[asterisk-users] Re: ECHO Cancellation in SIP Calls
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with echocancel=... and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP ! Are you hearing the echo, or is the far end party? I can hear the Echo, the end party never got this problem. ___ --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] dialplan issue - 1 0 should be evaluated false
Maybe a missing space between expr1 and the sign on extension s
priority 3 ?
[]'s
MM
-Original Message-
From: Esteban Guana-Jarrin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Fri, 27 Oct 2006 15:11:33 +1000
Delivered: Fri, 27 Oct 2006 01:51:20
Subject:[asterisk-users] dialplan issue - 1 0 should be evaluated false
Helo List,
Sorry I missed the rest of my email in my previous post. Please see below.
I'm having an issue using the AND () operator evaluation in the code of my
dialplan. The dial plan is coded to detect inbound DTMF digits from callers.
key 1 is equivalent to yes and key 2 is equivalent to no in two
diferent contexts in the dial plan. When a caller presses 1, yes is passed
as a value in a varialble and same when 2 is pressed a no is passed.
When the values are both set to yes from previous contexts in the
dialplan; the evaluation of the AND operation in the following expression is
true and works fine jumping to the correct priority within the context,
exten = s,3,Gotoif($[${test11} = yes] $[${test12} = yes]?7:4)
Debug output
-- Goto (test-check,s,1)
-- Executing NoOp(SIP/123-e131, yes) in new stack
-- Executing NoOp(SIP/123-e131, yes) in new stack
-- Executing GotoIf(SIP/123-e131, 1 1?7:4) in new stack
-- Goto (test-check,s,7)
-- Executing System(SIP/123-e131, mail -s Test=Yes
[EMAIL PROTECTED]) in new stack
-- Executing Goto(SIP/123-e131, 14) in new stack
-- Goto (test-check,s,14)
-- Executing Playback(SIP/123-e131, end) in new stack
The issue occurs when the values are passed with different values. For
instance when the variables test11 and test12, shown below, are set with
values yes and no respectively. The AND operation is evaluated as true, as
seen in the debug output below, it jumps to priority 7 within the context
instead of jumping to priority 4 as it should be the case when evaluated as
false as it should,
exten = s,3,Gotoif($[${test11} = yes] $[${test12} = yes]?7:4)
exten = s,4,Gotoif($[${test11} = yes] $[${test12} = no]?9:5)
Debug output:
-- Executing NoOp(SIP/123-1c20, yes) in new stack
-- Executing NoOp(SIP/123-1c20, no) in new stack
-- Executing GotoIf(SIP/123-1c20, 1 0?7:4) in new stack
-- Goto (test-check,s,7)
-- Executing System(SIP/123-1c20, mail -s Test=Yes
[EMAIL PROTECTED]) in new stack
-- Executing Goto(SIP/123-1c20, 14) in new stack
-- Goto (test-check,s,14)
-- Executing Playback(SIP/123-1c20, end) in new stack
Following is the rest of code for the context processing the AND operations,
[test-check]
exten = s,1,NoOp(${test11})
exten = s,2,NoOp(${test12})
exten = s,3,Gotoif($[${test11} = yes] $[${test12} = yes]?7:4)
exten = s,4,Gotoif($[${test11} = yes] $[${test12} = no]?9:5)
exten = s,5,Gotoif($[${test11} = no] $[${test12} = no]?11:6)
exten = s,6,Gotoif($[${test11} = yes] $[${test12} = yes]?13:15)
exten = s,7,System(mail -s Test=Yes [EMAIL PROTECTED])
exten = s,8,Goto(14)
exten = s,9,System(mail -s Test=yes,no [EMAIL PROTECTED])
exten = s,10,Goto(14)
exten = s,11,System(mail -s Test=no [EMAIL PROTECTED])
exten = s,12,Goto(14)
exten = s,13,System(mail -s Test=no,yes [EMAIL PROTECTED])
exten = s,14,Playback(ExetelIVR/end)
exten = s,15,Goto(ack-message,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
Can someone please assist and explain how the AND operation works in
Asterisk?
Regards,
Paul
_
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Re: [asterisk-users] dialplan issue - 1 0 should be evaluated false
Try this:
exten = s,3,Gotoif($[$[${test11} = yes] $[${test12} = yes]]?7:4)
exten = s,4,Gotoif($[$[${test11} = yes] $[${test12} = no]]?9:5)
exten = s,5,Gotoif($[$[${test11} = no] $[${test12} = no]]?11:6)
exten = s,6,Gotoif($[$[${test11} = yes] $[${test12} = yes]]?13:15)
A $[] for the entire stuff.
[]'s
MM
-Original Message-
From: Esteban Guana-Jarrin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Fri, 27 Oct 2006 15:11:33 +1000
Delivered: Fri, 27 Oct 2006 01:51:20
Subject:[asterisk-users] dialplan issue - 1 0 should be evaluated false
Helo List,
Sorry I missed the rest of my email in my previous post. Please see below.
I'm having an issue using the AND () operator evaluation in the code of my
dialplan. The dial plan is coded to detect inbound DTMF digits from callers.
key 1 is equivalent to yes and key 2 is equivalent to no in two
diferent contexts in the dial plan. When a caller presses 1, yes is passed
as a value in a varialble and same when 2 is pressed a no is passed.
When the values are both set to yes from previous contexts in the
dialplan; the evaluation of the AND operation in the following expression is
true and works fine jumping to the correct priority within the context,
exten = s,3,Gotoif($[${test11} = yes] $[${test12} = yes]?7:4)
Debug output
-- Goto (test-check,s,1)
-- Executing NoOp(SIP/123-e131, yes) in new stack
-- Executing NoOp(SIP/123-e131, yes) in new stack
-- Executing GotoIf(SIP/123-e131, 1 1?7:4) in new stack
-- Goto (test-check,s,7)
-- Executing System(SIP/123-e131, mail -s Test=Yes
[EMAIL PROTECTED]) in new stack
-- Executing Goto(SIP/123-e131, 14) in new stack
-- Goto (test-check,s,14)
-- Executing Playback(SIP/123-e131, end) in new stack
The issue occurs when the values are passed with different values. For
instance when the variables test11 and test12, shown below, are set with
values yes and no respectively. The AND operation is evaluated as true, as
seen in the debug output below, it jumps to priority 7 within the context
instead of jumping to priority 4 as it should be the case when evaluated as
false as it should,
exten = s,3,Gotoif($[${test11} = yes] $[${test12} = yes]?7:4)
exten = s,4,Gotoif($[${test11} = yes] $[${test12} = no]?9:5)
Debug output:
-- Executing NoOp(SIP/123-1c20, yes) in new stack
-- Executing NoOp(SIP/123-1c20, no) in new stack
-- Executing GotoIf(SIP/123-1c20, 1 0?7:4) in new stack
-- Goto (test-check,s,7)
-- Executing System(SIP/123-1c20, mail -s Test=Yes
[EMAIL PROTECTED]) in new stack
-- Executing Goto(SIP/123-1c20, 14) in new stack
-- Goto (test-check,s,14)
-- Executing Playback(SIP/123-1c20, end) in new stack
Following is the rest of code for the context processing the AND operations,
[test-check]
exten = s,1,NoOp(${test11})
exten = s,2,NoOp(${test12})
exten = s,3,Gotoif($[${test11} = yes] $[${test12} = yes]?7:4)
exten = s,4,Gotoif($[${test11} = yes] $[${test12} = no]?9:5)
exten = s,5,Gotoif($[${test11} = no] $[${test12} = no]?11:6)
exten = s,6,Gotoif($[${test11} = yes] $[${test12} = yes]?13:15)
exten = s,7,System(mail -s Test=Yes [EMAIL PROTECTED])
exten = s,8,Goto(14)
exten = s,9,System(mail -s Test=yes,no [EMAIL PROTECTED])
exten = s,10,Goto(14)
exten = s,11,System(mail -s Test=no [EMAIL PROTECTED])
exten = s,12,Goto(14)
exten = s,13,System(mail -s Test=no,yes [EMAIL PROTECTED])
exten = s,14,Playback(ExetelIVR/end)
exten = s,15,Goto(ack-message,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
Can someone please assist and explain how the AND operation works in
Asterisk?
Regards,
Paul
_
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Re: [asterisk-users] Re: SIP v IAX2
On Thu, 2006-10-26 at 13:08 -0700, Martin Joseph wrote: On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said: Since they are incorporated in a single product which is doing the configuration, consistency where possible would be good... That product is designed to link the two things together, are you suggesting lowest common denominator configuration? Surely the best is to exploit each to it's maximum to achieve that goal even if it does make for slight differences in configuration, it is the day to day phone user experience that really matters. In computing inconsistances exist everywhere it is the job of a sysadmin to sort all this out so that it is transparent to the users. Sorry for the rant, I'm just going to a client to sort out the problems created, yet again, by his incompetent so called sysadmin. -- Dave Cotton [EMAIL PROTECTED] ___ --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
[asterisk-users] dialplan issue - 1 0 should be evaluated false
MM,
The $[] made it work. Thanks a lot for your assistance.
obligado :)
See the debug output below.
-- Executing NoOp(SIP/123-d14f, no) in new stack
-- Executing NoOp(SIP/123-d14f, yes) in new stack
-- Executing GotoIf(SIP/123-d14f, 0?7:4) in new stack
-- Goto (test-check,s,4)
-- Executing GotoIf(SIP/123-d14f, 0?9:5) in new stack
-- Goto (test-check,s,5)
-- Executing GotoIf(SIP/123-d14f, 0?11:6) in new stack
-- Goto (test-check,s,6)
-- Executing GotoIf(SIP/123-d14f, 1?13:15) in new stack
-- Goto (test-check,s,13)
-- Executing System(SIP/123-d14f, mail -s Test=no|yes
[EMAIL PROTECTED]) in new stack
-- Executing Playback(SIP/123-d14f, end) in new stack
-- Executing NoOp(SIP/123-9fb1, yes) in new stack
-- Executing NoOp(SIP/123-9fb1, no) in new stack
-- Executing GotoIf(SIP/123-9fb1, 0?7:4) in new stack
-- Goto (test-check,s,4)
-- Executing GotoIf(SIP/123-9fb1, 1?9:5) in new stack
-- Goto (test-check,s,9)
-- Executing System(SIP/123-9fb1, mail -s Test=yes|no
[EMAIL PROTECTED]) in new stack
-- Executing Goto(SIP/123-9fb1, 14) in new stack
-- Goto (test-check,s,14)
-- Executing Playback(SIP/123-9fb1, end) in new stack
Try this:
exten = s,3,Gotoif($[$[${test11} = yes] $[${test12} = yes]]?7:4)
exten = s,4,Gotoif($[$[${test11} = yes] $[${test12} = no]]?9:5)
exten = s,5,Gotoif($[$[${test11} = no] $[${test12} = no]]?11:6)
exten = s,6,Gotoif($[$[${test11} = yes] $[${test12} = yes]]?13:15)
A $[] for the entire stuff.
