Re: [asterisk-users] Notify me when the call is answered
2011/3/17 Eric Smith e...@fruitcom.com Hi I want to have some signal when a call is answered. I can watch the asterisk debug or logs and see when a call is answered of course but I want a sound notification. I would try using Dial M or U options with which a macro or routine is run when the call is answered. Cheers -- _ -- 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] Problem routing call to fax machine on DAHDI FXS port
2011/3/18 Frank Tarczynski ft...@mindspring.com
I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS
modules. I'm trying to set-up things to route analog fax calls from a FXO
port to an analog fax machine on a FXS port on the same card.
Outgoing faxes work just fine. But incoming faces are routed to the right
DAHDI extension, but the call dropped right as the fax machine rings for
the first time. The fax machine works fine when connected directly to the
analog telephone line and I see the same behavior if I route the fax call to
anyother DAHDI or SIP extension.
Can anyone help?
I see this in the asterisk log:
(Send fax out to HP's fax check line)
[2011-03-17 13:40:17.4] VERBOSE[8825] chan_dahdi.c: -- Starting simple
switch on 'DAHDI/1-1'
[2011-03-17 13:40:24.0] VERBOSE[8825] pbx.c: -- Executing
[18884732963@from-fax-machine:1] Set(DAHDI/1-1,
CALLERID(num)=19195718465) in new stack
[2011-03-17 13:40:24.0] VERBOSE[8825] pbx.c: -- Executing
[18884732963@from-fax-machine:2] Dial(DAHDI/1-1, DAHDI/4/18884732963)
in new stack
[2011-03-17 13:40:24.0] VERBOSE[8825] app_dial.c: -- Called
4/18884732963
[2011-03-17 13:40:26.2] VERBOSE[8825] app_dial.c: -- DAHDI/4-1 answered
DAHDI/1-1
[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax
extension
[2011-03-17 13:41:13.4] VERBOSE[8825] chan_dahdi.c: -- Hungup
'DAHDI/4-1'
[2011-03-17 13:41:13.4] VERBOSE[8825] pbx.c: == Spawn extension
(from-fax-machine, 18884732963, 2) exited non-zero on 'DAHDI/1-1'
[2011-03-17 13:41:13.4] VERBOSE[8825] chan_dahdi.c: -- Hungup
'DAHDI/1-1'
(Incoming fax attempt)
[2011-03-17 13:43:18.3] VERBOSE[8834] chan_dahdi.c: -- Starting simple
switch on 'DAHDI/4-1'
[2011-03-17 13:43:19.3] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:1] Wait(DAHDI/4-1, 1) in new stack
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:2] Verbose(DAHDI/4-1, CALLERID is 8884732963) in new
stack
[2011-03-17 13:43:20.4] VERBOSE[8834] app_verbose.c: CALLERID is 8884732963
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:3] Verbose(DAHDI/4-1, Time is 20110317-134320) in new
stack
[2011-03-17 13:43:20.4] VERBOSE[8834] app_verbose.c: Time is
20110317-134320
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:4] Answer(DAHDI/4-1, ) in new stack
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:5] Ringing(DAHDI/4-1, ) in new stack
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:6] Wait(DAHDI/4-1, 6) in new stack
[2011-03-17 13:43:21.4] VERBOSE[8834] chan_dahdi.c: -- Redirecting
DAHDI/4-1 to fax extension
[2011-03-17 13:43:21.4] VERBOSE[8834] pbx.c: == Spawn extension
(from-pstn-4, fax, 1) exited non-zero on 'DAHDI/4-1'
[2011-03-17 13:43:21.4] VERBOSE[8834] pbx.c: -- Executing
[fax@from-pstn-4:1] NoOp(DAHDI/4-1, Fax Detected) in new stack
[2011-03-17 13:43:21.4] VERBOSE[8834] pbx.c: -- Executing
[fax@from-pstn-4:2] Dial(DAHDI/4-1, DAHDI/1,40,tr) in new stack
[2011-03-17 13:43:21.4] VERBOSE[8834] app_dial.c: -- Called 1
[2011-03-17 13:43:21.4] VERBOSE[8834] app_dial.c: -- DAHDI/1-1 is
ringing
[2011-03-17 13:43:23.4] VERBOSE[8834] app_dial.c: -- DAHDI/1-1 is
ringing
(Call is routed to fax machine, but then dropped before it can answer)
[2011-03-17 13:43:24.8] VERBOSE[8834] chan_dahdi.c: -- Hungup
'DAHDI/1-1'
[2011-03-17 13:43:24.8] VERBOSE[8834] pbx.c: == Spawn extension
(from-pstn-4, fax, 2) exited non-zero on 'DAHDI/4-1'
[2011-03-17 13:43:24.8] VERBOSE[8834] chan_dahdi.c: -- Hungup
'DAHDI/4-1'
My dialplan looks like this:
[from-pstn-4]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,Dial(DAHDI/1,,rtT)
exten = fax,3,Congestion()
exten = fax,104,Busy()
exten = s,1,Wait(1)
exten = s,n,Verbose(CALLERID is ${CALLERID(num)})
