Re: [asterisk-users] Unable to pickup an extension
The only solution I see to have a PICKUPMARK-variable created on an
incoming channel, and have the same PICKUPMARK on another created
channel (the one that does the pickup) is to work with a database like
MySQL.
I see no other way to separate multiple incoming channels (with their
own PICKUPMARK) and have the different PICKUPMARK's available for the
local Phone (with a new channel) that does the Pickup.
You see, I have multiple incoming lines, and the resulting Phones that
need to ring are set dynamically. So it's perfectly possible that one
time phone1 and phone2 ring, and the other time phone2 and phone4.
When I press the BLF-button on phone4, I do not want the incoming call
that rang phone1/phone2.
So in my case, the local phone I'm trying to pick up determines the
incoming channel that needs to be picked up.
I need some identifier inside the PICKUPMARK-variable that uniquely
defines which phone/phones are rang. This info is simply not available
on the channel that is doing the pickup.
I see no other option than working with a database to write this data into.
Jonas.
On 06/17/2010 10:14 PM, Philipp von Klitzing wrote:
Hi!
exten = **XX
-- This is a local extension, a certain phone which is monitored with
BLF-lights. So if I press the button I want the phone call that made this
phone ring, not another phone.
This is NOT a local extension: It is a special local PICKUP extension
(you even named it [example-pickup]). So, in order to pickup an inbound
call that has been placed to ...67 you need to dial **67.
Small note for SNOM phones: Those can be configured to prefix a monitored
extions (67) with f.e. ** so that the phone dials **67 when pressing a
blinking BLF button.
exten = 1234567,n,Set(_PICKUPMARK=${EXTEN:5})
-- If I set the PICKUPMARK-variable the same as the DID that is called,
how can I know which DID that is set in the context [example-pickup] ??
My phone, who tries to do the pickup, creates a new channel, and the
PICKUPMARK-channel variable is of course not inherited to this newly
created channel...
I am not sure I got your question, but the phone the does the pickup does
not need to worry about inheritance of the PICKUPMARK variable - it
_sets_ that variable itself (see code for **XX) and then ask Asterisk to
search for a corresponding chanel with that value that is currently in
RINGING state.
The variable inheritance only matters for the device that is being
called, in this case IPphone-1, because that is the phone that you want
to relieve of its inbound caller.
In general: PICKUPMARK is for directed pickup where you know exactly
which call you want. Look at *8 undirected pickup (and pickup groups) if
you just want to catch whatever just started calling me.
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weaknesses of Asterisk still there?
Hello Out of curiosity, are those weaknesses still there in Asterisk 1.6, or have they been fixed? How does FreeSWITCH compare to Asterisk? http://www.freeswitch.org/node/117 Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] device or sound card busy
Hi guys, Thanx a lot to all of you. My call is now forwarded to sip form PSTN, but again a new problem is coming. When i pick up the call from my softphone it says the can not access speaker or microphone. But i have my headphone plugged in and in working stage. on softphone: Fri 18:08:17 Warning: Failed to open sound card: Device or resource busy this message is displayed and on CLI -- SIP/2001-081fa758 is ringing -- Got SIP response 480 User not responding back from 172.26.48.113 -- SIP/2001-081fa758 is circuit-busy is displayed, even though i pick up, i can hear the ringing tone in the phone which called and not able to talk . What may be the problem. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}
hi,all
for a long time, i cant understand the difference between
${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}
i know ${CDR(start)} mean when a call is start. and ${CDR(answer)}
means when a call was pick up.
but what's ${CDR(calldate)} mean?
Could you help me ?
Thansk a lot!
--
Thanks for your supporting,
have a nice day.
Sucan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDRs not getting generated on Free PBX
Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijha...@oxygen8.com Skype: deepika-nijhawan W: http://www.oxygen8.com/ www.oxygen8.com This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the intended recipient/s. If you are not the intended recipient/s please note that any distribution, copying or use of this communication or the information in it is strictly prohibited. If you have received this communication in error please notify us by email or by telephone (08082060808) and then delete the email and any copies of it. This communication is from Oxygen8 Communications UK Ltd - Company Number 03383285. Registered Address; 12th Floor, Lyndon House, 58-62 Hagley Road, Birmingham, B16 8PE. VAT Registration Number: 792 4494 89 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic attendant - Error in CLI.
Hello dear list.
I am currently working on a Automatic attendant, and the core things work, but
I think the loop function isn't working as expected.
I am testing this environment: a sip internal call from 301 to 501.
The setup here is when 301 calls 501, and 301 doesn't enter an extension, it
will go in loop, 3 times, and then hangup...Can't get that working.
Could someone please help me?
Extensions.conf
[mainmenu]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n,(LoopEnd),EndWhile
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
CLI Output
Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Answer(SIP/301-0248, ) in new stack
-- Executing [...@phones:2] Wait(SIP/301-0248, 2) in new stack
-- Executing [...@phones:3] Playback(SIP/301-0248,
velkommen_abacus) in new stack
-- SIP/301-0248 Playing 'velkommen_abacus.slin' (language 'en')
-- Executing [...@phones:4] Set(SIP/301-0248, Loop=0) in new stack
-- Executing [...@phones:5] While(SIP/301-0248, 1) in new stack
-- Executing [...@phones:6] BackGround(SIP/301-0248, tast123vent_)
in new stack
-- SIP/301-0248 Playing 'tast123vent_.slin' (language 'en')
-- Executing [...@phones:7] WaitExten(SIP/301-0248, 5) in new stack
-- Timeout on SIP/301-0248, continuing...
-- Executing [...@phones:8] Set(SIP/301-0248, Loop=1) in new stack
[Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No
application '' for extension (phones, 501, 9)
== Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-0248'
asterisk*CLI
sip.conf regarding sip 501
[501]
type=friend
secret=XX
host=dynamic
context=phones
mailbox=...@default
callerid=Sentralbord
qualify=yes
Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner
Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic attendant - Error in CLI.
On Fri, Jun 18, 2010 at 10:51:40AM +0200, Aksel Celasun wrote:
Hello dear list.
I am currently working on a Automatic attendant, and the core things work,
but I think the loop function isn't working as expected.
I am testing this environment: a sip internal call from 301 to 501.