[]'s
MM
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[asterisk-users] Re: Cheapest way to determine channels in a group from outside asterisk?
They aren't zap interfaces unfortunately. They are SIP/IAX channels started from originate and the manager API. Lenz wrote: why not using a zap show command and parse the results externally? l. On Thu, 26 Oct 2006 13:12:46 +0200, Nick Adams [EMAIL PROTECTED] wrote: I need to determine the number of active calls in a group from outside of Asterisk. Currently I poll the manager API and parse the channel status list but this is becoming too expensive on CPU. What are my options? What is considered standard practice ? Update a DB field? Poll the manager api? Use an asterisk -rv 'some command' call? --Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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
[asterisk-users] Re: ECHO Cancellation in SIP Calls
On 2006-10-26 23:02:40 -0700, Stefan Agethen [EMAIL PROTECTED] said: Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP-Asterisk-SIPProvider-TELEKOM-ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with echocancel=... and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP ! Are you hearing the echo, or is the far end party? I can hear the Echo, the end party never got this problem. If you can adjust the microphone gain on your local handset down, you may find that reduces audio coming back from the far ends mic (ie If you audio is very loud it could be bleeding across from the far end ear piece to the far end mic. Worth a try anyhow... Marty ___ --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] asterisk not detecting hangup
I've enabled those options but it's the same. On 10/25/06, Maxi Belino [EMAIL PROTECTED] wrote: i'm having similar problems (if you find out the solution please post it) did you try enabling 'callprogress' or 'busydetect' in zapata.conf ? Maxi 2006/10/23, Arkaitz [EMAIL PROTECTED]: Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Identified as: *CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard X101P Board 1 OK 0 0 0 And the following lines in zapata.conf(for spanish lines): answeronpolarityswitch=yes hanguponpolarityswitch=yes The problem is that although the calls work correctly the system is unable to detect a pstn hangup and it keeps running even when the other side is calling to another number(not an asterisk ones, asterisk line keeps busy) Any hint? Thanks for your time -- Arkaitz ___ --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 -- Arkaitz ___ --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
[asterisk-users] Re: Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For my home Asterisk setup I have a single PSTN line, and then I use a variety of different voip providers. I use two different providers for my DID's (one toll free, and one normal). I use yet a different provider for terminating outgoing calls. So, when making an outgoing call via voip, what number should I use to identify myself? I currently use the number of my PSTN line, since that is our public inbound number. Hi John! I have same situation, and I certainly agree about everything you said. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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
[asterisk-users] chan_skype license?
Hi guys, is there a comment from digium on the license of chan_skype? I could not find the GPL_KEY in the precompiled module, and they don't release the source. So i'm guessing, they'd need a commercial license...? Regards, Andreas _ Become a fitness fanatic @ http://xtramsn.co.nz/health ___ --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] chan_capi and bristuff
On Fri, 27 Oct 2006, Michiel van Baak wrote: On 23:11, Thu 26 Oct 06, Armin Schindler wrote: snip/snip chan-capi-cm (chan-capi.org) is a complete rewritten version of chan-capi with more features and as far as I can tell, much more stable. You do faxing with chan-capi 0.3.5? But this isn't faxing via CAPI, right? As far as I know, chan-capi 0.3.5 does not support CAPI faxing. You are correct. chan-capi 0.3.5 does not support faxing out of the box. But there's a patch out there that fixes this. It will add CapiAnswerFax(). This function saves a .sff (structured fax file) stream to disk which you can run through sfftobmp and stuff. We create PDF files with some commandline tools and it's rock stable (2 years without a missed/corrupted fax on a system that takes like 10 to 15 faxes a week) Because it's working and we really believe in if it aint broken dont fix it we did not look into chan-capi-cm. This feature provides chan-capi.org for a long time and since version 0.7 you can send faxes too. offtopic My personal view on things: avoid PSTN/ISDN connections where possible and go with ITSP services. sangoma, quadbri, capi, tdm,... they all caused headaches where IAX simply works. Off course faxes wont work really good with ITSP but most of them have fax2email and email2fax. This is from a viewpoint of office PBX integration etc, not from ITSP viewpoint /offtopic I don't agree here. I have a few servers running with chan-capi.org and Eicon DIVA Server cards no problems at all. Armin ___ --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
[asterisk-users] Iax bug ?
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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Re: [asterisk-users] Asterisk and ISDN and Hylafax
Hi. I have now many customers using hylafax + asterisk, and all of them have proven to be reliable. Two are using diva server 4bri 8m + CAPI (asterisk here is not involved, as the incoming fax call gets directly to ttyds0x devices, and the numbers assigned to the lines by our telco are excluded from extensions.conf), which is absolutely *perfect*, not even a single miss (except of course for remote party's troubles). Another couple of them is using a Digium T110 pri card with asterisk+iaxmodem (0.1.14) on the same box, and they work just fine (the only limitation is running at 9600 instead of 14400 bps) The latter two boxes have 30 software instances of iaxmodem (ttyIAX1..ttyIAX30) and, although we've never tested what happens with cpu (P4 xeon 2.8 uniproc) when all of them are DSPing, the server average load is 500 incoming faxes/day, with peaks of 6-7 simultaneous incoming fax jobs. Bye, Alberto. Thomas Winter ha scritto: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes and/or use capi4hylafax in parallel with asterisk/chan-capi. sounds good, you think it will run reliable? ___ --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 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Asterisk and ISDN and Hylafax
On Fri, 27 Oct 2006, Thomas Winter wrote: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes and/or use capi4hylafax in parallel with asterisk/chan-capi. sounds good, you think it will run reliable? I do think so. I have this exactly this setup running twice. One setup even has 2 DIVA 4BRI cards, where the send one is running all ports in NT-mode to connect a legacy PBX. Regarding the 'parallel' hylafax, you just need to make sure that your setup is correct, e.g. asterisk should not be configured to accept calls which are meant for hylafax. Armin ___ --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
[asterisk-users] Auto Dial problem!
Hello list, I try to configure auto dial from asterisk (called server B) to another asterisk (server A) using SIP but I have a strange problem ! (Softphone connected to server B calling clients of server A works) On server B, I have : sip.conf : [to_serverA] type=peer username=from_serverB fromdomain=domainB fromuser=from_serverB host=server_A_IP secret= insecure=very nat=no test.call Channel: SIP/to_serverA/99123456 WaitTime: 30 RetryTime: 2 MaxRetries: 2 Context:autodial Extension:99123456 Priority:1 In plain words, I want to call server A using channel to_serverB. Server A conf: sip.conf [from_serverB] type=friend secret= host=dynamic insecure=very context=autodial nat=no extensions.conf [autodial] exten = s,1,Answer() exten = s,2,Playback(dir-intro) exten = s,3,Playback(vm-goodbye) exten = s,4,Hangup() ;and tried also to substitute s by _99XX The problem : Logs on server A show that extension autodial is called and everything seems to work! However, logs on server B says : autodial,99123456,1 failed so falling back to exten 's' sent into invalid extension 's' in context 'default', but no invalid handler In my opinion, the call really failed because it lasted 0 sec!!! ;-) What is my problem?? I admit that, for an auto dial, it is useless to Playback something! It can be replaced by a temporisation !! If you have a suggestion, help me ! Thank you! ___ --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
[asterisk-users] lines usage statistics
Hi list, I want to make a statistics about the number of parallel calls on my * running a beronet E1 card. The easy variant would be to get a number of maximal parallel calls to my machine during a day. The extended would be a graph showing the load over the day. If noone knows a direct solution to my question I would have an idea how to make up the easy variant with extensions. The only thing I would be missing for that would be a way to read in the current date and time. The application DateTime does not what I first thought it would. So, does anyone of you know how to get it? Backticks is not working on that machine. I am running asterisk 1.2.7.1with chan_misdn 0.3.1rc17. I know this is not absolutely up to date but I cannot afford a longer downtime. TIA regards, Christophorus ___ --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] Iax bug ?
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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Com os melhores cumprimentos,
Marco Mouta
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Re: [asterisk-users] Iax bug ?
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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--
Jean-Baptiste Bellet
Ingénieur Développpement
Lucyde SAS
Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex
+33 (0)5 34 31 86 36
http://www.lucyde.com
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[Asterisk-Users] Would you support a Bristuff mailing list ?
Hi,It seems to me that Bristuff usage has reached a point which implies a dedicated mailing list.This list would be of major use for :- bugs assessment- features requests- comments on Asterisk news Who seconds that ?Would it be difficult to make this happen ?Regards ___ --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] Is SQLite a replacement for Mysql while using ARA in 1.2.x
On Fri, Oct 27, 2006 at 07:56:16AM +0800, Leo Ann Boon wrote: Moises Silva wrote: AFAIK, you will need to do the first. ARA-odbc-sqlite res_sqlite3 in asterisk-addons supports ARA res_sqlite3 from aadd-ons is a strange beast. It uses its own, private copy of sqlite and acceses internal data structures. So while the database that it uses is hopefully sqlite3, it is not perfectly guaranteed. (This is why it's not part of the Debian packages built of asterisk-addons) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Asterisk and ISDN and Hylafax
2006/10/27, Armin Schindler [EMAIL PROTECTED]: On Fri, 27 Oct 2006, Thomas Winter wrote: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes and/or use capi4hylafax in parallel with asterisk/chan-capi. sounds good, you think it will run reliable?I do think so. I have this exactly this setup running twice. One setup evenhas 2 DIVA 4BRI cards, where the send one is running all ports in NT-mode to connect a legacy PBX.Regarding the 'parallel' hylafax, you just need to make sure that your setupis correct, e.g. asterisk should not be configured to accept calls which aremeant for hylafax.Armin What about telephony features using chan-capi and Asterisk ?Are those features on par with msidn+Asterisk or bristuff+Asterisk (maybe I'm mixing up things together) ?Cheers ___ --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] Re: ECHO Cancellation in SIP Calls
On Thu, Oct 26, 2006 at 04:17:16PM -0700, Alyed Tzompa wrote: Echo is generated by the analog end to where you place the call, not the IP side of it. As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it) I'm afraid there is little you can do to here. A digital zaptel card (PRI/BRI) does not generate them either. If the echo would be generated on your side, you wouldn't be the one to hear it. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Iax bug ?