exten = s,n,Verbose(Time is ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,n,Answer
exten = s,n,Ringing
exten = s,n,Wait(6)
exten = s,n,Dial(SIP/1000)
exten = s,n,Voicemail(1000,u)
exten = s,n,Hangup
My chan_dahdi.conf file looks like:
[trunkgroups]
;trunkgroup = 1,1
;trunkgroup = 2,2
;trunkgroup = 3,3
;trunkgroup = 4,4
;spanmap = 1,1
;spanmap = 2,2
;spanmap = 3,3
;spanmap = 4,4
[channels]
language=en
context=incoming
toneduration=40
;usedistinctiveringdetection=yes
answeronpolarityswitch=no
usecallerid=yes
cidsignalling=bell
cidstart=ring
;hidecallerid=yes
;hidecalleridname=yes
;waitfordialtone=yes
;mwimonitor=no
;mwilevel=512
;mwimonitornotify=/usr/local/bin/dahdinotify.sh
;mwisendtype=rpas,lrev
callwaiting=yes
;restrictcid=no
usecallingpres=yes
sendcalleridafter = 1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=2.0
txgain=8.0
group=1
callgroup=1
Re: [asterisk-users] Status of Queue Members
Probably this will help you... http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901 Check the section 'Controlling when to join and leave a queue'. --AM On Thu, Mar 17, 2011 at 9:15 PM, Dan Journo d...@keshercommunications.comwrote: Hi, I'm trying to work out an issue with call queues. I need the calls that are in a queue to be kicked out if all members are unavailable (for example if all SIP members are having network problems). I tried leavewhenempty = yes but that only seems works when all queue members specifically log out of a queue. I've looked at autopause, but we need it to automatically un-pause once it comes back online. Any idea how I can do this? Preferably without using the AMI or AGI scripts, but if that's the only way, then i'll have to use that. Thanks Dan Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.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 -- _ -- 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] Status of Queue Members
Probably this will help you... http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901 Check the section 'Controlling when to join and leave a queue'. Thanks. Thats perfect! Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.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
[asterisk-users] DISA DTMF problem
Hello, Im tryng to setup DISA on my server. Outlines comes via VoIP to my asterisk server. When i dial from outside to my disa number it answers. I dial the extension that i want to dial but Dial tone keeps up playing about 3-4 seconds more even i start to enter numbers..Then when timeout occurs, it dials the number. What is strange is even if detects the dialed number in a mysterious way, if i add P parameter to DISA to wait for #. It keeps on waiting even if i press # several times. Asterisk version is:1.6.2.9-2 Thanks -- _ -- 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] [1.4] Asterisk doesn't hang up?
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr wrote: I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: For those trying to do the same thing: Zaptel/Dahdi does detect that the remote party has hung up when using busydetect=yes in zapata.conf/chan_dahdi.conf. -- _ -- 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] [1.4] Failed callfile doesn't jump to failedextension
On Tue, 15 Mar 2011 11:44:20 -0500, Danny Nicholas
da...@debsinc.com wrote:
Don't depend on the tutorials you read to be 100% accurate or up-to-date.
The default action on a failure in Asterisk is usually going to be an s
jump, either to s,1 or s+100. Personally, I would replace failed,1 with
start-NOANSWER,1.
Thanks for the info. After calling out through a call file, Asterisk
plays the MOH and detects that the callee has hung up, but either
doesn't jump to the extension or does jump to h but ${REASON} is
empty:
===
[callback]
;how to wait until callee has answered?
exten = start,1,Wait(2)
exten = start,n,NoOp(${DEVICE_STATE(Dahdi/1)})
exten = start,n,Answer()
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended with ${REASON})
;empty
;exten = h,1,NoOp(Call ended with ${REASON})
;not run
exten = start-NOANSWER,1,NoOp(Call ended with ${REASON})
===
Is this what you had in mind?
Thank you.