The setup here is when 301 calls 501, and 301 doesn't enter an extension, it
will go in loop, 3 times, and then hangup...Can't get that working.
Could someone please help me?
Extensions.conf
[mainmenu]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n,(LoopEnd),EndWhile
This should be:
exten = 501,n(LoopEnd),EndWhile
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
CLI Output
Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Answer(SIP/301-0248, ) in new stack
-- Executing [...@phones:2] Wait(SIP/301-0248, 2) in new stack
-- Executing [...@phones:3] Playback(SIP/301-0248,
velkommen_abacus) in new stack
-- SIP/301-0248 Playing 'velkommen_abacus.slin' (language 'en')
-- Executing [...@phones:4] Set(SIP/301-0248, Loop=0) in new stack
-- Executing [...@phones:5] While(SIP/301-0248, 1) in new stack
-- Executing [...@phones:6] BackGround(SIP/301-0248,
tast123vent_) in new stack
-- SIP/301-0248 Playing 'tast123vent_.slin' (language 'en')
-- Executing [...@phones:7] WaitExten(SIP/301-0248, 5) in new
stack
-- Timeout on SIP/301-0248, continuing...
-- Executing [...@phones:8] Set(SIP/301-0248, Loop=1) in new stack
[Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No
application '' for extension (phones, 501, 9)
You put '(LoopEnd)' in the place for the application. Hence empty
application with 'LoopEnd' as its input.
== Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-0248'
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic attendant - Error in CLI.
Extensions.conf
[mainmenu]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n,(LoopEnd),EndWhile
This should be:
exten = 501,n(LoopEnd),EndWhile
I don't understand, i do have the same thing you wrote above.
Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Answer(SIP/301-0248, ) in new stack
-- Executing [...@phones:2] Wait(SIP/301-0248, 2) in new stack
-- Executing [...@phones:3] Playback(SIP/301-0248,
velkommen_abacus) in new stack
-- SIP/301-0248 Playing 'velkommen_abacus.slin' (language 'en')
-- Executing [...@phones:4] Set(SIP/301-0248, Loop=0) in new stack
-- Executing [...@phones:5] While(SIP/301-0248, 1) in new stack
-- Executing [...@phones:6] BackGround(SIP/301-0248,
tast123vent_) in new stack
-- SIP/301-0248 Playing 'tast123vent_.slin' (language 'en')
-- Executing [...@phones:7] WaitExten(SIP/301-0248, 5) in new
stack
-- Timeout on SIP/301-0248, continuing...
-- Executing [...@phones:8] Set(SIP/301-0248, Loop=1) in new stack
[Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No
application '' for extension (phones, 501, 9)
You put '(LoopEnd)' in the place for the application. Hence empty
application with 'LoopEnd' as its input.
== Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-0248'
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs not getting generated on Free PBX
Cdr status shows: CDR logging: enabled CDR mode: simple CDR output unanswered calls: no It is not showing 'CDR registered backend' Thanks, Deepika From: Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com] Sent: 18 June 2010 09:37 To: 'asterisk-users@lists.digium.com' Subject: CDRs not getting generated on Free PBX Hi, We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and asterisk is connecting to it. CDR modules are all loaded as well. For some reason, it is not creating master.csv and no cdrs are generated. Can anyone help please. --- Kind Regards, Deepika Nijhawan VoIP Engineer Oxygen8 Communications T: +44(0) 871 434 9151 +44(0) 121 620 9151 Email: deepika.nijha...@oxygen8.com Skype: deepika-nijhawan W: www.oxygen8.com http://www.oxygen8.com/ This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the intended recipient/s. If you are not the intended recipient/s please note that any distribution, copying or use of this communication or the information in it is strictly prohibited. If you have received this communication in error please notify us by email or by telephone (08082060808) and then delete the email and any copies of it. This communication is from Oxygen8 Communications UK Ltd - Company Number 03383285. Registered Address; 12th Floor, Lyndon House, 58-62 Hagley Road, Birmingham, B16 8PE. VAT Registration Number: 792 4494 89 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic attendant - Error in CLI.
Aksel Celasun wrote: This should be: exten = 501,n(LoopEnd),EndWhile I don't understand, i do have the same thing you wrote above. The difference between yours and his is that you had a n,(LoopEnd) and it should be n(LoopEnd) 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 http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs not getting generated on Free PBX
On 18 June 2010 10:38, Deepika Nijhawan deepika.nijha...@oxygen8.comwrote: Cdr status shows: CDR logging: enabled CDR mode: simple CDR output unanswered calls: no It is not showing ‘CDR registered backend’ Thanks, Deepika Have you compiled asterisk-addons and selected to compile and install cdr_addon_mysql ? Stephen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic attendant - Error in CLI.
Ah, I missed the comma, thank you, and thank you Tzafrir Cohen! Best regards Aksel -Opprinnelig melding- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Doug Lytle Sendt: 18. juni 2010 11:41 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Automatic attendant - Error in CLI. Aksel Celasun wrote: This should be: exten = 501,n(LoopEnd),EndWhile I don't understand, i do have the same thing you wrote above. The difference between yours and his is that you had a n,(LoopEnd) and it should be n(LoopEnd) 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 http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk issue
Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday June 18th at 12 Noon EDT: Session Border Controllers, 1PM Bria iPhone SIP app
This week our guest at 12 noon EDT (http://vuc.me/next for your local time) is Acme Packet, maker of Session Border Controllers. We look forward to learning more about these. Join in G722 wideband by calling sip:200...@login.zipdx.com starting just before 12 Noon EDT or Skype:vuc.me or see http://vuc.me for more ways to connect. At 1PM EDT, Counterpath will join us to talk about their new Bria iPhone app. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Physical SIP phone with inbuilt VPN support
Hi, all Would any of you be able to suggest physical SIP phones that support inbuilt VPN capabilities; akin to the Snom 370/870 ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Physical SIP phone with inbuilt VPN support
I have a Cisco SPA525G that seems to support it, but I've never needed that. I would assume most Cisco SPA phones would support that too. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: Friday, June 18, 2010 8:25 To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT: Physical SIP phone with inbuilt VPN support Hi, all Would any of you be able to suggest physical SIP phones that support inbuilt VPN capabilities; akin to the Snom 370/870 ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hello again dear list.
Could you please help with this?
Thank you for all support, you are great, and i am now at a late stage in the
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I
have a playback function there.