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160, this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and debug),
the call is executed in the context of the last section's context of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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--
Jean-Baptiste Bellet
Ingénieur Développpement
Lucyde SAS
Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex
+33 (0)5 34 31 86 36
http://www.lucyde.com
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Com os melhores cumprimentos,
Marco Mouta
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Re: [asterisk-users] Iax bug ?
Thanks to take time to write me back (oola I' don't no if this is a
correct sentence !)
I think the variable sevret is empty is not a problem : without it it's
the same !
I will try to debug with type=peer and type=user
I didn't know this site, hope it will be helpfull !
jb
Marco Mouta a écrit :
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa
authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
The iax.renoir.conf file contains :
[VOIP_RENOIR]
type=friend
host=renoir.lucyde
auth=rsa
inkey=key_184
outkey=key_Renoir
context=CONTEXT_RENOIR
trunk=yes
allow=gsm
Thanks to the variable context, when .184 receive a call from .160,
this
call should be executed in the CONTEXT_VOIP1. In fact the call is
executed in the CONTEXT_RENOIR. Exactly (with a lot of test and
debug),
the call is executed in the context of the last section's context
of the
iax.conf file (e.g. CONTEXT_RENOIR here).
Anyone who has any idea ?
Thanks,
jb
PS :
(The debug in the .184 machine :
-- Accepting UNAUTHENTICATED call from 10.0.0.160:
requested format = ulaw,
requested prefs = (alaw),
actual format = gsm,
host prefs = (gsm),
priority = mine
-- Executing NoOp(IAX2/10.0.0.160:4569-1, I'm in
CONTEXT_RENOIR) in new stack
-- Executing Macro(IAX2/10.0.0.160:4569-1, check_forward|106)
in new stack
with the following extensions.conf :
[CONTEXT_VOIP1]
exten = _X.,1,NoOp(I'm in CONTEXT_VOIP1)
exten = _X.,2,Macro(check_forward,${EXTEN})
[CONTEXT_RENOIR]
exten = _X.,1,NoOp(I'm in CONTEXT_RENOIR)
exten = _X.,2,Macro(check_forward,${EXTEN})
)
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--
Jean-Baptiste Bellet
Ingénieur Développpement
Lucyde SAS
Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex
+33 (0)5 34 31 86 36
http://www.lucyde.com
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asterisk-users mailing list
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--
Jean-Baptiste Bellet
Ingénieur Développpement
Lucyde SAS
Prologue 1 - La Pyrénéenne BP 27201 LABEGE cedex
+33 (0)5 34 31 86 36
http://www.lucyde.com
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Re: [asterisk-users] Iax bug ?
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
Incoming Connections
When Asterisk receives an incoming IAX connection, the initial call
information can include a username (in the IAX2 USERNAME field) or
not. In addition, the incoming connection has a source IP address that
Asterisk can use for authentication as well.
If a username is supplied, Asterisk does the following:
* Search iax.conf for a type=user entry with a section name (eg
[username]) matching the supplied username; if no matching entry is
found, refuse the connection.
* If the found entry has allow and/or deny settings, compare the
IP address of the caller to these lists. If the connection is not
allowed, refuse the connection.
* Perform the desired secret checking (plaintext, md5 or rsa); if
it fails, refuse the connection.
* Accept the connection and send the caller to the context
specified in the context setting for this iax.conf entry.
If a username is not supplied, Asterisk does the following:
* Search for a type=user entry in iax.conf with no secret
specified and also allow and/or deny restrictions that do not restrict
the caller from connecting. If such an entry is found, accept the
connection, and use the name of the found iax.conf entry as the
connecting username.
* Search for a type=user entry in iax.conf with no secret
specified and no allow and/or deny restrictions at all. If such an
entry is found, accept the connection. and use the name of the found
iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and also allow and/or deny restrictions that do not
restrict the caller from connecting. If such an entry is found,
attempt to authenticate the caller using the specified secret or key,
and if that passes, accept the connection, and use the name of the
found iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and no allow and/or deny restrictions at all. If such
an entry is found, attempt to authenticate the caller using the
specified secret or key, and if that passes, accept the connection,
and use the name of the found iax.conf entry as the connecting
username.
Hope this helps!
I didn't read all, but what i guess is: the incoming call isn't being
correctly authenticated, so can't go to VOIP1 as you desire, then as
is mention above:
Search for a type=user entry in iax.conf with no secret specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
[asterisk-users] RE: ECHO Cancellation in SIP Calls
Message: 7 Date: Thu, 26 Oct 2006 22:56:58 -0400 From: Michael Araba [EMAIL PROTECTED] Subject: [asterisk-users] RE: ECHO Cancellation in SIP Calls To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I am surprised that you are getting echo on SIP calls. You can get echo in two scenarios on SIP calls. 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need to enable echo canceller and AGGRESSIVE if needed in zconfig.h. 2. Second source of echo on SIP calls could be ACOUSTIC. The phone sets you are using may not handle this well. In my experience sound quality deteriorates if there is network trouble or congestion on SIP calls I hope this helps. Michael Hi Michael, For sure, i can get echo in the 2 to 4 wire scenario, this is right, but this cant be happen in MY way, only the provider can produce this scenario, my asterisk use zap and isdn, but the echo occure in pure sip calls, in my zap and isdn channels i use the patch from mgernoth, named mg2, great stuff. The second is one echo i already know, one other caller parties use very cheap phones, so the sound of the telephone speaker is not shielded enough to put no sound in the telephone mic - this is not the case with my phones, i use SNOM, they are build to used with VoIP and the best one i know, in my case. I checked the latency and loss between me and my provider this morning again, and i figured out a routing point which lost 3% of my packets, first time for me to see this after one year of working good, i wrote a mail to my provider, and asked him to check this on his own, but i cant imagine that this produce all the echo...must wait, i guess. I tested my Network, good results, tested other VoIP Provider's Server, Result is good to ok. Recap : To minimize echo i can check : Phone (ok), Channels in Asterisk (crossing) (ok), My Network Connections between Phone and Asterisk (ok), Network between Asterisk and Router (ok), Connection, Loss and Latency between Asterisk/Router and my VoiPProvider (waiting..) Any other ways to produce echo in pure *SIP* ! Thanks for your great help ! Stefan ___ --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] porting numbers in UK telewest/bt/adept
Any experts on porting numbers in the uk here? ;-) Yep, it is your legal _right_ to have the numbers ported in a reasonable time/cost. Point this out to them and ask what the complaints escalation procedure is. That should get their attention. Can you point me to the law that gives you the legal right to port numbers between providers? As far as I was aware they had to have a porting agreement with the new carrier to able to do this. ___ --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
[asterisk-users] asterisk misdn incoming line not working.
Hi list, I have a server running a simple hfs isdn card running with chan_misdn. the problem is, I can't get asterisk to pick up the phone, outgoing calls work fine. when running asterisk with -vvvc I get the following log when I try to dial the isdn server. P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's' extension P[ 1] MGMT: SSTATUS: L1_ACTIVATED == Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's' == Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to context 'default' Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2' sent into invalid extension 's' in context 'default', but no invalid handler 087822291 is the number i dial from, 0594643637 is the number that the asterisk server should respond to. in misdn.conf i created a kpn section like this: [kpn] ports=1ptmp ;group=1 immediate=yes always_immediate=yes context=kpn-in hold_allowed=yes msns=* and in extensions.conf i created a very basic kpn-in section like this: [kpn-in] exten = kpn-in,1,Dial(SIP/mark,25,tr) I don't really have much experience with asterisk so I probably did something wrong here, but I couldn't really figure out how to get it done. anyone out there any ideas? Mark ___ --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
[asterisk-users] ISDN-BRI issue
Hello all Asterisk 1.2.10 BRIstuff PRE-1s Debian sarge I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is down, no matter is I have an actual line plugged in or not. == Primary D-Channel on span 1 down I somehow got it to work once! The config tools do not indicate any errors whatsoever. Here is a rundown of various listings I can think about. (some things may show some FXO stuff, but I took the card out to make sure that it wasn't interfering). I am at a loss. It seems that I get the same result whether the ISDN signal is on or off. However, the LED on my card is red when no signal, and well green when there's a signal. Please help! Any hints, advices... anything is welcome. Thank you. cat /proc/zaptel/* (when a line is in, one of them shows ACTIVATED) == Span 1: ztqoz/2/1 quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0) Layer 1 DEACTIVATED (F3) 1 ztqoz2/1/1 Clear (In use) 2 ztqoz2/1/2 Clear (In use) 3 ztqoz2/1/3 HDLCFCS (In use) Span 2: ztqoz/2/2 quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 0) Layer 1 DEACTIVATED (F3) 4 ztqoz2/2/1 Clear (In use) 5 ztqoz2/2/2 Clear (In use) 6 ztqoz2/2/3 HDLCFCS (In use) Span 3: ztqoz/2/3 quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 0) Layer 1 DEACTIVATED (F3) 7 ztqoz2/3/1 Clear (In use) 8 ztqoz2/3/2 Clear (In use) 9 ztqoz2/3/3 HDLCFCS (In use) Span 4: ztqoz/2/4 quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0) Layer 1 DEACTIVATED (F3) 10 ztqoz2/4/1 Clear (In use) 11 ztqoz2/4/2 Clear (In use) 12 ztqoz2/4/3 HDLCFCS (In use) ztcfg -vv = Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: D-channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: D-channel (Default) (Slaves: 12) 12 channels configured. pro show span 1: asterisk*CLI pri show span 1 Primary D-channel: 3 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE (PtMP) Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 asterisk*CLI asterisk*CLI pri show span 1 Primary D-channel: 3 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE (PtMP) Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 zapata.conf === ;;; ISDN channel ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) usecallerid = yes signalling = bri_cpe_ptmp switchtype = euroisdn echocancel = yes echocancelwhenbridged = yes echotraining = 100 ;callerid = Vision IT Group 26 44 36 1 ;resetinterval=never context=isdn-incoming group = 1 ; S/T port 1-4 ;channel = 25-26 ;channel = 28-29 ;channel = 31-32 ;channel = 34-35 channel = 1-2 channel = 4-5 channel = 7-8 channel = 10-11 zaptel.conf === ### ISDN span config ### span=1,1,3,ccs,hdb3 span=2,2,3,ccs,hdb3 span=3,3,3,ccs,hdb3 span=4,0,3,ccs,hdb3 #BRI's: #bchan=25,26 #dchan=27 #bchan=28,29 #dchan=30 #bchan=31,32 #dchan=33 #bchan=34,35 #dchan=36 bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 ### FXO span config ### #fxoks=13-24 loadzone=fr defaultzone=fr signature.asc Description: This is a digitally signed message part ___ --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
[asterisk-users] Direct call vs Block call
Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel we're enable block dial, so alcatel only send to asterisk when user end dialing all digits. How i permit the first case to work ?? Thanks. -- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br ___ --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
[asterisk-users] DTMF detection problem in PABX trunk
Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br ___ --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] VoiceOne 0.4.0 released: a new web-based and open source GUI
Alex ha scritto: Hi all! We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. Thanks guys, translators and testers are welcome! We have a dedicated forum at http://www.voiceone.it/forum/viewforum.php?f=5 where you'll be able to obtain all details by our translations responsible. However, we're sending you a personal e-mail with instructions attached. Regards, Alex ___ --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] asterisk misdn incoming line not working.