--
_
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Re: [asterisk-users] [1.4] Failed callfile doesn't jump tofailedextension
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, March 18, 2011 10:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Failed callfile doesn't jump
tofailedextension
On Tue, 15 Mar 2011 11:44:20 -0500, Danny Nicholas
da...@debsinc.com wrote:
Don't depend on the tutorials you read to be 100% accurate or up-to-date.
The default action on a failure in Asterisk is usually going to be an s
jump, either to s,1 or s+100. Personally, I would replace failed,1 with
start-NOANSWER,1.
Thanks for the info. After calling out through a call file, Asterisk
plays the MOH and detects that the callee has hung up, but either
doesn't jump to the extension or does jump to h but ${REASON} is
empty:
===
[callback]
;how to wait until callee has answered?
exten = start,1,Wait(2)
exten = start,n,NoOp(${DEVICE_STATE(Dahdi/1)})
exten = start,n,Answer()
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended with ${REASON})
;empty
;exten = h,1,NoOp(Call ended with ${REASON})
;not run
exten = start-NOANSWER,1,NoOp(Call ended with ${REASON})
===
Is this what you had in mind?
Thank you.
That's the ticket.
--
_
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Re: [asterisk-users] [1.4] Failed callfile doesn't jump tofailedextension
On Fri, 18 Mar 2011 10:14:37 -0500, Danny Nicholas
da...@debsinc.com wrote:
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended with ${REASON})
;empty
;exten = h,1,NoOp(Call ended with ${REASON})
;not run
exten = start-NOANSWER,1,NoOp(Call ended with ${REASON})
===
Is this what you had in mind?
Thank you.
That's the ticket.
Unfortunately, it can only jump to h, and ${REASON} is empty.
Based on...
www.voip-info.org/wiki/view/Asterisk+dial+plan+-+working+example
... I also tried this, but Asterisk doesn't jump to any of those
extensions:
= extensions.conf
...
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
;exten = start,n,Goto(${EXTEN}-${REASON})
exten = start,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy ;Only works with SIP calls
exten = s-CHANUNAVAIL,1,Verbose(Not available)
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
= CLI
-- Executing [start@callback:5] Playback(DAHDI/1-1,
manolo_camp-morning_coffee) in new stack
-- DAHDI/1-1 Playing 'manolo_camp-morning_coffee.ulaw' (language
'fr')
== Spawn extension (callback, start, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
[Mar 18 16:41:35] NOTICE[1200]: pbx_spool.c:349 attempt_thread: Call
completed to Dahdi/1/5551234
=
Is there no way to know how a call ended?
Thank you.
--
_
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, March 18, 2011 10:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Failed callfile doesn't
jumptofailedextension
On Fri, 18 Mar 2011 10:14:37 -0500, Danny Nicholas
da...@debsinc.com wrote:
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended with ${REASON})
;empty
;exten = h,1,NoOp(Call ended with ${REASON})
;not run
exten = start-NOANSWER,1,NoOp(Call ended with ${REASON})
===
Is this what you had in mind?
Thank you.
That's the ticket.
Unfortunately, it can only jump to h, and ${REASON} is empty.
Based on...
www.voip-info.org/wiki/view/Asterisk+dial+plan+-+working+example
... I also tried this, but Asterisk doesn't jump to any of those
extensions:
= extensions.conf
...
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
;exten = start,n,Goto(${EXTEN}-${REASON})
exten = start,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy ;Only works with SIP calls
exten = s-CHANUNAVAIL,1,Verbose(Not available)
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
= CLI
-- Executing [start@callback:5] Playback(DAHDI/1-1,
manolo_camp-morning_coffee) in new stack
-- DAHDI/1-1 Playing 'manolo_camp-morning_coffee.ulaw' (language
'fr')
== Spawn extension (callback, start, 5) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
[Mar 18 16:41:35] NOTICE[1200]: pbx_spool.c:349 attempt_thread: Call
completed to Dahdi/1/5551234
=
Is there no way to know how a call ended?
Thank you.
I believe you will achieve the desired result by replacing ${REASON} with
${HANGUP_CAUSE}.
--
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Behalf Of Gilles
Unfortunately, it can only jump to h, and ${REASON} is empty.