But CLI reports:
CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes:
Context 'internal' tries to include nonexistent context
'nighttime|12:30-8:00|mon-fri|*|*'
Extensions.conf
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include = internal
include = hovedmeny
[internal]
include = to_SIPtrunk
include = nighttime
exten = _10X,1,NoOp()
exten = _10X,n,Dial(SIP/${EXTEN},10)
exten = _10X,n,Playback(kuntiltestt_)
;exten = _10X,n,Playback(vm-nobodyavailtt-monkeysintrott-monkeys)
exten = _10X,n,Hangup()
exten = 4767209600,1,NoOp();
exten = 4767209600,n,Verbose(Callerid num ${CALLERID(num)});
exten = 4767209600,n,Dial(SIP/501,5);
;exten = 4767209600,n,Background(velkommen_abacustast123vent_);
;exten = 4767209600,n,WaitExten;
;exten = 4767209600,1,Dial(SIP/200,15);
;exten = 4767209600,1,Goto(submenu,s,1);
exten = 4767209600,n,Playback(kuntiltestt_);
exten = 4767209600,n,Hangup();
[hovedmeny]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n(LoopEnd),EndWhile()
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
[nighttime]
exten = s,1,Wait(2);
exten = s,n,Playback(tt-somethingwrong);
exten = s,n,Hangup;
[to_SIPtrunk]
exten = _[2-9]XXX, 1, Macro(dial-trunk-sip,${EXTEN});
exten = _0, 1, Macro(dial-trunk-sip,${EXTEN});
[incoming]
exten = s,1,Noop();
exten = s,n,Verbose(Call ${EXTEN});
exten = s,n,Dial(SIP/501);
exten = s,n,Hangup();
[macro-dial-trunk-sip]
exten = s,1,Noop(${ARG1},${CALLERID(num)})
exten = s,n,Set(CALLERID(num)=67209600)
exten = s,n,Dial(SIP/phonect_01/${ARG1})
exten = s,n,Hangup
exten = s,n,MacroExit
Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner
Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
I do a cron to execute /usr/sbin/asterisk -rx restart when convenient each day at 4:45 AM. This doesn't really solve any problems, just does housekeeping to keep a clean environment, since some installs/os'es lend themselves to memory leaks. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Friday, June 18, 2010 5:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart asterisk service asterisk restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|*
Correct now.
Fra: Aksel Celasun
Sendt: 18. juni 2010 14:30
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: Error trying to add context: Context 'internal' tries to include
nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hello again dear list.
Could you please help with this?
Thank you for all support, you are great, and i am now at a late stage in the
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I
have a playback function there.
But CLI reports:
CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes:
Context 'internal' tries to include nonexistent context
'nighttime|12:30-8:00|mon-fri|*|*'
Extensions.conf
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
include = internal
include = hovedmeny
[internal]
include = to_SIPtrunk
include = nighttime|12:30-8:00|mon-fri|*|*
exten = _10X,1,NoOp()
exten = _10X,n,Dial(SIP/${EXTEN},10)
exten = _10X,n,Playback(kuntiltestt_)
;exten = _10X,n,Playback(vm-nobodyavailtt-monkeysintrott-monkeys)
exten = _10X,n,Hangup()
exten = 4767209600,1,NoOp();
exten = 4767209600,n,Verbose(Callerid num ${CALLERID(num)});
exten = 4767209600,n,Dial(SIP/501,5);
;exten = 4767209600,n,Background(velkommen_abacustast123vent_);
;exten = 4767209600,n,WaitExten;
;exten = 4767209600,1,Dial(SIP/200,15);
;exten = 4767209600,1,Goto(submenu,s,1);
exten = 4767209600,n,Playback(kuntiltestt_);
exten = 4767209600,n,Hangup();
[hovedmeny]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten = 501,n(LoopEnd),EndWhile()
exten = 501,n,Hangup()
exten = 1,1,Playback(tt-weasels)
exten = 1,2,Dial(SIP/200,10,rg)
exten = 1,3,Hangup()
exten = 2,1,Playback(tt-monkeys)
exten = 2,n,Dial(SIP/302,60,rg)
exten = 2,n,Hangup()
exten = 3,1,Dial(SIP/402,60,rg)
exten = 3,n,Hangup
exten = 9,n,Hangup()
exten = i,1,Set(Loop=$[${Loop}+1])
exten = i,n,Goto(LoopEnd)
exten = t,1,Set(Loop=$[${Loop}+1])
exten = t,n,Goto(LoopEnd)
[nighttime]
exten = s,1,Wait(2);
exten = s,n,Playback(tt-somethingwrong);
exten = s,n,Hangup;
[to_SIPtrunk]
exten = _[2-9]XXX, 1, Macro(dial-trunk-sip,${EXTEN});
exten = _0, 1, Macro(dial-trunk-sip,${EXTEN});
[incoming]
exten = s,1,Noop();
exten = s,n,Verbose(Call ${EXTEN});
exten = s,n,Dial(SIP/501);
exten = s,n,Hangup();
[macro-dial-trunk-sip]
exten = s,1,Noop(${ARG1},${CALLERID(num)})
exten = s,n,Set(CALLERID(num)=67209600)
exten = s,n,Dial(SIP/phonect_01/${ARG1})
exten = s,n,Hangup
exten = s,n,MacroExit
Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner
Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR extension dialing error
Hi, I tell you I've made some calls from a land-phone to my IVR in order to avoid the possible poor quality of cell phone's DTMF, and when I called extension 1003 I was connected to extension 1000 againthe same error. My IVR says dial 1 to connect to operator or dial the extension in case you know.and my extension ranges is 1000-1999, so I think it could be a problem that extensions and IVR option start with the same digit: 1. When I'll be at work I'm thinking in modify the IVR speech in order to say dial 0 to connect to operator., and not dial 1 to connect to operator, so IVR option and extensions will not start with the same digit. Do you think this may be the problem ??? Thanks a lot and sorry for my interruption. Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: According to this link http://www.smallnetbuilder.com/content/view/30469/82/1/2/ You probably want to make 80 be 120. This is a millisecond delay value, so the 500 value is a give it up proposition; 200 might be doable for your outliers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Thursday, June 17, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error OK, now I understand..but just one more question...In the DTMF settings tab from the GSM gateway manager I have this line: Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms What does this setting really mean and do I have to modify the current value ??? Final thanks :) 2010/6/17 Zeeshan Zakaria zisha...@gmail.com: I once setup a callback system for someone and we had these DTMF issues on constant basis, and all the complains were from cell phone users. At that time I found out that even my own cellphone would not DTMF correctly from certain locations, including my home, but would work perfectly fine from my work location. Probably times of the day matters too, but yes, calling from cell phones does result in DTMF issues, and the reason is that it is just the audio signals, which get distorted based on various factors like the signal strength, cell tower transmission quality, transcodings, etc. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF pass... -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR extension dialing error