Hi Mark, why exten = *kpn-in*,1,Dial(SIP/mark,25,tr) ?? Try: exten = s,1,Dial(SIP/mark,25,tr) and exten = _X.,1,Dial(SIP/mark,25,tr) Giorgio Incantalupo Mark Hannessen wrote: Hi list, I have a server running a simple hfs isdn card running with chan_misdn. the problem is, I can't get asterisk to pick up the phone, outgoing calls work fine. when running asterisk with -vvvc I get the following log when I try to dial the isdn server. P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's' extension P[ 1] MGMT: SSTATUS: L1_ACTIVATED == Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's' == Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to context 'default' Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2' sent into invalid extension 's' in context 'default', but no invalid handler 087822291 is the number i dial from, 0594643637 is the number that the asterisk server should respond to. in misdn.conf i created a kpn section like this: [kpn] ports=1ptmp ;group=1 immediate=yes always_immediate=yes context=kpn-in hold_allowed=yes msns=* and in extensions.conf i created a very basic kpn-in section like this: [kpn-in] exten = kpn-in,1,Dial(SIP/mark,25,tr) I don't really have much experience with asterisk so I probably did something wrong here, but I couldn't really figure out how to get it done. anyone out there any ideas? Mark ___ --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
[asterisk-users] Taking a Polycom IP601 home
I am taking a Polycom IP601 home to try to figure out how to provision it outside of the office for our outsides sales people. Our asterisk server has a direct outside IP. The IP601 will be behind a router at home so it will not have an outside IP. I am fully opening the company firewall for my home IP so that all services (ir the FTP server) will be available to the phone. What else will I need to have in place to get the phone to work? Has anyone done this yet? Can it be done? TIA, Warren ___ --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] SIP problem - ACT p160s error
Thanks. I will give that a try. Do you know if removing that line will affect other phones I might have? If so, maybe I am better off getting someone else's phone. ACT's support seems a bit problematic. They responded to my first email right away, but never, so far, to my second. Then I realized their first one was a brush off, simply saying essentially only you are having this problem. joe a. [EMAIL PROTECTED] Wrote on: 10/26/2006 11:40 PM: I saw this problem before... to solve that, I needed to hack asterisk to remove a header SIP field. Check your ACT phone log, and you can figure out which filed is that. Then, comment that filed from your chan_sip.c and recompile asterisk.. and that's it.. it only happens with ACT phones. I wonder if anybody knows from where we can download ACT firmware updates. Isamar On Wed, 25 Oct 2006, joe, at j4computers ([EMAIL PROTECTED]) wrote: I have a setup with a polycom 601 and an act p160s. All on local segment, no NAT. Can call the act p160s, from the polycom, rings, connects, and a conversation can take place. The reverse is not true, Dialing from the act to the polycom does not work. SIP debug shows, at the end, Incoming call: got sip response 416 unsupported URI Scheme back from 192.168.0.xxx. Which is the act phone, the orginator. One presumes this is a configuration issue with the Act phone. Any clues? Such as what a proper config for this phone should look like? Act support has made an initial response, but there is a big time lag them being on the other side of the earth. joe a. ___ --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] Direct call vs Block call
pls post your misdn.conf as well as extensions.conf, so someone could help you on this. On 10/27/06, Frederico Madeira [EMAIL PROTECTED] wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel we're enable block dial, so alcatel only send to asterisk when user end dialing all digits. How i permit the first case to work ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] asterisk misdn incoming line not working.
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.
when running asterisk with -vvvc I get the following log when I try
to dial the isdn server.
P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's'
extension
P[ 1] MGMT: SSTATUS: L1_ACTIVATED
== Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's'
== Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to
context 'default'
Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2'
sent into invalid extension 's' in context 'default', but no invalid handler
087822291 is the number i dial from, 0594643637 is the number that the
asterisk server should respond to.
in misdn.conf i created a kpn section like this:
[kpn]
ports=1ptmp
;group=1
immediate=yes
always_immediate=yes
context=kpn-in
hold_allowed=yes
msns=*
and in extensions.conf i created a very basic kpn-in section like this:
[kpn-in]
HERE IS your PROBLEM:
exten = kpn-in,1,Dial(SIP/mark,25,tr)
1- Be sure of of MSNs string your telco is sending you.
2- Do this:
[kpn-is]
exten= _X.,1,answer
exten= _X.,1,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
reload your asterisk after this changes, and dial again.
Now you may understand what your telco is sending you and then start
routing it on your way.
Hope this helps,
Pls. give me some feedback.
I don't really have much experience with asterisk so I probably did something
wrong here, but I couldn't really figure out how to get it done.
anyone out there any ideas?
Mark
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--
Com os melhores cumprimentos,
Marco Mouta
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Re: [asterisk-users] asterisk misdn incoming line not working.
My mistake:
[kpn-is]
exten= _X.,1,answer
exten= _X.,2,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Plse Read bellow:
On 10/27/06, Mark Hannessen [EMAIL PROTECTED] wrote:
Hi list,
I have a server running a simple hfs isdn card running with chan_misdn.
the problem is, I can't get asterisk to pick up the phone, outgoing calls work
fine.
when running asterisk with -vvvc I get the following log when I try
to dial the isdn server.
P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's'
extension
P[ 1] MGMT: SSTATUS: L1_ACTIVATED
== Starting mISDN/1-2 at kpn-in,s,1 failed so falling back to exten 's'
== Starting mISDN/1-2 at kpn-in,s,1 still failed so falling back to
context 'default'
Oct 27 01:28:42 WARNING[3514]: pbx.c:2357 __ast_pbx_run: Channel 'mISDN/1-2'
sent into invalid extension 's' in context 'default', but no invalid handler
087822291 is the number i dial from, 0594643637 is the number that the
asterisk server should respond to.
in misdn.conf i created a kpn section like this:
[kpn]
ports=1ptmp
;group=1
immediate=yes
always_immediate=yes
context=kpn-in
hold_allowed=yes
msns=*
and in extensions.conf i created a very basic kpn-in section like this:
[kpn-in]
HERE IS your PROBLEM:
exten = kpn-in,1,Dial(SIP/mark,25,tr)
1- Be sure of of MSNs string your telco is sending you.
2- Do this:
[kpn-is]
exten= _X.,1,answer
exten= _X.,1,Noop(My telco is sending me this MSN string: ${EXTEN})
exten= _X.,3,wait(1)
exten= _X.,4,Playback(vm-goodbye)
exten= _X.,5,hangup
reload your asterisk after this changes, and dial again.
Now you may understand what your telco is sending you and then start
routing it on your way.
Hope this helps,
Pls. give me some feedback.
I don't really have much experience with asterisk so I probably did something
wrong here, but I couldn't really figure out how to get it done.
anyone out there any ideas?