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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[asterisk-users] One PRI card with 2 (or more) Telcos
Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate. Problem clears immediately tho. I send a copy of the log with pri debug at a time of problems... Is there a problem having 2 telcos on the same PRI card? Would somebody help? asterisk*CLI pri show span 7 Primary D-channel: 202 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No and [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (I): T200 expired N200 times sending RR/RNR in state 8(Timer recovery) [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: Changing from state 8(Timer recovery) to 5(Awaiting establishment) [Mar 18 17:04:06] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:07] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:08] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): *T200 expired N200 times sending SABME in state 5(Awaiting establishment)* [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 5(Awaiting establishment) to 4(TEI assigned) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=56 on channel 2 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=64 on channel 3 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=58 on channel 4 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=66 on channel 6 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI assigned) to 5(Awaiting establishment) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: == Primary D-Channel on span 7 down [Mar 18 17:04:09] WARNING[19844] chan_dahdi.c: No D-channels available! Using Primary channel 202 as D-channel anyway! [Mar 18 17:04:10] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:11] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:12] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 5(Awaiting establishment) to 4(TEI assigned) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI assigned) to 5(Awaiting establishment) [Mar 18 17:04:13] WARNING[19844] chan_dahdi.c: No D-channels available! Using Primary channel 202 as D-channel anyway! [Mar 18 17:04:14] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for data link re-establishment [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750 pri_internal_clear: Call 56 enters state 0 (Null). Hold state: Idle [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: clearing, alive 1, hangupack 0 [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: == Primary D-Channel on span 7 up [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: -- Channel 0/2, span 7 got hangup, cause 27 [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for data link re-establishment [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750 pri_internal_clear: Call 64 enters state 0 (Null). Hold state: Idle [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: clearing, alive 1, hangupack 0 [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: -- Channel 0/3, span 7 got hangup, cause 27 [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for data link re-establishment [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750 pri_internal_clear: Call 58 enters state 0 (Null). Hold state: Idle [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: clearing, alive 1, hangupack 0 [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: -- Channel 0/4, span 7 got hangup, cause 27 [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for data link re-establishment [Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750 pri_internal_clear: Call 66 enters state 0 (Null). Hold
Re: [asterisk-users] One PRI card with 2 (or more) Telcos
Just a follow up with a bit more information asterisk*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 *res_timing_dahdi.soDAHDI Timing Interface 40* 2 modules loaded asterisk*CLI -- [root@asterisk ~]# dahdi_test -c 100 Opened pseudo dahdi interface, measuring accuracy... 99.999% 99.999% 99.992% 99.997% 99.998% 99.995% 99.998% 99.996% 99.997% 99.998% 99.997% 99.994% 99.991% 99.999% 99.998% 99.998% 99.995% 99.993% 99.998% 99.999% 99.998% 99.995% 99.992% 99.998% 100.000% 99.998% 99.995% 99.992% 99.999% 99.998% 99.998% 99.999% 99.995% 99.999% 99.999% 99.998% 99.999% 99.997% 99.999% 99.998% 99.998% 99.996% 99.992% 99.998% 99.998% 99.999% 99.996% 99.992% 99.999% 99.998% 99.997% 99.997% 99.997% 99.998% 99.995% 99.994% 99.995% 99.992% 99.999% 99.993% 99.990% 99.995% 99.993% 99.999% 99.997% 99.993% 99.999% 99.996% 99.998% 99.996% 99.993% 99.995% 99.992% 99.998% 99.993% 99.993% 99.999% 99.993% 99.998% 99.996% 99.993% 99.996% 99.996% 99.994% 99.999% 99.996% 99.996% 99.992% 99.999% 99.996% 99.991% 99.996% 99.992% 99.998% 99.997% 99.994% 99.998% 99.995% --- Results after 98 passes --- Best: 100.000 -- Worst: 99.990 -- Average: 99.996163, Difference: 99.998235 -- [root@asterisk ~]# cat /sys/devices/system/clocksource/clocksource0/current_clocksource *tsc* [root@asterisk ~]# cat /sys/devices/system/clocksource/clocksource0/available_clocksource tsc hpet acpi_pm jiffies On 18 March 2011 17:52, Tiago Geada tiago.ge...