I would definitely change the prompt from 1 to 0. It is not an advisable practice to have an IVR selection that can be misinterpreted like this. Assuming that all of your extensions are in 1000-1999, 2 for the operator would be just as good; the important thing is that you don't have a single digit extension 1. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Friday, June 18, 2010 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error Hi, I tell you I've made some calls from a land-phone to my IVR in order to avoid the possible poor quality of cell phone's DTMF, and when I called extension 1003 I was connected to extension 1000 againthe same error. My IVR says dial 1 to connect to operator or dial the extension in case you know.and my extension ranges is 1000-1999, so I think it could be a problem that extensions and IVR option start with the same digit: 1. When I'll be at work I'm thinking in modify the IVR speech in order to say dial 0 to connect to operator., and not dial 1 to connect to operator, so IVR option and extensions will not start with the same digit. Do you think this may be the problem ??? Thanks a lot and sorry for my interruption. Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: According to this link http://www.smallnetbuilder.com/content/view/30469/82/1/2/ You probably want to make 80 be 120. This is a millisecond delay value, so the 500 value is a give it up proposition; 200 might be doable for your outliers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Sent: Thursday, June 17, 2010 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IVR extension dialing error OK, now I understand..but just one more question...In the DTMF settings tab from the GSM gateway manager I have this line: Mobile DTMF debounce: 80 (range: 40 ~ 500, default: 80 ) step: 10ms What does this setting really mean and do I have to modify the current value ??? Final thanks :) 2010/6/17 Zeeshan Zakaria zisha...@gmail.com: I once setup a callback system for someone and we had these DTMF issues on constant basis, and all the complains were from cell phone users. At that time I found out that even my own cellphone would not DTMF correctly from certain locations, including my home, but would work perfectly fine from my work location. Probably times of the day matters too, but yes, calling from cell phones does result in DTMF issues, and the reason is that it is just the audio signals, which get distorted based on various factors like the signal strength, cell tower transmission quality, transcodings, etc. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Danny, so you say it's a problem of the cell phone and not the Astreisk or GSM Gateway ??? OK, in this case if I call from a fixed phone (not a cell phone) to the IVR, the DTMF quality problem will not be presentthis may be a good test, isn't it ??? Or do you suggest another test I can implement ??? Thanks again Alejandro 2010/6/17 Danny Nicholas da...@debsinc.com: The physical location of the phone (access to towers) can vastly affect the quality of DTMF pass... -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on nortel sip connection
I am using asterisk 1.4.32 and wish to connect using SIP to a nortel 1000 switch with the ability to have 90 calls at a one time outgoing or incoming. the nortel reseller is asking me what to do. I dont know nortel at all. I thought I just needed a SIP trunk and IP address of the their server and an account name, and provide her my IP address. They didn't know what to do with that. What do I tell them? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 18, 2010 9:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? _ Anahi Ludueña _ From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... _ Anahi Ludueña _ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, _ Anahi Ludueña _ Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! http://serviciosmoviles.es.msn.com/hotmail/yoigo.aspx _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! http://www.ayudartepodria.com _ Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
Hi, As Danny said, asterisk is looking for slin or ulaw files. Are your wav files in any of these formats? Did you just copied them from somewhere without changing their format? Also note they should be 8KHz mono 16 bit files. You can do this in a simple utility like Windows Recorder. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-18 10:35 AM, Danny Nicholas da...@debsinc.com wrote: Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Anahi Ludueña *Sent:* Friday, June 18, 2010 9:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? Anahi Ludueña ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
what do you mean unblock the calls exactly? -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Fri, 18 Jun 2010 11:12:55 +0100 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart asterisk service asterisk restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Relaunch of the Kansas City Asterisk User Group
Just a note to anyone in the Kansas City area that I've relaunched the KCAUG website/group at http://kcaug.org. Please drop by and join the group. :) -- Kyle Sexton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
The list of /var/lib/asterisk/mohmp3 is: -rw-rw 4 asterisk asterisk 184 Oct 19 2009 LICENSE-asterisk-moh-freeplay-wav -rw-rw-r-- 4 asterisk asterisk 882748 Oct 19 2009 QuajiroPromo.sln -rw-rw-r-- 4 asterisk asterisk 834682 Oct 19 2009 TristeAlegriaPromo.sln -rw-rw 4 asterisk asterisk 1939794 Oct 19 2009 fpm-calm-river.wav -rw-rw 4 asterisk asterisk 2582196 Oct 19 2009 fpm-sunshine.wav -rw-rw 4 asterisk asterisk 2217318 Oct 19 2009 fpm-world-mix.wav And the musiconhold.conf is: [default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [none] mode=files directory=/dev/null Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 09:26:16 -0500 Subject: Re: [asterisk-users] Music on Hold problema Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 18, 2010 9:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
do you have any tool in order to check what happened in asterisk during the hangs of calls 2010/6/18 salaheddine elharit salah.elharit...@gmail.com Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}
On Friday 18 June 2010 03:21:32 Zhang Shukun wrote:
hi,all
for a long time, i cant understand the difference between
${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}
i know ${CDR(start)} mean when a call is start. and ${CDR(answer)}
means when a call was pick up.
but what's ${CDR(calldate)} mean?
It could mean whatever you want. CDRs (at least the internal representation)
have support for arbitrary additional variables. Whether a particular backend
has support to carry those over into permanent storage is another question
(in 1.6.2, most CDR backends have it, as long as the underlying table has a
column to receive the data).
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
On Friday 18 June 2010 09:49:39 Warren Selby wrote: On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. Actually, it is an old method that still works, but as Warren mentioned, you should endeavor to switch to using GotoIfTime, as there's a nasty race condition inherent in using timed includes. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why asterisk down when inet server down?