Mark
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--
Com os melhores cumprimentos,
Marco Mouta
--
Com os melhores cumprimentos,
Marco Mouta
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[asterisk-users] Re: Re: ECHO Cancellation in SIP Calls
If you are calling from a SIP phone through asterisk and through a Digium card, one could argue that the Digium card IS farside of the SIP phone. SIP -- asterisk -- Digium -- T1/PRI/analog -- PSTN -- PSTN -- PSTN -- Destination. I would argue that the Digium card IS on the farside of asterisk as far as the SIP phone is concerned. We just switched from Legacy PBX to Asterisk and we get occasional echo. Everything past the Digium card is the same as the old PBX. We never got echo on the old PBX. -- -- Steven http://www.glimasoutheast.org Tzafrir Cohen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On Thu, Oct 26, 2006 at 04:17:16PM -0700, Alyed Tzompa wrote: Echo is generated by the analog end to where you place the call, not the IP side of it. As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it) I'm afraid there is little you can do to here. A digital zaptel card (PRI/BRI) does not generate them either. If the echo would be generated on your side, you wouldn't be the one to hear it. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] DTMF detection problem in PABX trunk
Check your dtmfmode I use dtmfmode=rfc2833 Check with your provider Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --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 Inbound (clean). Database: 0643-6, 10/26/2006 - 10/27/2006 7:52:49 AM ___ --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] ISDN-BRI issue
Frédéric Blaise ha scritto: Hello all Asterisk 1.2.10 BRIstuff PRE-1s Debian sarge I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is down, no matter is I have an actual line plugged in or not. == Primary D-Channel on span 1 down Try with signalling=bri_cpe even if your lines are set as point to multipoint, at least that should make your card trying to keep layer 1 up, even if this won't probably solve it. As a matter of fact I'm getting to conclude that bristuff + hfc-4s card is not working whatsoever. I believe there's something wrong in the way bristuff manages layer 1 (not sure if that's a driver problem, hardware problem or both). Alberto. ___ --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] DTMF detection problem in PABX trunk
Hi Frederico, I had digits detection problems with my ISDN beronet cards too, do not know if u are using those cards but in case try to add s parameter to Dial command: dial(mISDN/1/123/s) It worked for me. :) Giorgio Incantalupo Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.madeira.eng.br http://www.madeira.eng.br ___ --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
[asterisk-users] Re: asterisk not detecting hangup
I've enabled those options but it's the same. On 10/25/06, Maxi Belino [EMAIL PROTECTED] wrote: i'm having similar problems (if you find out the solution please post it) did you try enabling 'callprogress' or 'busydetect' in zapata.conf ? Maxi 2006/10/23, Arkaitz [EMAIL PROTECTED]: Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Identified as: *CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard X101P Board 1 OK 0 0 0 And the following lines in zapata.conf(for spanish lines): answeronpolarityswitch=yes hanguponpolarityswitch=yes The problem is that although the calls work correctly the system is unable to detect a pstn hangup and it keeps running even when the other side is calling to another number(not an asterisk ones, asterisk line keeps busy) Any hint? Thanks for your time -- Arkaitz There is a good bit of info on the web about this issue. http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision http://kb.digium.com/entry/1/6/ http://www.google.com/search?hl=enlr=q=asterisk+disconnect+supervision+ Bottom line, you have to call your service provider and get them to turn on disconnect supervision so Asterisk will properly detect the far-end hang-up. Depending on the service provider, what switch they use, and what part of the country you are in, this feature is called something different. Good Luck. JR ___ --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] Asterisk and ISDN and Hylafax
Olivier ha scritto: What about telephony features using chan-capi and Asterisk ? Are those features on par with msidn+Asterisk or bristuff+Asterisk (maybe I'm mixing up things together) ? Cheers I'm running my own company's pbx with diva 4bri, diva server for linux 8.2, chan_capi from melware.org and everything is working just fine. I was able to use early B3 connect and everything else related to CLID. I have two isdn lines as point-to-multipoint and one as p2p. One isdn line is shared between asterisk/chan_capi and hylafax via diva's tty interfaces. Everything has been perfectly working since november 2005, when we first started this new pbx as a replacement of the old Samsung DCS, we almost forgot about its existence, as we did never have to put hands on it to fix problems (except for some ordinary maintenance and diva server software upgrades). Alberto. ___ --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
[asterisk-users] IAX2 show peers - description
Hi people, pls does anybody know what (T) and (D) letter means? server3*CLI iax2 show peers Name/UsernameHost Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0 unmonitored] thanks, Marian -- Marian Rychtecky [EMAIL PROTECTED] Tel. +420 724 397 441 ICQ 76582857 ___ --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
[asterisk-users] How to hung up , While in Conference going on.
Hello Users,
Good Morning,
In Conferemcing How to Disconnect the phone while in between the
Conference .
When I press the ' # ' key for Disconnecting the
Conference..
Below the Following to shows some Warning, ( in Red Color )
from-sip en
*CLI -- Executing Playback("SIP/9002-08f9feb8",
"conf-hasentered") in new stack
-- Playing 'conf-hasentered' (language 'en')
-- Executing Wait("SIP/9002-08f9feb8", "2") in new stack
-- Executing MeetMe("SIP/9002-08f9feb8", "12345|p") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '12345'
-- Playing 'conf-getpin' (language 'en')
-- Playing 'conf-onlyperson' (language 'en')
-- Executing Playback("SIP/9001-08fb34d0", "conf-hasentered") in
new stack
-- Playing 'conf-hasentered' (language 'en')
-- Executing Wait("SIP/9001-08fb34d0", "2") in new stack
-- Executing MeetMe("SIP/9001-08fb34d0", "12345|p") in new stack
-- Playing 'conf-getpin' (language 'en')
Oct 26 18:52:47 WARNING[23516]: pbx.c:2415
__ast_pbx_run: Timeout, but no rule 't' in context 'from-sip'
-- Hungup 'Zap/pseudo-656465881'
Oct 26 18:53:35 WARNING[23485]: pbx.c:2415 __ast_pbx_run: Timeout, but
no rule 't' in context 'from-sip'
--
Thanks Regards,
Ravi
Prakash Sunkara
M:+91
9985077535
O:+91 40 23114549
F:+91 40 40208727
[EMAIL PROTECTED]
www.hyperion-tech.com
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Re: [asterisk-users] Iax bug ?
Thanks a lot.
I think UNAUTHENTICATED call is the source of my problems.
How I can solve it ?
Because allowguest is a sip.conf option ...
jb
Marco Mouta a écrit :
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
Incoming Connections
When Asterisk receives an incoming IAX connection, the initial call
information can include a username (in the IAX2 USERNAME field) or
not. In addition, the incoming connection has a source IP address that
Asterisk can use for authentication as well.
If a username is supplied, Asterisk does the following:
* Search iax.conf for a type=user entry with a section name (eg
[username]) matching the supplied username; if no matching entry is
found, refuse the connection.
* If the found entry has allow and/or deny settings, compare the
IP address of the caller to these lists. If the connection is not
allowed, refuse the connection.
* Perform the desired secret checking (plaintext, md5 or rsa); if
it fails, refuse the connection.
* Accept the connection and send the caller to the context
specified in the context setting for this iax.conf entry.
If a username is not supplied, Asterisk does the following:
* Search for a type=user entry in iax.conf with no secret
specified and also allow and/or deny restrictions that do not restrict
the caller from connecting. If such an entry is found, accept the
connection, and use the name of the found iax.conf entry as the
connecting username.
* Search for a type=user entry in iax.conf with no secret
specified and no allow and/or deny restrictions at all. If such an
entry is found, accept the connection. and use the name of the found
iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and also allow and/or deny restrictions that do not
restrict the caller from connecting. If such an entry is found,
attempt to authenticate the caller using the specified secret or key,
and if that passes, accept the connection, and use the name of the
found iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and no allow and/or deny restrictions at all. If such
an entry is found, attempt to authenticate the caller using the
specified secret or key, and if that passes, accept the connection,
and use the name of the found iax.conf entry as the connecting
username.
Hope this helps!
I didn't read all, but what i guess is: the incoming call isn't being
correctly authenticated, so can't go to VOIP1 as you desire, then as
is mention above:
Search for a type=user entry in iax.conf with no secret specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa
authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include iax.voip1.conf
#include iax.renoir.conf
The iax.voip1.conf file contains :
[VOIP1]
Re: [asterisk-users] ISDN-BRI issue
On Fri, 2006-10-27 at 14:14 +0200, Alberto Pastore wrote: Try with signalling=bri_cpe even if your lines are Yes, I tried with all kind of signalling, including this one, but this doesn't work either. bri_cpe_ptmp seems to be the one... set as point to multipoint, at least that should make your card trying to keep layer 1 up, even if this won't probably solve it. As a matter of fact I'm getting to conclude that bristuff + hfc-4s card is not working whatsoever. hmmm. Well, I am trying to see if I can get some info out of Junghanns support. If I make it work, I'll let you guys know. I believe there's something wrong in the way bristuff manages layer 1 (not sure if that's a driver problem, hardware problem or both). Alberto. Thanks fred ___ --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 signature.asc Description: This is a digitally signed message part ___ --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
[asterisk-users] Advice on GUI
Hello all I would like to know your opinions on free GUI used to manage Asterisk. Which is better? My setup is quite small, about 15-20 phones. I've seen the liste on voip-info. Thanks all. fred signature.asc Description: This is a digitally signed message part ___ --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
[asterisk-users] Snom, mute and rtptimeout
I have a bunch of Snom phones. When I press the mute button, the phone stops sending RTP frames. If I have rtptimeout set, that means that the connection will eventually be cut off. It also affects sound generated by asterisk, since timing is generated from the incoming frames. Are there any workarounds I can try? Disabling rtptimeout is one, obviously. /Benny ___ --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] Iax bug ?
Why r u using rsa authentication? you should start with something
simple. test the link i sent u.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Thanks a lot.
I think UNAUTHENTICATED call is the source of my problems.
How I can solve it ?
Because allowguest is a sip.conf option ...
jb
Marco Mouta a écrit :
Hi,
I think i found your problem, look that in your debug you have, -
Accepting UNAUTHENTICATED call from 10.0.0.160:
Take a look on incoming call authentication, and how asterisk handles this:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication
Incoming Connections
When Asterisk receives an incoming IAX connection, the initial call
information can include a username (in the IAX2 USERNAME field) or
not. In addition, the incoming connection has a source IP address that
Asterisk can use for authentication as well.
If a username is supplied, Asterisk does the following:
* Search iax.conf for a type=user entry with a section name (eg
[username]) matching the supplied username; if no matching entry is
found, refuse the connection.
* If the found entry has allow and/or deny settings, compare the
IP address of the caller to these lists. If the connection is not
allowed, refuse the connection.
* Perform the desired secret checking (plaintext, md5 or rsa); if
it fails, refuse the connection.
* Accept the connection and send the caller to the context
specified in the context setting for this iax.conf entry.
If a username is not supplied, Asterisk does the following:
* Search for a type=user entry in iax.conf with no secret
specified and also allow and/or deny restrictions that do not restrict
the caller from connecting. If such an entry is found, accept the
connection, and use the name of the found iax.conf entry as the
connecting username.
* Search for a type=user entry in iax.conf with no secret
specified and no allow and/or deny restrictions at all. If such an
entry is found, accept the connection. and use the name of the found
iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and also allow and/or deny restrictions that do not
restrict the caller from connecting. If such an entry is found,
attempt to authenticate the caller using the specified secret or key,
and if that passes, accept the connection, and use the name of the
found iax.conf entry as the connecting username.
* Search for a type=user entry in iax.conf with a secret (or RSA
key) specified and no allow and/or deny restrictions at all. If such
an entry is found, attempt to authenticate the caller using the
specified secret or key, and if that passes, accept the connection,
and use the name of the found iax.conf entry as the connecting
username.
Hope this helps!
I didn't read all, but what i guess is: the incoming call isn't being
correctly authenticated, so can't go to VOIP1 as you desire, then as
is mention above:
Search for a type=user entry in iax.conf with no secret specified
and no allow and/or deny restrictions at all. If such an entry is
found, accept the connection. and use the name of the found iax.conf
entry as the connecting username.
Pls give some feedback if you solved the problem.