@gmail.com wrote: Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate. Problem clears immediately tho. I send a copy of the log with pri debug at a time of problems... Is there a problem having 2 telcos on the same PRI card? Would somebody help? asterisk*CLI pri show span 7 Primary D-channel: 202 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No and [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (I): T200 expired N200 times sending RR/RNR in state 8(Timer recovery) [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: Changing from state 8(Timer recovery) to 5(Awaiting establishment) [Mar 18 17:04:06] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:07] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:08] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): *T200 expired N200 times sending SABME in state 5(Awaiting establishment)* [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 5(Awaiting establishment) to 4(TEI assigned) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=56 on channel 2 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=64 on channel 3 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=58 on channel 4 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=66 on channel 6 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI assigned) to 5(Awaiting establishment) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: == Primary D-Channel on span 7 down [Mar 18 17:04:09] WARNING[19844] chan_dahdi.c: No D-channels available! Using Primary channel 202 as D-channel anyway! [Mar 18 17:04:10] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:11] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:12] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 5(Awaiting establishment) to 4(TEI assigned) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI assigned) to 5(Awaiting
Re: [asterisk-users] One PRI card with 2 (or more) Telcos
Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian -- _ -- 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] One PRI card with 2 (or more) Telcos
On Fri, Mar 18, 2011 at 3:15 PM, Tiago Geada tiago.ge...@gmail.com wrote: Just a follow up with a bit more information asterisk*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.so DAHDI Timing Interface 40 2 modules loaded asterisk*CLI -- [root@asterisk ~]# dahdi_test -c 100 Opened pseudo dahdi interface, measuring accuracy... 99.999% 99.999% 99.992% 99.997% 99.998% 99.995% 99.998% 99.996% 99.997% 99.998% 99.997% 99.994% 99.991% 99.999% 99.998% 99.998% 99.995% 99.993% 99.998% 99.999% 99.998% 99.995% 99.992% 99.998% 100.000% 99.998% 99.995% 99.992% 99.999% 99.998% 99.998% 99.999% 99.995% 99.999% 99.999% 99.998% 99.999% 99.997% 99.999% 99.998% 99.998% 99.996% 99.992% 99.998% 99.998% 99.999% 99.996% 99.992% 99.999% 99.998% 99.997% 99.997% 99.997% 99.998% 99.995% 99.994% 99.995% 99.992% 99.999% 99.993% 99.990% 99.995% 99.993% 99.999% 99.997% 99.993% 99.999% 99.996% 99.998% 99.996% 99.993% 99.995% 99.992% 99.998% 99.993% 99.993% 99.999% 99.993% 99.998% 99.996% 99.993% 99.996% 99.996% 99.994% 99.999% 99.996% 99.996% 99.992% 99.999% 99.996% 99.991% 99.996% 99.992% 99.998% 99.997% 99.994% 99.998% 99.995% --- Results after 98 passes --- Best: 100.000 -- Worst: 99.990 -- Average: 99.996163, Difference: 99.998235 -- [root@asterisk ~]# cat /sys/devices/system/clocksource/clocksource0/current_clocksource tsc [root@asterisk ~]# cat /sys/devices/system/clocksource/clocksource0/available_clocksource tsc hpet acpi_pm jiffies On 18 March 2011 17:52, Tiago Geada tiago.ge...@gmail.com wrote: Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate. Problem clears immediately tho. I send a copy of the log with pri debug at a time of problems... Is there a problem having 2 telcos on the same PRI card? Would somebody help? asterisk*CLI pri show span 7 Primary D-channel: 202 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No and [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (I): T200 expired N200 times sending RR/RNR in state 8(Timer recovery) [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: Changing from state 8(Timer recovery) to 5(Awaiting establishment) [Mar 18 17:04:06] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:07] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:08] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 5(Awaiting establishment) to 4(TEI assigned) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=56 on channel 2 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=64 on channel 3 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=58 on channel 4 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=66 on channel 6 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI assigned) to 5(Awaiting establishment) [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: == Primary D-Channel on span 7 down [Mar 18 17:04:09] WARNING[19844] chan_dahdi.c: No D-channels available! Using Primary channel 202 as D-channel anyway! [Mar 18 17:04:10] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:11] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:12] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 5(Awaiting establishment) to 4(TEI assigned) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3) [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending
Re: [asterisk-users] One PRI card with 2 (or more) Telcos
