband
callerid=MaxTNT maxtnt
context=toll-access
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=g729
allow=ulaw
extensions.conf
(xxx.xxx.xxx.xxx would be the address of your MaxTNT)
[toll-trunks]
;
; Outbound 1-nxx-nxx- goes via: PSTN
;
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _1NXXNXX,2,Hangup
[local-trunks]
;
; Outbound to nxx- goes via: PSTN
;
exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED],60)
exten = _NXX,2,Hangup
;
[local-access]
;
; Extensions that are this context are allowed to only call local PSTN
numbers and other extensions
;
include = extensions
include = local-trunks ; Access to Local numbers
[toll-access]
;
; Extensions that are this context are allowed to call local and long
distance PSTN numbers and other extensions
;
include = local-access ; Everything local-access has
include = toll-trunks ; Access to toll numbers
- Darren
On Mon, 2004-11-08 at 10:36, James Taylor wrote:
Your question indicates that there may be a better way...
???
I want to use the voice mail and extension features of Asterisk, and
sometimes there is this NAT problem that Asterisk seems to handle very
well.
I've been using H.323 with the TNT.
Do you have an alternate solution?
On Mon, 8 Nov 2004 10:41:31 -0500 (EST), [EMAIL PROTECTED] wrote:
On Tue, 2 Nov 2004, James Taylor wrote:
I can't get my MAX TNT to register with Asterisk.
TAOS 11.0.
SIP phone registeration show up in Asterisk like this:
sip:[EMAIL PROTECTED] and works.
The TNT shows up as:
sip:@ip_address.
Does anyone have this working?
Am I missing something here?
Where does the TNT get it's user name? Or, can it work without one?
It works without one.
Why do you need to register TNT to asterisk anyway?
--alex
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Re: [Asterisk-Users] MAX TNT SIP / Asterisk
Have you attempted to use SIP? It's working quite well for me. sip.conf [maxtnt] type=friend host=xxx.xxx.xxx.xxx dtmfmode=inband callerid=MaxTNT maxtnt context=toll-access qualify=yes reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw extensions.conf (xxx.xxx.xxx.xxx would be the address of your MaxTNT) [toll-trunks] ; ; Outbound 1-nxx-nxx- goes via: PSTN ; exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _1NXXNXX,2,Hangup [local-trunks] ; ; Outbound to nxx- goes via: PSTN ; exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _NXX,2,Hangup ; [local-access] ; ; Extensions that are this context are allowed to only call local PSTN numbers and other extensions ; include = extensions include = local-trunks ; Access to Local numbers [toll-access] ; ; Extensions that are this context are allowed to call local and long distance PSTN numbers and other extensions ; include = local-access ; Everything local-access has include = toll-trunks ; Access to toll numbers - Darren On Mon, 2004-11-08 at 10:36, James Taylor wrote: Your question indicates that there may be a better way... ??? I want to use the voice mail and extension features of Asterisk, and sometimes there is this NAT problem that Asterisk seems to handle very well. I've been using H.323 with the TNT. Do you have an alternate solution? On Mon, 8 Nov 2004 10:41:31 -0500 (EST), [EMAIL PROTECTED] wrote: On Tue, 2 Nov 2004, James Taylor wrote: I can't get my MAX TNT to register with Asterisk. TAOS 11.0. SIP phone registeration show up in Asterisk like this: sip:[EMAIL PROTECTED] and works. The TNT shows up as: sip:@ip_address. Does anyone have this working? Am I missing something here? Where does the TNT get it's user name? Or, can it work without one? It works without one. Why do you need to register TNT to asterisk anyway? --alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone use AdvancedVOIP ?
Has anyone used the Voip Billing System from http://advancedvoip.com/ ? They seem to also offer a billing solution for Interconnections. I'm curious if anyone has some experience using their software? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
Hi Gary, WireBill looks interesting. You mentioned that you are using the source code to build your own platform, but how does it hold up on its own? Can I ask what it can't do that requires you to build your own? Thanks, - Darren On Wed, 2004-08-04 at 08:14, Gary Carr wrote: That sigh will turn to cursing after a couple of months. We currently use Rodopi, have for 10 years but the inflexability is too much to deal with anymore so we are moving away from it. To what? I am also a cursed Rodopi owner. :-( Tom We bought the source code to wirebill and are building our own platform. Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
Well, we currently use Rodopi and are trying to find away to use it for our VOIP billing. However, because it's based on radius I'm unsure if it will be suitable for Asterisk. I was just curious if anyone else has used the 2 together before. - Darren On Sat, 2004-07-31 at 13:47, [EMAIL PROTECTED] wrote: Rodopi is a radius system. Just build your own using freeradius. Are you using Cisco that you need radius? Cheers Clive On Fri, 30 Jul 2004 11:44:14 -0700 Darren Bentley [EMAIL PROTECTED] wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rodopi Billing
Well, can anyone recommend a full featured ISP billing system that would handle VOIP/Asterisk? - Darren On Tue, 2004-08-03 at 11:09, Brian D'Arcy wrote: On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. OMG.. I had to support a rodopi installation myself for 2 years.. Closest I've ever come to suicide. While I have not managed another system but RODOPI, I have to say, there must be better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rodopi Billing
Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interface to generate Statements?
On Thu, 2004-07-08 at 13:15, Philipp von Klitzing wrote: Hi! Is there any downloadable software to generate Statements from the mysql call log? I am not sure what statements you want, but look here: http://www.voip-info.org/wiki-Asterisk+GUI Cheers, Philipp Yeah, I've looked there already. I guess I'm just thinking of a basic interface where you can generate invoices for customers. I guess I better start coding :) Thanks, - Darre ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