On 10/27/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
Unfortunately i'm not able to debug this with you now :( I'm busy.
[VOIP1]
type=friend
host=10.0.0.160
auth=rsa
secret=
This secret empty is this allowed?
inkey=voip3
outkey=voip1
context=CONTEXT_VOIP1
allow=all
ipaddr=10.0.0.160
port=4569
qualify=yes
trunk=yes
Try a simple test with this, and then step by step go to rsa
authentication.
http://astrecipes.net/index.php?n=204
If in troubles, post here i'll try to help you
By the way, to understand much better what's going on i would
recommend you to not use type=friend and use type=user and type=peer.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Here the .160's iax.conf file :
[general]
bandwidth=high
tos=reliability
bandwidth=low
disallow=all; Icky sound quality... Mr. Roboto.
allow=alaw ; Always allow GSM, it's cool :)
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
[VOIP1]
type=friend
host=10.0.0.184
auth=rsa
inkey=voip3
outkey=voip1
context=VOIPLINK3
qualify=1
trunk=yes
allow=all
How .160 call .184 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
How .184 call .160 :
exten = _1XXX,1,Dial(IAX2/VOIP1/${EXTEN:1:4})
(the same)
Thanks,
jb
Marco Mouta a écrit :
pls post iax.conf of Both machines , as well as your dial() string on
both servers to connect each other.
That way would be easier to help you.
On 10/27/06, Jean-Baptiste Bellet [EMAIL PROTECTED] wrote:
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of
RE: [asterisk-users] SipAddHeader
In the source that I've read (admittedly it's pretty old - 1.2.7.1) SipAddHeader() only appears to work on INVITEs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 26 October 2006 23:23 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SipAddHeader Does SipAddHeader only allow headers to be added to INVITEs, or should it also allow headers to be added BYEs or SIP responses as well? ___ --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
[asterisk-users] Enhancements for the Queue application
issue #8126 (http://bugs.digium.com/view.php?id=8216) on mantis is a patch for the queue system which allows you to specify a macro to run when a member is connected to a queue call, either by a configuration parameter in queues.conf or as an optional parameter on the Queue application. It also allows you to specify that certain variables are set when the member is connected. if setinterfacevar is yes MEMBERCALLS is the number of calls that interface has taken, MEMBERLASTCALL is the last time the member took a call. MEMBERPENALTY is the penalty of the member MEMBERDYNAMIC indicates if a member is dynamic or not if setqueueentryvar is yes QEHOLDTIME callers hold time QEORIGINALPOS original position of the caller in the queue if setqueuevar is yes QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue; QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time QUEUECOMPLETED number of completed calls for the queue QUEUEABANDONED number of abandoned calls QUEUESRVLEVEL queue service level QUEUESRVLEVELPERF current service level performance It also adds a new function called QUEUE_VARIABLES(queuename) which sets the Queue variables described above. This means now that you can access queue stats for a certain queue from within the dialplan. If anybody can, I would appreciate it if you could test it to see if it works for you. Suggestions and comments are most welcome. Julian ___ --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
[asterisk-users] set outgoing msn on chan_misdn
hi, does anyone know if it is possible to set the outgoing msn number with chan_misdn (the number the people on the other side will see as the caller) I already tried Set(CALLERID(num)=1234) SetVar(CALLERIDNUM=1234) Set(CALLERID(name)=1234[|a]) Set(CALLERID(number)=1234) but none of them seem to work :( Mark ___ --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] Taking a Polycom IP601 home
Make sure you set nat=yes for the sip user. Asterisk will then send replies back to the source IP address, rather than what's in the Via: header. -Original Message- From: Warren (mailing lists) [mailto:[EMAIL PROTECTED] Sent: Friday, October 27, 2006 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Taking a Polycom IP601 home I am taking a Polycom IP601 home to try to figure out how to provision it outside of the office for our outsides sales people. Our asterisk server has a direct outside IP. The IP601 will be behind a router at home so it will not have an outside IP. I am fully opening the company firewall for my home IP so that all services (ir the FTP server) will be available to the phone. What else will I need to have in place to get the phone to work? Has anyone done this yet? Can it be done? TIA, Warren ___ --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] Asterisk and ISDN and Hylafax
On Fri, 27 Oct 2006, Olivier wrote: 2006/10/27, Armin Schindler [EMAIL PROTECTED]: On Fri, 27 Oct 2006, Thomas Winter wrote: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes and/or use capi4hylafax in parallel with asterisk/chan-capi. sounds good, you think it will run reliable? I do think so. I have this exactly this setup running twice. One setup even has 2 DIVA 4BRI cards, where the send one is running all ports in NT-mode to connect a legacy PBX. Regarding the 'parallel' hylafax, you just need to make sure that your setup is correct, e.g. asterisk should not be configured to accept calls which are meant for hylafax. Armin What about telephony features using chan-capi and Asterisk ? Are those features on par with msidn+Asterisk or bristuff+Asterisk (maybe I'm mixing up things together) ? You can compare it with e.g. misdn+Asterisk. chan-capi is just another channel driver for Asterisk which provides access to ISDN/POTS hardware and drivers which support the CAPI interface. This includes - standard voice - DTMF - echo-cancel - Line-Interconnect - Fax - RTP ... Don't mix it with bristuff. As far as I know, bristuff consists of a) some additional zap driver. b) Asterisk changes/patches which alter asterisk features. [c) the old version of chan-capi] but actually it is not a channel-driver like chan-capi itself or chan-misdn. Armin ___ --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] Is SQLite a replacement for Mysql while using ARA in 1.2.x
Cohen, so you vote for the ARA-odbc-sqlite route? this is for embedded, so that's why sqlite instead of mysql or postgres. when you say it is not guaranteed, what do you mean? On 10/27/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Oct 27, 2006 at 07:56:16AM +0800, Leo Ann Boon wrote: Moises Silva wrote: AFAIK, you will need to do the first. ARA-odbc-sqlite res_sqlite3 in asterisk-addons supports ARA res_sqlite3 from aadd-ons is a strange beast. It uses its own, private copy of sqlite and acceses internal data structures. So while the database that it uses is hopefully sqlite3, it is not perfectly guaranteed. (This is why it's not part of the Debian packages built of asterisk-addons) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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
[asterisk-users] polycom's don't register with 2.6.18
Hello, Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but I'd really like to understand what's going on there... Any idea? ___ --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] Re: SIP v IAX2
Hi Which is most resistant to the loss of packages in a dirty link ? SIP or IAX ? roberto 2006/10/27, Dave Cotton [EMAIL PROTECTED]: On Thu, 2006-10-26 at 13:08 -0700, Martin Joseph wrote: On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said: Since they are incorporated in a single product which is doing the configuration, consistency where possible would be good... That product is designed to link the two things together, are you suggesting lowest common denominator configuration? Surely the best is to exploit each to it's maximum to achieve that goal even if it does make for slight differences in configuration, it is the day to day phone user experience that really matters. In computing inconsistances exist everywhere it is the job of a sysadmin to sort all this out so that it is transparent to the users. Sorry for the rant, I'm just going to a client to sort out the problems created, yet again, by his incompetent so called sysadmin. -- Dave Cotton [EMAIL PROTECTED] ___ --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 -- Ing. Roberto Pereyra ContenidosOnline Looking for Linux Virtual Private Servers ? Click here: http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426a_bid=56 ___ --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
[asterisk-users] [OT] wi-fi ip phone scenario
Hello everyone. I know it's a little bit off-topic, but I was just wondering... Has anyone ever had any experience with asterisk, a wi-fi meshed lan (with more than one access point) and wi-fi sip phones? I made some tests but I'm not really satisfied Wi-fi phones are a curse (as far as I know even Nokia eSeries -I personally own an e70 model- have their flaws): - random sip registration failures - ridiculous battery life - bad audio quality even with optimal radio environments - crashes, system freezes - ... - slow responsiveness to asterisk qualify pings (OPTIONS) but I can even live with that. The major problem is... roaming between cells. Is that a dream or something that can actually work? Unfortunately I have to replace a good old DECT network (I know it'll never compare to DECT)... Alberto -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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's don't register with 2.6.18
Louis-David Mitterrand wrote: Hello, Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. I'm running just 2.6.18 fine Under 1.2 Branch without issue. Connected to Asterisk SVN-branch-1.2-r44580 currently running on livonia (pid = 7349) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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's don't register with 2.6.18
On Fri, Oct 27, 2006 at 12:15:15PM -0400, Doug Lytle wrote: Louis-David Mitterrand wrote: Hello, Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. I'm running just 2.6.18 fine Under 1.2 Branch without issue. Connected to Asterisk SVN-branch-1.2-r44580 currently running on livonia (pid = 7349) Are you running Polycom's on this setup? ___ --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
[asterisk-users] Asterisk stopps matching extensions after first digit
Hi all,
I have problems receiving calls from PSTN with an Wildcard T207P.
All internal SIP devices have a 3 digit extension, e.g. 873.
When I call the extension from the PSTN this way everything works fine:
1. enter the number on the phone
2. lift off the handset
But when I call it that way Asterisk stopps matching the extension after the
first extension digit (8 in that case):
1. lift off the handset
2. enter the number on the phone
Asterisk then says that the extension does not exist and that the call is
rejected.