We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
Just a guess - the phones are down because they can't get to the DCHP server. If you can't ping 10.10.10.44 you'll never reach 10.10.10.180. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, June 18, 2010 11:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Why asterisk down when inet server down? We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
- sean darcy seandar...@gmail.com wrote: We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? Are your network settings correct or is that a typo? We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 ...which tells me your mask should probably be 255.255.0.0. BUT, in the phone config, you have the mask as 255.255.255.0. This might not tbe cause and 'be-all end-all' of your issues, but it's worth some attention. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On Fri, 18 Jun 2010, sean darcy wrote: We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? If you mean that the phones simply can't dial and don't actually un-register themselves, it probably because asterisk needs a working DNS server somewhere, and with no Internet, then your DNS server - which looks like your gateway - which I'm guessing is just a dumb forwarder - is failling. I've never undersood quite why asterisk wants to do a DNS lookup for every call, but not been bothered enough to do something about it other than to run a cacheing DNS server on the asterisk box itself just for it's own use. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 12:53 PM, Danny Nicholas wrote: Just a guess - the phones are down because they can't get to the DCHP server. If you can't ping 10.10.10.44 you'll never reach 10.10.10.180. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, June 18, 2010 11:40 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Why asterisk down when inet server down? We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? sean I can ping 10.10.10.44, and 10.10.10.180. In fact I can log into the web server on 10.10.10.44 when the internet server is down. The 10.10.10.0 network works fine. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 12:57 PM, Tim Nelson wrote: - sean darcyseandar...@gmail.com wrote: We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? Are your network settings correct or is that a typo? We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 ...which tells me your mask should probably be 255.255.0.0. BUT, in the phone config, you have the mask as 255.255.255.0. This might not tbe cause and 'be-all end-all' of your issues, but it's worth some attention. --Tim Yes, it is a typo. The network is 10.10.10.0/255.255.255.0. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 01:19 PM, Gordon Henderson wrote: On Fri, 18 Jun 2010, sean darcy wrote: We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ] Enabled IP Address 10.10.10.44_ Subnet Mask 255.255.255.0___ Gateway 10.10.10.180 Primary DNS 10.10.10.180 Secondary DNS0.0.0.0_ If the network server is down, the phones go out of service. From a workstation I can still ssh into the asterisk server. But for some reason the phones don't work. I can console dial out from the server. The asterisk server doesn't need the internet for connectivity. Why do the internal phones go down? If you mean that the phones simply can't dial and don't actually un-register themselves, it probably because asterisk needs a working DNS server somewhere, and with no Internet, then your DNS server - which looks like your gateway - which I'm guessing is just a dumb forwarder - is failling. I've never undersood quite why asterisk wants to do a DNS lookup for every call, but not been bothered enough to do something about it other than to run a cacheing DNS server on the asterisk box itself just for it's own use. Gordon I don't know if the phones unregister themselves. They simply show No service and sip show peers: 44/44(Unspecified)D N A 5060 UNKNOWN I do run a caching/nameserver on the asterisk box. I didn't realize asterisk did a DNS lookup for every call, even one going out over PRI/DAHDI. That makes no sense, IMHO. It means that you couldn't use asterisk as a straight PBX for a T1 (or POTS). Is there any way of turning off the DNS lookup? Or at least not having it become a major failure event? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
Based on my somewhat similar experience a few times, when this happens, try to resolve an IP, to make sure there is a valid DNS server accessible to Asterisk. If not, either make asterisk a DNS as well, or remove any domain name entries from /etc/resolv file and replace them with the IP addresses of your DNS. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-18 1:29 PM, sean darcy seandar...@gmail.com wrote: On 06/18/2010 12:57 PM, Tim Nelson wrote: - sean darcyseandar...@gmail.com wrote: We h... Yes, it is a typo. The network is 10.10.10.0/255.255.255.0. sean -- _ -- Bandwidth and Colocat... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
thanks for your response how can i create and execute this cron 2010/6/18 Danny Nicholas da...@debsinc.com I do a cron to execute “/usr/sbin/asterisk –rx “restart when convenient” “ each day at 4:45 AM. This doesn’t really “solve” any problems, just does “housekeeping” to keep a clean environment, since some installs/os’es lend themselves to memory leaks. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Friday, June 18, 2010 5:13 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 01:42 PM, Zeeshan Zakaria wrote: Based on my somewhat similar experience a few times, when this happens, try to resolve an IP, to make sure there is a valid DNS server accessible to Asterisk. If not, either make asterisk a DNS as well, or remove any domain name entries from /etc/resolv file and replace them with the IP addresses of your DNS. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-06-18 1:29 PM, sean darcy seandar...@gmail.com mailto:seandar...@gmail.com wrote: On 06/18/2010 12:57 PM, Tim Nelson wrote: - sean darcyseandar...@gmail.com mailto:seandar...@gmail.com wrote: We h... Yes, it is a typo. The network is 10.10.10.0/255.255.255.0 http://10.10.10.0/255.255.255.0. sean If the internet server is down, there can't be a valid DNS server accessible to Asterisk. The asterisk server is a caching name server, but obviously won't be able to resolve addresses not in its cache. Asterisk clearly doesn't need to resolve addresses to connect calls internally or over the T1. Is there any way to turn off its requirement for a DNS server? Or at least not fail catastrophically? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
Crontab -e will open your crontab for editing (if you are root) Add this line 45 4 * * * /usr/sbin/asterisk -rx restart when convenient And exit the editor This will restart your asterisk at 4:45 am every day unless a call is active at that time. If a call is active, asterisk will restart when the call hangs up. If you want to Damn the torpedoes, change when convenient to now. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Friday, June 18, 2010 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk issue thanks for your response how can i create and execute this cron 2010/6/18 Danny Nicholas da...@debsinc.com I do a cron to execute /usr/sbin/asterisk -rx restart when convenient each day at 4:45 AM. This doesn't really solve any problems, just does housekeeping to keep a clean environment, since some installs/os'es lend themselves to memory leaks. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Friday, June 18, 2010 5:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart asterisk service asterisk restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