On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini adrian-li...@wombit.com wrote: Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian Adrian This only works when all the providers are using a common clock like some areas in the USA. This is not the case all around the world. -- ~~~ Andrew lathama Latham lath...@gmail.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] One PRI card with 2 (or more) Telcos
Hi! I can try that tho. Where do I configure what timer to use??! Thanks in advance. On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote: On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini adrian-li...@wombit.com wrote: Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian Adrian This only works when all the providers are using a common clock like some areas in the USA. This is not the case all around the world. -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] One PRI card with 2 (or more) Telcos
OK I found it. In /etc/dahdi/system.conf I have for this span: # Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 HDB3/CCS/CRC4 span=7,7,0,ccs,hdb3,crc4 # termtype: te bchan=187-201,203-217 dchan=202 echocanceller=mg2,187-201,203-217 should I use span=7,*5*,0,ccs,hdb3,crc4 instead? (5 is the first telco on that card) On 18 March 2011 18:23, Tiago Geada tiago.ge...@gmail.com wrote: Hi! I can try that tho. Where do I configure what timer to use??! Thanks in advance. On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote: On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini adrian-li...@wombit.com wrote: Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian Adrian This only works when all the providers are using a common clock like some areas in the USA. This is not the case all around the world. -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] One PRI card with 2 (or more) Telcos
On 03/18/2011 01:23 PM, Tiago Geada wrote: Hi! I can try that tho. Where do I configure what timer to use??! If your telcos are not synchronizing their network clocks to each other, you will not be able to solve this problem on a multi-port Digium T1/E1 card. Digium T1/E1 cards select a single master clock (either the onboard clock or the clock recovered from one of the spans) to use as the 'board clock', which is then used to transmit data on all the spans. If the master clock is not in synchronization with the clocks at the other end of those spans, then bit slips will occur and cause various sorts of problems. This is why a card is always configured to use the recovered clock from a telco span if there is one, because the onboard clock would never by in sync with it. If you have a board connected to two telcos and their clocks are not synchronized, not only will you have trouble using a Digium card, but even using a card that can handle using multiple transmit clocks at once will not solve the underlying bit slip problem that will occur if you ever connect a channel from Telco1 to a channel from Telco2. If you *never* connect channels between Telcos, then you don't have to worry about that problem, but if you do, at some point during the call there will be buffer overruns or underruns and there will be some effect (for a normal voice call, the effect might be a short audio artifact, and fairly harmless... unless the call is a modem or FAX call, in which case it could cause the call to fail). For your sanity, I would strongly suggest that you don't connect spans from multiple telcos/networks/etc. on a single card, but keep each span provider on their own card. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] One PRI card with 2 (or more) Telcos
Sorry to keep bugging, but after making changes to /etc/dahdi/system.conf, do I need unload res_timing_dahdi.so and chan_dahdi.so; and load them, or can I just reload them?? Thanks in advance On 18 March 2011 18:26, Tiago Geada tiago.ge...@gmail.com wrote: OK I found it. In /etc/dahdi/system.conf I have for this span: # Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 HDB3/CCS/CRC4 span=7,7,0,ccs,hdb3,crc4 # termtype: te bchan=187-201,203-217 dchan=202 echocanceller=mg2,187-201,203-217 should I use span=7,*5*,0,ccs,hdb3,crc4 instead? (5 is the first telco on that card) On 18 March 2011 18:23, Tiago Geada tiago.ge...@gmail.com wrote: Hi! I can try that tho. Where do I configure what timer to use??! Thanks in advance. On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote: On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini adrian-li...@wombit.com wrote: Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian Adrian This only works when all the providers are using a common clock like some areas in the USA. This is not the case all around the world. -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] One PRI card with 2 (or more) Telcos