The context for receiving incoming calls in extensions.conf looks like this
[zap-in]
exten = _999[8-9]XX,1,Goto(internal,${EXTEN:3},1)
Is it possible to tell asterisk that it should match only 3 digits
extensions and where can this be configured? (extensions.conf, zapata.conf
or anywhere else)
Thanks in advance
Regards, Jens
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[asterisk-users] Re: polycom's don't register with 2.6.18
On Fri, Oct 27, 2006 at 05:11:24PM +0200, Louis-David Mitterrand wrote: Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but I'd really like to understand what's going on there... After disabling SIP NAT support in 2.6.18 kernel, Polycoms work again. ___ --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's don't register with 2.6.18
Louis-David Mitterrand wrote: On Fri, Oct 27, 2006 at 12:15:15PM -0400, Doug Lytle wrote: Louis-David Mitterrand wrote: I'm running just 2.6.18 fine Under 1.2 Branch without issue. Connected to Asterisk SVN-branch-1.2-r44580 currently running on livonia (pid = 7349) Are you running Polycom's on this setup? 95% of them are Polycom 301s with a few IP501s. We are running Firmware 1.5.2 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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
[asterisk-users] meet me
Please help I am using [EMAIL PROTECTED] 2.6 Since I enter the conference prompt its will ask for the password ,after that it said invalid conference number Remark the password is correct but it cant know that it have a conference number (555) == Parsing '/etc/asterisk/sip_notify.conf': Found -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 69.71.143.243 -- Executing Set(SIP/99909998-99d5, MEETME_ROOMNUM=555) in new stack -- Executing GotoIf(SIP/99909998-99d5, 0?READPIN) in new stack -- Executing Answer(SIP/99909998-99d5, ) in new stack -- Executing Wait(SIP/99909998-99d5, 1) in new stack -- Executing Read(SIP/99909998-99d5, PIN|enter-conf-pin-number||) in new stack -- Executing GotoIf(SIP/99909998-99d5, 0?USER) in new stack -- Executing GotoIf(SIP/99909998-99d5, 0?ADMIN) in new stack -- Executing Playback(SIP/99909998-99d5, conf-invalidpin) in new stack -- Playing 'conf-invalidpin' (language 'en') -- Executing Goto(SIP/99909998-99d5, READPIN) in new stack -- Goto (from-internal,555,5) -- Executing Read(SIP/99909998-99d5, PIN|enter-conf-pin-number||) in new stack -- Playing 'enter-conf-pin-number' (language 'en') -- User disconnected * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Asterisk PBX to a Nortel MICS PBX
Can you be more specific? What sort of linkages are available between the two offices? CP On 22-Oct-06, at 10:38 PM, dthurn wrote: What's the best way to connect an Asterisk PBX to a Nortel MICS PBX. I have two offices that I want to link together. TTFN ___ --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
[asterisk-users] Voicemail and OSX 10.4 Intel
Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav. I think tha could be something related to this : x=0, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav49, 0x518fe0 -- x=1, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav, 0x180a200 but I don't know what this means ... something I need to compile extra? thanyou in advance Dp. ___ --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] [OT] wi-fi ip phone scenario
You might check with Aastra, they are showing a DECT phone that will work with Asterisk via sip. I know the release is next year for me, but since you are in Europe it may be avaliable sooner. On 10/27/06, Alberto Pastore [EMAIL PROTECTED] wrote: Hello everyone.I know it's a little bit off-topic, but I was just wondering...Has anyone ever had any experience with asterisk,a wi-fi meshed lan (with more than one access point)and wi-fi sip phones? I made some tests but I'm not really satisfiedWi-fi phones are a curse (as far as I know evenNokia eSeries -I personally own an e70 model- have their flaws):- random sip registration failures - ridiculous battery life- bad audio quality even with optimal radio environments- crashes, system freezes- ...- slow responsiveness to asterisk qualify pings (OPTIONS)but I can even live with that. The major problem is... roaming between cells.Is that a dream or something that can actually work?Unfortunately I have to replace a good oldDECT network (I know it'll never compare to DECT)... AlbertoAlberto PastoreB-Press Srl - Gruppo MSoftP.IVA 01697420030P.le Lombardia, 4 - 28100 Novara - ItalyTel. 0321-499508Fax 0321-492974http://www.msoft.it ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --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] DTMF detection problem in PABX trunk
I'm already try this configuration, but don't have sucess.-- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br 2006/10/27, Al Bochter [EMAIL PROTECTED]: Check your dtmfmode I use dtmfmode=rfc2833 Check with your provider Best regards,Al BochterBochter Services(Voip PBX) Toll Free: 866-638-1254 EXT: 250(Voip PBX) Free World DialUp: 780217 EXT: 250(Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___--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 Inbound (clean). Database: 0643-6, 10/26/2006 - 10/27/2006 7:52:49 AM ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] Re: Re: ECHO Cancellation in SIP Calls
On Fri, Oct 27, 2006 at 08:10:30AM -0400, Steven wrote: If you are calling from a SIP phone through asterisk and through a Digium card, one could argue that the Digium card IS farside of the SIP phone. SIP -- asterisk -- Digium -- T1/PRI/analog -- PSTN -- PSTN -- PSTN -- Destination. If you're calling from a SIP phone through Asterisk to a digital PRI card to the PSTN, then your system has no hybrid. Except, potentally, the SIP handset itself. You may be required to cancel echo generated by other people's systems. E.g: somewhere on the PSTN. I would argue that the Digium card IS on the farside of asterisk as far as the SIP phone is concerned. We just switched from Legacy PBX to Asterisk and we get occasional echo. Everything past the Digium card is the same as the old PBX. We never got echo on the old PBX. How do you connect to the PSTN? Digital or analog? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
[asterisk-users] detecting ring
We have a large number of numbers (!) that we need to clean from our database. I've been asked if we can do this automatically, by checking if the number is valid or not from asterisk. what I don't want to do is to disturb the phone owners if the number is valid. obviously I can catch all the bad numbers, out of order etc by checking the hangup code. is there any way of intercepting the fact that the target number is about to ring, and hangup before that happens ? Julian ___ --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] Is SQLite a replacement for Mysql while using ARA in 1.2.x
On Fri, Oct 27, 2006 at 09:56:09AM -0500, Erick Perez wrote: Cohen, so you vote for the ARA-odbc-sqlite route? Can't think of anything better, now. But I haven't actually tried using it. this is for embedded, so that's why sqlite instead of mysql or postgres. when you say it is not guaranteed, what do you mean? The point of ARA is that others tools will manipulate the database. So you'll have to have other (independent) clients accessing this database to do somthing useful. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Asterisk and ISDN and Hylafax
On Fri, Oct 27, 2006 at 04:35:52PM +0200, Armin Schindler wrote: On Fri, 27 Oct 2006, Olivier wrote: 2006/10/27, Armin Schindler [EMAIL PROTECTED]: On Fri, 27 Oct 2006, Thomas Winter wrote: Am Thursday 26 October 2006 23:35 schrieben Sie: On Thu, 26 Oct 2006, Thomas Winter wrote: I would recommend the Eicon DIVA Server 4BRI cards. They have a capi interface which is used by chan-capi (chan-capi.org) and onboards DSPs for the faxing. You can use this for send and receive faxes and/or use capi4hylafax in parallel with asterisk/chan-capi. sounds good, you think it will run reliable? I do think so. I have this exactly this setup running twice. One setup even has 2 DIVA 4BRI cards, where the send one is running all ports in NT-mode to connect a legacy PBX. Regarding the 'parallel' hylafax, you just need to make sure that your setup is correct, e.g. asterisk should not be configured to accept calls which are meant for hylafax. Armin What about telephony features using chan-capi and Asterisk ? Are those features on par with msidn+Asterisk or bristuff+Asterisk (maybe I'm mixing up things together) ? You can compare it with e.g. misdn+Asterisk. chan-capi is just another channel driver for Asterisk which provides access to ISDN/POTS hardware and drivers which support the CAPI interface. This includes - standard voice - DTMF - echo-cancel - Line-Interconnect - Fax - RTP ... Don't mix it with bristuff. As far as I know, bristuff consists of a) some additional zap driver. and a replacement of much of chan_zap. It is said to have a suppreior ISDN stack. b) Asterisk changes/patches which alter asterisk features. [c) the old version of chan-capi] I know that there have been a bit of development of it. This is not chan-capi 0.3.5 . But I'm realy not familiar with it. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ISDN-BRI issue
On Fri, Oct 27, 2006 at 02:14:43PM +0200, Alberto Pastore wrote: Frédéric Blaise ha scritto: Hello all Asterisk 1.2.10 BRIstuff PRE-1s Debian sarge I have an issue with my ISDN-BRI line. Asterisk tells me my D-Channel is down, no matter is I have an actual line plugged in or not. == Primary D-Channel on span 1 down Try with signalling=bri_cpe even if your lines are set as point to multipoint, at least that should make your card trying to keep layer 1 up, even if this won't probably solve it. As a matter of fact I'm getting to conclude that bristuff + hfc-4s card is not working whatsoever. Works fine with Junghanns' cards. One simple thing for you to test: set one port in TE mode and one port in NT mode (move all 5 jumbers of that port to the other position to get it into NT mode). Then try to make a loopback connection (using a standard ethernet cable). Here you control both ends and thus there are less configuration pains. I believe there's something wrong in the way bristuff manages layer 1 (not sure if that's a driver problem, hardware problem or both). One thing that could be wrong is if both sides do not agree on the line settings. Where do you connect to? What do you have on zaptel.conf ? On zapata.conf? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
[asterisk-users] Re: Voicemail and OSX 10.4 Intel
On 2006-10-27 09:59:10 -0700, David Parcerisa [EMAIL PROTECTED] said: Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav. I think tha could be something related to this : x=0, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav49, 0x518fe0 -- x=1, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav, 0x180a200 but I don't know what this means ... something I need to compile extra? thanyou in advance Why are you using 1.2.1? try updating to something a bit fresher like 1.2.12.1;~) I have never seen any issue with this on my mac asterisk systems so I don't think it's something extra to build. You should see these in your /usr/lib/asterisk/modules by default. Did you mess around with your module loading or your modules? You might have screwed things up that way... Dunno really, just reaching, Marty ___ --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
[asterisk-users] Digium TE110P
Hi Groupies,I am sort of new to the whole asterisk thing, especially when it comes to the Digium TE110P card. Does anyone have experience setting this up? If so can you help me out? The provider for the PRI is going to be ATT/SBC.ThanksJulian ___ --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
[asterisk-users] Re: [OT] wi-fi ip phone scenario
On 2006-10-27 08:49:44 -0700, Alberto Pastore [EMAIL PROTECTED] said: Hello everyone. I know it's a little bit off-topic, but I was just wondering... Has anyone ever had any experience with asterisk, a wi-fi meshed lan (with more than one access point) and wi-fi sip phones? I don't think I have a mesh network technically speaking, but I have wired my how with two Zyxel X-550's one acting as a router to the internet and the other as a WDS repeater. This works well with my Nokia e60 (although the phone still has flaws). I made some tests but I'm not really satisfied Wi-fi phones are a curse (as far as I know even Nokia eSeries -I personally own an e70 model- have their flaws): - random sip registration failures Saw that when I has issues with WIFI signal, once I got that sorted AND update the firmware on my Nokia e60, this seems to be fixed. - ridiculous battery life Heh. No comparison to my old T68i which I would travel for a week with, and not bring the charger. The e60 is one day, if WIFI is used. - bad audio quality even with optimal radio environments Not true here. I am using ulaw and it sounds fine. - crashes, system freezes Yes there where many, but the firmware updater resolve many of these. NOT perfect, but much better. - ... - slow responsiveness to asterisk qualify pings (OPTIONS) Again, an issue of firmware and WIFI coverage but I can even live with that. The major problem is... roaming between cells. Is that a dream or something that can actually work? Mine works perfectly. Here is what I did. Set both bases to be locked to a particular channel(ie 6). Set both bases to be locked to 802.11b only.(don't know if this is necessary, but it works for me.) make sure that through both bases the same DHCP server is providing IP's (in my case neither of the WIFI bases is the DHCP server. Make sure they are close enough that there is no rough spot in the middle (for WDS this is closer then you might think). Set each WIFI router to have the mac address of the other in the WDS config. Make sure your network is secure. Random leeches can hurt you. Many WDS products ONLY support WEP, so this is an issue also. I am running 64 bit (40 bit) WEP, which is good enough to lock out the casual pucks. Good Luck, Marty Unfortunately I have to replace a good old DECT network (I know it'll never compare to DECT)... Never is a long time ;~) ___ --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
[asterisk-users] fully featured and robust * client gui?