Nice and colorful tutorial for cronjobs. http://www.linuxconfig.org/Linux_Cron_Guide -Bruce On Fri, Jun 18, 2010 at 1:55 PM, salaheddine elharit salah.elharit...@gmail.com wrote: thanks for your response how can i create and execute this cron 2010/6/18 Danny Nicholas da...@debsinc.com I do a cron to execute “/usr/sbin/asterisk –rx “restart when convenient” “ each day at 4:45 AM. This doesn’t really “solve” any problems, just does “housekeeping” to keep a clean environment, since some installs/os’es lend themselves to memory leaks. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Friday, June 18, 2010 5:13 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
Un-top-posting... On Fri, 18 Jun 2010, salaheddine elharit wrote: I have a problem in Asterisk 1.4 each day I need to restart asterisk service asterisk restart in order to unblock the calls 2010/6/18 Danny Nicholas da...@debsinc.com I do a cron to execute “/usr/sbin/asterisk –rx “restart when convenient” each day at 4:45 AM. This doesn’t really “solve” any problems, just does “housekeeping” to keep a clean environment, since some installs/os’es lend themselves to memory leaks. On Fri, 18 Jun 2010, salaheddine elharit wrote: how can i create and execute this cron Restarting Asterisk daily is a band-aid. Band-aids have their place, but to carry the analogy further, not if they allow the infection to fester. If you have an issue with whatever you mean by a blocked call, you should resolve the issue. Either it is something you (or your provider) are doing wrong (which will come back to bite you later) or you have identified a bug in Asterisk (unlikely, but we would all benefit from it's resolution). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
On Fri, 18 Jun 2010, bruce bruce wrote: Nice and colorful tutorial for cronjobs. http://www.linuxconfig.org/Linux_Cron_Guide Colorful, but missing valuable content like: setting environment variables, especially MAILTO and PATH; and time specification nicknames like @daily. man 5 crontab is also a good resource. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
Did you check /etc/resolv? Does it point to any DNS by domain name? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-18 2:04 PM, sean darcy seandar...@gmail.com wrote: On 06/18/2010 01:42 PM, Zeeshan Zakaria wrote: Based on my somewhat similar experience a few times... www.ilovetovoip.com http://www.ilovetovoip.com On 2010-06-18 1:29 PM, sean darcy seandar...@gmail.com mailto:seandar...@gmail.com wrote: On 06/18/2010 12:57 PM, Tim Nelson wrote: - ... mailto:seandar...@gmail.com wrote: We h... Yes, it is a typo. The network is 10.10... http://10.10.10.0/255.255.255.0. sean If the internet server is down, there can't be a valid DNS server accessible to Asterisk. The asterisk server is a caching name server, but obviously won't be able to resolve addresses not in its cache. Asterisk clearly doesn't need to resolve addresses to connect calls internally or over the T1. Is there any way to turn off its requirement for a DNS server? Or at least not fail catastrophically? sean -- _ -- Bandwidth and C... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk issue
thank you so much for your help and support :) 2010/6/18 Danny Nicholas da...@debsinc.com Crontab –e will open your crontab for editing (if you are root) Add this line 45 4 * * * /usr/sbin/asterisk –rx “restart when convenient” And exit the editor This will restart your asterisk at 4:45 am every day unless a call is active at that time. If a call is active, asterisk will restart when the call hangs up. If you want to “Damn the torpedoes”, change “when convenient” to “now”. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Friday, June 18, 2010 12:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] asterisk issue thanks for your response how can i create and execute this cron 2010/6/18 Danny Nicholas da...@debsinc.com I do a cron to execute “/usr/sbin/asterisk –rx “restart when convenient” “ each day at 4:45 AM. This doesn’t really “solve” any problems, just does “housekeeping” to keep a clean environment, since some installs/os’es lend themselves to memory leaks. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Friday, June 18, 2010 5:13 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] asterisk issue Hello, I have a problem in Asterisk 1.4 each day I need to restart *asterisk service asterisk* restart in order to unblock the calls My question how can I do in order to check the issue, and if there is any tool or log? Thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
Abandoning all hope of un-top-posting... On Fri, 18 Jun 2010, sean darcy wrote: (Sean has a problem and several posters suspect it is DNS related.) On Fri, 18 Jun 2010, Zeeshan Zakaria wrote: Did you check /etc/resolv? Does it point to any DNS by domain name? If you mean /etc/resolv.conf and the nameserver option, an IP address is required -- otherwise all attempts to use the resolver library fail. Have you tried running tcpdump -i [eth0|eth1|lo] port domain to see if it is a DNS query (and what the query is for) that is the issue? Have you tried entering the host names and IP addresses in /etc/hosts? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000
All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help. Also, I would like for the user to be able to set up their own password. I set the initial password the same as the extension, so that forces voicemail to prompt for a new password. The problem is, I can see where asterisk is trying to write the password in the voicemail.conf file, but it is denied because the user doesn't have permission. I hate to open /etc/asterisk directory to the incorrect permissions. What would be the best way to enable the user to be able to change their password? Thanks, Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hi again Thank you Warren, GotoIfTime was the deal! And easy to use! Gr8. Best regards. Aksel Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Warren Selby Sendt: 18. juni 2010 16:50 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*' On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.nomailto:ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Thank you for the info. As I wrote to Warren GotoIfTime was easy to use and seemed more flexible, Got it working now! Perfect! Only one thing left now, and my system is pretty much ready for live testing, Surely easy for the user list, so it will come in another mail soon, after I have done Some more research. (how the receptionist can transfer calls to SIP extensions internally) Best regards Aksel -Opprinnelig melding- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tilghman Lesher Sendt: 18. juni 2010 18:01 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*' On Friday 18 June 2010 09:49:39 Warren Selby wrote: On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote: Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|* Correct now. This isn't how you do time based checks in asterisk. Lookup the application GotoIfTime. Actually, it is an old method that still works, but as Warren mentioned, you should endeavor to switch to using GotoIfTime, as there's a nasty race condition inherent in using timed includes. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 03:09 PM, Steve Edwards wrote: Abandoning all hope of un-top-posting... On Fri, 18 Jun 2010, sean darcy wrote: (Sean has a problem and several posters suspect it is DNS related.) On Fri, 18 Jun 2010, Zeeshan Zakaria wrote: Did you check /etc/resolv? Does it point to any DNS by domain name? If you mean /etc/resolv.conf and the nameserver option, an IP address is required -- otherwise all attempts to use the resolver library fail. I'm running named as a caching nameserver. /etc/resolv.conf point to localhost. But, obviously, it only responds from the cache, since the root servers are unavailable. Have you tried running tcpdump -i [eth0|eth1|lo] port domain to see if it is a DNS query (and what the query is for) that is the issue? Have you tried entering the host names and IP addresses in /etc/hosts? tcpdump is an interesting idea. The only trouble is that I'm not sure when to run it. The phones don't go dead immediately. And not sure what to sort on if I do run it. Still puzzled about why asterisk does an address lookup, and why it unregisters sip phones with a hard ip address if it fails. sean A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.33 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.33. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.33 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Remove arbitrary size limitation for hints (Closes issue #17257. Reported, patched by tim_ringenbach) * Fix incorrectly typed indications for [nz] stutter and dialrecall (Closes issue #17359. Reported, patched by alecdavis) * Make AgentComplete message more consistent (Closes issue #15638. Reported, patched by elbriga) * Missing fallback to audio fax feature when T.38 re-INVITE failed (Closes issue #16692. Reported, patched by vrban) * Don't hang up on a queue caller if the file we attempt to play does not exist (Closes issue #17061. Reported by RoadKill) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.9 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix the PickupChan() application (Closes issue #16863. Reported, patched by schern. Patched by cjacobsen. Tested by Graber, cjacobsen, lathama, rickead2000, dvossel) * Improve logging by displaying line number (Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by dant, pabelanger, lmadsen) * Notify CLI when modules are loaded/unloaded (Closes issue #17308. Reported, patched by pabelanger. Tested by russell) * Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems (Closes issue #17028. Reported by pabelanger. Patched by seanbright, tilghman. Tested by pabelanger) * Manager cookies are not compatible with RFC2109. Make that no longer true. (Closes issue #17231. Reported, patched by ecarruda) * With IMAP backend, messages in INBOX were counted twice for MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.org/r/622/) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On Fri, 2010-06-18 at 13:59 -0400, sean darcy wrote: If the internet server is down, there can't be a valid DNS server accessible to Asterisk. The asterisk server is a caching name server, but obviously won't be able to resolve addresses not in its cache. Asterisk clearly doesn't need to resolve addresses to connect calls internally or over the T1. Is there any way to turn off its requirement for a DNS server? Or at least not fail catastrophically? Even when your connection to Internet is lost, doesn't mean there can not be a valid dns server. I would recommend to run bind on your asterisk machine to resolve all addresses that your asterisk server and/or phones need. If that machine goes down, your phones can not call anyway ;-) Same for doing DHCP for handing out addresses to your phones... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.9 Now Available
Hello everyone. Successfully patched to the new version, but when trying to compile, I get this : /usr/src/asterisk/asterisk/include/asterisk/options.h:102:56: error: operator '' has no right operand Dahdi is fresh from the SVN trunk. Am I missing something ? Thanks ! Hoggins! Le 18/06/2010 23:03, Asterisk Development Team a écrit : The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix the PickupChan() application (Closes issue #16863. Reported, patched by schern. Patched by cjacobsen. Tested by Graber, cjacobsen, lathama, rickead2000, dvossel) * Improve logging by displaying line number (Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by dant, pabelanger, lmadsen) * Notify CLI when modules are loaded/unloaded (Closes issue #17308. Reported, patched by pabelanger. Tested by russell) * Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems (Closes issue #17028. Reported by pabelanger. Patched by seanbright, tilghman. Tested by pabelanger) * Manager cookies are not compatible with RFC2109. Make that no longer true. (Closes issue #17231. Reported, patched by ecarruda) * With IMAP backend, messages in INBOX were counted twice for MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.org/r/622/) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 05:08 PM, Hans Witvliet wrote: On Fri, 2010-06-18 at 13:59 -0400, sean darcy wrote: If the internet server is down, there can't be a valid DNS server accessible to Asterisk. The asterisk server is a caching name server, but obviously won't be able to resolve addresses not in its cache. Asterisk clearly doesn't need to resolve addresses to connect calls internally or over the T1. Is there any way to turn off its requirement for a DNS server? Or at least not fail catastrophically? Even when your connection to Internet is lost, doesn't mean there can not be a valid dns server. I would recommend to run bind on your asterisk machine to resolve all addresses that your asterisk server and/or phones need. I do run bind on the asterisk machine. If that machine goes down, your phones can not call anyway ;-) But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? The internal phones can't even call each other, even though they have hard ip addresses. Same for doing DHCP for handing out addresses to your phones... All the phones have manual ip addresses. No DHCP. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On Fri, 18 Jun 2010, sean darcy wrote: I'm running named as a caching nameserver. /etc/resolv.conf point to localhost. But, obviously, it only responds from the cache, since the root servers are unavailable. If the root servers are not available, what is available to cache? tcpdump is an interesting idea. The only trouble is that I'm not sure when to run it. The phones don't go dead immediately. And not sure what to sort on if I do run it. How about this: sudo -b tcpdump -i eth0 port domain eth0-domain sudo -b tcpdump -i lo port domain lo-domain and then look at *-domain when you have an issue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? The internal phones can't even call each other, even though they have hard ip addresses. Same for doing DHCP for handing out addresses to your phones... All the phones have manual ip addresses. No DHCP. sean Do the phones find the sip server by IP or by domain name. I.e. 1.1.1.1 Or sip.yourdomain.com If domain name, what are they using for DNS? Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 06/18/2010 06:19 PM, Cary Fitch wrote: But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? The internal phones can't even call each other, even though they have hard ip addresses. Same for doing DHCP for handing out addresses to your phones... All the phones have manual ip addresses. No DHCP. sean Do the phones find the sip server by IP or by domain name. I.e. 1.1.1.1 Or sip.yourdomain.com If domain name, what are they using for DNS? Cary Fitch The sip proxy server and the sip registrar server are both set to 10.10.10.180. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