Probably overkill, but Every time I make a change to dahdi, I do this Service asterisk stop Service dadhi restart Service asterisk start _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Friday, March 18, 2011 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One PRI card with 2 (or more) Telcos Sorry to keep bugging, but after making changes to /etc/dahdi/system.conf, do I need unload res_timing_dahdi.so and chan_dahdi.so; and load them, or can I just reload them?? Thanks in advance On 18 March 2011 18:26, Tiago Geada tiago.ge...@gmail.com wrote: OK I found it. In /etc/dahdi/system.conf I have for this span: # Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 HDB3/CCS/CRC4 span=7,7,0,ccs,hdb3,crc4 # termtype: te bchan=187-201,203-217 dchan=202 echocanceller=mg2,187-201,203-217 should I use span=7,5,0,ccs,hdb3,crc4 instead? (5 is the first telco on that card) On 18 March 2011 18:23, Tiago Geada tiago.ge...@gmail.com wrote: Hi! I can try that tho. Where do I configure what timer to use??! Thanks in advance. On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote: On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini adrian-li...@wombit.com wrote: Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco. Then the other spans are secondary, tertiary, quaternary timers. Adrian Adrian This only works when all the providers are using a common clock like some areas in the USA. This is not the case all around the world. -- ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] One PRI card with 2 (or more) Telcos
Hi Kevin, Thanks for your elaborated answer. I will try and set them on the same clock and see if no problem occurs. If so, Different telco's clocks would be in SYNC (I do doubt it). This machine has no more PCI slots available and hardware is damn expensive. Will have to look into it with my boss.. Thanks you. On 18 March 2011 18:30, Kevin P. Fleming kpflem...@digium.com wrote: On 03/18/2011 01:23 PM, Tiago Geada wrote: Hi! I can try that tho. Where do I configure what timer to use??! If your telcos are not synchronizing their network clocks to each other, you will not be able to solve this problem on a multi-port Digium T1/E1 card. Digium T1/E1 cards select a single master clock (either the onboard clock or the clock recovered from one of the spans) to use as the 'board clock', which is then used to transmit data on all the spans. If the master clock is not in synchronization with the clocks at the other end of those spans, then bit slips will occur and cause various sorts of problems. This is why a card is always configured to use the recovered clock from a telco span if there is one, because the onboard clock would never by in sync with it. If you have a board connected to two telcos and their clocks are not synchronized, not only will you have trouble using a Digium card, but even using a card that can handle using multiple transmit clocks at once will not solve the underlying bit slip problem that will occur if you ever connect a channel from Telco1 to a channel from Telco2. If you *never* connect channels between Telcos, then you don't have to worry about that problem, but if you do, at some point during the call there will be buffer overruns or underruns and there will be some effect (for a normal voice call, the effect might be a short audio artifact, and fairly harmless... unless the call is a modem or FAX call, in which case it could cause the call to fail). For your sanity, I would strongly suggest that you don't connect spans from multiple telcos/networks/etc. on a single card, but keep each span provider on their own card. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
Thanks guys. In which case does Asterisk jump to the failed
extension?
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On 03/18/2011 05:43 PM, Gilles wrote:
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
Thanks guys. In which case does Asterisk jump to the failed
extension?
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Fri, 18 Mar 2011 17:56:12 -0500, Anthony Messina
amess...@messinet.com wrote:
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
Thanks but for some reason, after calling out through a call file,
Asterisk doesn't jump to it although the callee hangs up while
Asterisk is still playing:
===
[callback]
exten = start,1,Wait(2)
exten = start,n,ChanIsAvail(Dahdi/1)
exten = start,n,NoOp(${AVAILORIGCHAN})})
exten = start,n,Answer()
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
;not run
exten = failed,1,NoOp(Call ended with ${REASON})
===
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Re: [asterisk-users] [1.4] Failed callfile doesn't jumptofailedextension
On Sat, 19 Mar 2011, Gilles wrote: Thanks but for some reason, after calling out through a call file, Asterisk doesn't jump to it although the callee hangs up while Asterisk is still playing: Somehow, I'm guessing that 'failed' means that something failed while processing the call file or that the call failed to answer, not that somebody terminated the call. -- 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] Problem routing call to fax machine on DAHDI FXS port