Hi, My users are currently using a console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension, and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best gui? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --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] [OT] wi-fi ip phone scenario
Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). On 10/27/06, Alberto Pastore [EMAIL PROTECTED] wrote: Hello everyone.I know it's a little bit off-topic, but I was just wondering...Has anyone ever had any experience with asterisk,a wi-fi meshed lan (with more than one access point)and wi-fi sip phones? I made some tests but I'm not really satisfiedWi-fi phones are a curse (as far as I know evenNokia eSeries -I personally own an e70 model- have their flaws):- random sip registration failures - ridiculous battery life- bad audio quality even with optimal radio environments- crashes, system freezes- ...- slow responsiveness to asterisk qualify pings (OPTIONS)but I can even live with that. The major problem is... roaming between cells.Is that a dream or something that can actually work?Unfortunately I have to replace a good oldDECT network (I know it'll never compare to DECT)... AlbertoAlberto PastoreB-Press Srl - Gruppo MSoftP.IVA 01697420030P.le Lombardia, 4 - 28100 Novara - ItalyTel. 0321-499508Fax 0321-492974http://www.msoft.it ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Re: Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
I think the biggest issue with with telemarketers. I get blatently illegal calls all the time, besides the fact that I am on the do not call lists. Today I got a call from some group trying to sell me a Razr phone for $50, automated computer, no option to remove yourself and the callerid appears valid but when you call it you discover its not a valid number. Since I get this particular call every few weeks I've tried to talk to them, but even pretending to be interested in their scam they won't say anything more than its a t-mobile phone and sure as hell won't give me an address or valid telephone number. On 10/27/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For my home Asterisk setup I have a single PSTN line, and then I use a variety of different voip providers. I use two different providers for my DID's (one toll free, and one normal). I use yet a different provider for terminating outgoing calls. So, when making an outgoing call via voip, what number should I use to identify myself? I currently use the number of my PSTN line, since that is our publicinbound number.Hi John!I have same situation, and I certainly agree about everything you said. --Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)270248Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]e-mail: tparcina#lama.hr http://www.lama.hr___--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] Digium TE110P
Julian Varanini wrote: Hi Groupies, I am sort of new to the whole asterisk thing, especially when it comes to the Digium TE110P card. Does anyone have experience setting this up? If so can you help me out? The provider for the PRI is going to be ATT/SBC. ATT/SBC is pretty much a standard setup. Check out the following links for PRI/ISDN configuration: http://www.voip-info.org/wiki-PRI http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf The Wiki and Google is your friend: http://www.voip-info.org Welcome a board! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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
[asterisk-users] New Asterisk-GUI?
Hello, I am currently running 1.4-Beta3 on my test system and have enabled the new HTTP functionality. I have enabled http and web in both http.conf and manager.conf. I can succefuly reach: http://localhost:8088/asterisk/httpstatus http://localhost:8088/asterisk/static/ajamdemo.html My question is where can I find the GUI that Digium was demonstrating at Astricon? Is there an additional package or add-on I need to download? Is there a readme available? Thanks, Larry ___ --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
[asterisk-users] Confused about SIP Realtime Updates
I'm confused about SIP realtime updates. If I make a database change, and then do a "sip prune realtime peer peer", I can see Asterisk query the database, and retrieve the updated information. However, it still uses the old values. What's up with that? If I do a "reload", Asterisk queries the database and this time uses the new values. I have rtcachefriensd=yes in sip.conf. Thanks, Doug. ___ --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] New Asterisk-GUI?
On Fri, 27 Oct 2006 13:38:51 -0500, LJ wrote Hello, I am currently running 1.4-Beta3 on my test system and have enabled the new HTTP functionality. I have enabled http and web in both http.conf and manager.conf. I can succefuly reach: You have to download it manually from SVN as they mentioned in Astricon. Many people get confused and think that the new http manager interface is a web gui but it is the equivalent of the manager interface that exists on 1.2 but with support for connections using a browser. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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
[asterisk-users] Zultys Phones w/ Encryption
I've got a Zultys WIP2 and Zultys 2x2 both of which support encryption. I have patched my asterisk with srtp (srtp.sourceforce.net) as well as with the patch found at http://bugs.digium.com/view.php?id=5413. I'm trying to utilizing the encryption feature of the two Zultys phones to create an encrypted call, but am having extreme difficulty. I keep getting the following error message of: Oct 27 15:48:51 WARNING[5638]: chan_sip.c:4420 process_sdp: Can't provide secure audio requested in SDP offer Does anyone have any experience getting this to function correctly? Scott Higginbotham Systems / Network Operations Manager 215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED] ___ --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
[asterisk-users] Re: Snom, mute and rtptimeout
I am not familiar with the SNOM phone. On some mfg phones I think they have a setting to enable transmit silence. See if Snom has such a setting. Benny Amorsen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a bunch of Snom phones. When I press the mute button, the phone stops sending RTP frames. If I have rtptimeout set, that means that the connection will eventually be cut off. It also affects sound generated by asterisk, since timing is generated from the incoming frames. Are there any workarounds I can try? Disabling rtptimeout is one, obviously. /Benny ___ --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
[asterisk-users] Vancouver Asterisk User Group
Greetings, This is my annual post-Astricon attempt to start an Asterisk User Group in the Vancouver, BC, area. If you are interested, please reply off-list. Regards, -- Anthony Rodgers (CunningPike) Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --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
[asterisk-users] Enterprise Asterisk User Group
Greetings, This is my annual post-Astricon attempt to get an Enterprise Asterisk User Group off the ground. We are a municipal government using Asterisk to replace a legacy PBX. I'd be interested in starting a group of similar enterprise users (say, 100 seats or more) other than resellers, carriers and call-centers who are using Asterisk to support their non-telecom-related business - I don't envisage any geographical limitation to the group (there seem to be few enough of us as it is!). If you are interested, please let me know off-list. Regards, -- Anthony Rodgers (CunningPike) Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --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
[asterisk-users] Voicemail 'exitcontext'
This seems to be a bug. I can get exitcontext to work on a per mailbox basis in voicemail.conf. However, for realtime mailboxes, I added a new column called 'exitcontext' to my table, and the thing simply doesn't work. I can see asterisk selecting * from the table, but pressing 0 while in voicemail has no effect. If I press 0 while in voicemail for a mailbox defined in voicemail.conf, it works fine. Is this a known issue? Doug. ___ --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
[asterisk-users] autocreate peer + sippeers table entry = auth required?
In my setup, sip calls coming in through a proxy with a sip.conf entry set to autocreatepeer=yes and context=proxy get placed into context proxy in the dial plan. That is expected.However, if the username in the From: address exists in the sippeers table, it gets challenged for the password and on success is dropped into context default, even if the sip domain is not being served by asterisk. What could be the problem?Here is my sip.conf:[general]Autocreatepeer = nocontext=defaultdomain = b.comdomain = sip.b.com realm=b.combindport=5060bindaddr=4.2.2.2allow=g729,ulaw,alaw,speex,gsmdtmfmode=rfc2833rtcachefriends=yes;bindaddr=0.0.0.0 srvlookup=yesrtpkeepalive=1000rtupdate=yesport=5060defaultexpirey=3600tos=0x18insecure=no[ser]type=peercontext=serhost=4.2.2.3canreinvite=yes inseure=veryautocreatepeer=yes ___ --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] SIP problem - ACT p160s error
joe, at j4computers wrote: Thanks. I will give that a try. Do you know if removing that line will affect other phones I might have? If so, maybe I am better off getting someone else's phone. ACT's support seems a bit problematic. They responded to my first email right away, but never, so far, to my second. Then I realized their first one was a brush off, simply saying essentially only you are having this problem. joe a. [EMAIL PROTECTED] Wrote on: 10/26/2006 11:40 PM: I saw this problem before... to solve that, I needed to hack asterisk to remove a header SIP field. Check your ACT phone log, and you can figure out which filed is that. Then, comment that filed from your chan_sip.c and recompile asterisk.. and that's it.. it only happens with ACT phones. Strange, I have a P160S sitting on my desk and I have no problems calling and receiving calls from my other phones (Cisco 7940 and analog). I'm running firmware v2.08. Leo ___ --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
[asterisk-users] Waiting before executing System command
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten = h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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Re: [asterisk-users] Waiting before executing System command
what about
exten = h,n,System(mycommand /some/file /some/other/dir/)
Where mycommand is your custom shell script to sleep before moving the file.
On 10/27/06, Alexander Burke [EMAIL PROTECTED] wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten = h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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[asterisk-users] 0 channels configured with tdm400 (tdm04b rev. G)
Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel = 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with Asterisk Version 1.2.13 Zaptel Version 1.2.10 Libpri Version 1.2.4 Physically looking at the card, the four FXO ports have the green led turned on. It has no IRQ conflicts and zaptel compiled cleanly. Kernel is 2.6.9-42ELsmp (it's a dual core Intel machine in an Intel 945G board) Your comments are welcomed. ___ --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
[asterisk-users] dialing external number within meetme
hello, is it possible to dial out external number within running conference, for example dial out using zap channel and connect to pstn conference, thx bart ___ --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