Hi! But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? Probably because you have one or more register = statements in your sip.conf and Asterisk is trying badly - but without success - to register itself to one or more of your providers. This then happens to block all incoming LAN SIP traffic. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
On 6/18/2010 7:26 PM, sean darcy wrote: On 06/18/2010 06:19 PM, Cary Fitch wrote: But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? The internal phones can't even call each other, even though they have hard ip addresses. Same for doing DHCP for handing out addresses to your phones... All the phones have manual ip addresses. No DHCP. sean Do the phones find the sip server by IP or by domain name. I.e. 1.1.1.1 Or sip.yourdomain.com If domain name, what are they using for DNS? Cary Fitch The sip proxy server and the sip registrar server are both set to 10.10.10.180. sean Our company has set up hundreds of asterisk boxes over the years. One thing we learned early on was to avoid any type of DNS resolution by asterisk. Asterisk gets hung when it can't access your DNS server and all things grind to a halt (ie, phones can't register). The two things we make sure to do on any new installation is to: 1) use only IP addresses in all the config files. 2) set srvlookup = no in sip.conf. Try those 2 things out and then remove the internet connection from your server. Check and see if call processing works normally. If it doesn't, do a tcpdump or ngrep capture to see what DNS queries are being done and figure out why. Andres http://www.neuroredes.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi modules installed wrong location
CENTOS 5.5 dahdi 1.4.3.0.1 uname -r 2.6.18-194.3.1.el5PAE [r...@localhost dahdi-linux-2.3.0.1]# service dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. wct4xxp: FATAL: Module wct4xxp not found. [FAILED] wctc4xxp: FATAL: Module wctc4xxp not found. [FAILED] Error: missing /dev/dahdi! [r...@localhost dahdi-linux-2.3.0.1]# modprobe dahdi FATAL: Module dahdi not found. Asterisk starts OK After doing make then make install, dahdi creates /lib/modules/2.6.18-194.3.1.el5/dahdi with the modules in it. but the correct location all ready exist /lib/modules/2.6.18-194.3.1.el5PAE. When I delete /lib/modules/2.6.18-194.3.1.el5 and do a make clean, make, make install the /lib/modules/2.6.18-194.3.1.el5 is recreated with the dahdi dir in it. There was one error message doing the make: WARNING: could not find /usr/src/dahdi-linux-2.3.0.1/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/dahdi-linux-2.3.0.1/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Don't know if this is a problem. If I remove the link to the kernel source I get this doing make: You do not appear to have the sources for the 2.6.18-194.3.1.el5PAE kernel installed. thanks for the help Steve Casto -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James SIP trace: U external.ip:9375 - asterisk.ip:5060 NOTIFY sip:asterisk.ip SIP/2.0..Via: SIP/2.0/UDP 10.10.30.65:9375;branch=z9hG4bK-8ebce8bc..From: xxx-xxx- sip:9497197...@asterisk.ip;tag=3a6a735864619b8bo0..To: sip:asterisk.ip..Call-ID: 19a0bd7 c-3cb13...@10.10.30.65..cseq: 395 NOTIFY..Max-Forwards: 70..Contact: xxx-xxx- sip:xxx...@10.10.30.65:9375..Event: keep-alive..User-Agent: Linksys/SPA942-6.1.3(a)-000e08d87445..Content-Length: 0 # U asterisk.ip:5060 - 10.10.30.65:9375 SIP/2.0 489 Bad event..Via: SIP/2.0/UDP 10.10.30.65:9375;branch=z9hG4bK-8ebce8bc;received=external.ip..From: xxx-xxx- sip:9497197...@asterisk.ip;tag=3a6a735864619b8bo0..To: sip:asterisk.ip;tag=as4a 4466b0..Call-ID: 19a0bd7c-3cb13...@10.10.30.65..cseq: 395 NOTIFY..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}
Thank you.
another quesion is i want to get ${CDR(answer)} and ${CDR(end)} in the
hangup section. i can get ${CDR(answer)} sucessfully but get
${CDR(end)} is null.
i know i can set any variable i want into CDR table if i want .
but i want to know without any setting . which variabes will set automatic?
is that all of the following(i use asterisk 1.6.2.1)?
${CDR(clid)} Caller ID
${CDR(src)} Source
${CDR(dst)} Destination
${CDR(dcontext)} Destination context
${CDR(channel)} Channel name
${CDR(dstchannel)} Destination channel
${CDR(lastapp)} Last app executed
${CDR(lastdata)} Last app's arguments
${CDR(start)} Time the call started.
${CDR(answer)} Time the call was answered.
${CDR(end)} Time the call ended.
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
${CDR(accountcode)} The channel's account code (read-write).
${CDR(uniqueid)} The channel's unique id.
${CDR(userfield)} The channels uses specified field (read-write).
2010/6/18 Tilghman Lesher tles...@digium.com:
On Friday 18 June 2010 03:21:32 Zhang Shukun wrote:
hi,all
for a long time, i cant understand the difference between
${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}
i know ${CDR(start)} mean when a call is start. and ${CDR(answer)}
means when a call was pick up.
but what's ${CDR(calldate)} mean?
It could mean whatever you want. CDRs (at least the internal representation)
have support for arbitrary additional variables. Whether a particular backend
has support to carry those over into permanent storage is another question
(in 1.6.2, most CDR backends have it, as long as the underlying table has a
column to receive the data).
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks for your supporting,
have a nice day.
Sucan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can sip clients connect with each other directly (RTP session) ?
Dear Asterisk friends, Please help me to clarify my doubt. After monitor SIP and RTP traffic with Wireshark. I found that both SIP and RTP traffic between 2 sip clients must be passed through Asterisk. Is it possible that 2 sip clients connect with each other directly for RTP session after sip session completed ? Thank you, Kamonwat -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can sip clients connect with each other directly (RTP session) ?
On 06/19/10 15:19, Kamonwat Sookkara wrote: Dear Asterisk friends, Please help me to clarify my doubt. After monitor SIP and RTP traffic with Wireshark. I found that both SIP and RTP traffic between 2 sip clients must be passed through Asterisk. Is it possible that 2 sip clients connect with each other directly for RTP session after sip session completed ? By default it is yes, however within a LAN environment you can usually allow clients to re-invite directly between themselves. Check the canreinvite option out.// -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James Hello james, in the SPA config webpage on EXT 1 in the Nat Keep Alive MSG you should set $OPTIONS instead of $NOTIFY. then in your asterisk extension default context just set this: exten = s,1,Hangup then the phone will send a options packet and you will get a 200 OK instead of 489 Bad event. this should help. best regards steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