answeronpolarityswitch=no usecallerid=yes cidsignalling=bell cidstart=ring ;hidecallerid=yes ;hidecalleridname=yes ;waitfordialtone=yes ;mwimonitor=no ;mwilevel=512 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ;mwisendtype=rpas,lrev callwaiting=yes ;restrictcid=no usecallingpres=yes sendcalleridafter = 1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2.0 txgain=8.0 group=1 callgroup=1 pickupgroup=1 ;immediate=yes immediate=no callerid = asreceived useincomingcalleridondahditransfer = yes callprogress=no progzone=us ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no faxbuffers=6,full ;callerid=XXX1919XXX ;channel = 3 ;callerid=XXX1919XXX ;channel = 4 #include dahdi-channels.conf My dahdi-channels.conf file look like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Nov 30 19:08:07 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) ;;; line=1 WCTDM/4/0 FXOKS (In use) (SWEC: MG2) signalling=fxo_ks callerid=Channel 14001 mailbox=5000 group=5 context=from-fax-machine channel = 1 callerid= mailbox= group= context=default ;;; line=2 WCTDM/4/1 FXOKS (In use) (SWEC: MG2) signalling=fxo_ks callerid=Channel 24002 mailbox=6000 group=5 context=from-internal channel = 2 callerid= mailbox= group= context=default ;;; line=3 WCTDM/4/2 FXSKS (In use) (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-3 channel = 3 callerid= group= context=default ;;; line=4 WCTDM/4/3 FXSKS (In use) (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-4 channel = 4 callerid= group= context=default -- _ -- Bandwidth and Colocation Provided byhttp://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 What happens when you call the fax machine from a cell phone, for instance ? Can you ear the fax machine answering ? -- next part -- An HTML attachment was scrubbed... URL:http://lists.digium.com/pipermail/asterisk-users/attachments/20110318/3c81d121/attachment-0001.htm If I can the fax machine from the outside the call is routed to the expected voice extension as it is not a fax call: Starting simple switch on 'DAHDI/4-1' -- Executing [s@from-pstn-4:1] Wait(DAHDI/4-1, 1) in new stack -- Executing [s@from-pstn-4:2] Verbose(DAHDI/4-1, CALLERID is 919XXX) in new stack CALLERID is 919XXX -- Executing [s@from-pstn-4:3] Verbose(DAHDI/4-1, Time is 20110318-210123) in new stack Time is 20110318-210123 -- Executing [s@from-pstn-4:4] Answer(DAHDI/4-1, ) in new stack -- Executing [s@from-pstn-4:5] Ringing(DAHDI/4-1, ) in new stack -- Executing [s@from-pstn-4:6] Wait(DAHDI/4-1, 6) in new stack -- Executing [s@from-pstn-4:7] Dial(DAHDI/4-1, SIP/1000) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 1000 -- SIP/1000-000b is ringing == Spawn extension (from-pstn-4, s, 7) exited non-zero on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' This is what's supposed to happen for non-fax calls. Fax calls still don't route to the DAHDI port correctly. -- _ -- 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] Problem routing call to fax machine on DAHDI FXS port
,Answer exten = s,n,Ringing exten = s,n,Wait(6) exten = s,n,Dial(SIP/1000) exten = s,n,Voicemail(1000,u) exten = s,n,Hangup My chan_dahdi.conf file looks like: [trunkgroups] ;trunkgroup = 1,1 ;trunkgroup = 2,2 ;trunkgroup = 3,3 ;trunkgroup = 4,4 ;spanmap = 1,1 ;spanmap = 2,2 ;spanmap = 3,3 ;spanmap = 4,4 [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes answeronpolarityswitch=no usecallerid=yes cidsignalling=bell cidstart=ring ;hidecallerid=yes ;hidecalleridname=yes ;waitfordialtone=yes ;mwimonitor=no ;mwilevel=512 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ;mwisendtype=rpas,lrev callwaiting=yes ;restrictcid=no usecallingpres=yes sendcalleridafter = 1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2.0 txgain=8.0 group=1 callgroup=1 pickupgroup=1 ;immediate=yes immediate=no callerid = asreceived useincomingcalleridondahditransfer = yes callprogress=no progzone=us ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no faxbuffers=6,full ;callerid=XXX1919XXX ;channel = 3 ;callerid=XXX1919XXX ;channel = 4 #include dahdi-channels.conf My dahdi-channels.conf file look like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Nov 30 19:08:07 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) ;;; line=1 WCTDM/4/0 FXOKS (In use) (SWEC: MG2) signalling=fxo_ks callerid=Channel 14001 mailbox=5000 group=5 context=from-fax-machine channel = 1 callerid= mailbox= group= context=default ;;; line=2 WCTDM/4/1 FXOKS (In use) (SWEC: MG2) signalling=fxo_ks callerid=Channel 24002 mailbox=6000 group=5 context=from-internal channel = 2 callerid= mailbox= group= context=default ;;; line=3 WCTDM/4/2 FXSKS (In use) (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-3 channel = 3 callerid= group= context=default ;;; line=4 WCTDM/4/3 FXSKS (In use) (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-4 channel = 4 callerid= group= context=default -- _ -- Bandwidth and Colocation Provided byhttp://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 What happens when you call the fax machine from a cell phone, for instance ? Can you ear the fax machine answering ? -- next part -- An HTML attachment was scrubbed... URL:http://lists.digium.com/pipermail/asterisk-users/attachments/20110318/3c81d121/attachment-0001.htm If I can the fax machine from the outside the call is routed to the expected voice extension as it is not a fax call: Starting simple switch on 'DAHDI/4-1' -- Executing [s@from-pstn-4:1] Wait(DAHDI/4-1, 1) in new stack -- Executing [s@from-pstn-4:2] Verbose(DAHDI/4-1, CALLERID is 919XXX) in new stack CALLERID is 919XXX -- Executing [s@from-pstn-4:3] Verbose(DAHDI/4-1, Time is 20110318-210123) in new stack Time is 20110318-210123 -- Executing [s@from-pstn-4:4] Answer(DAHDI/4-1, ) in new stack -- Executing [s@from-pstn-4:5] Ringing(DAHDI/4-1, ) in new stack -- Executing [s@from-pstn-4:6] Wait(DAHDI/4-1, 6) in new stack -- Executing [s@from-pstn-4:7] Dial(DAHDI/4-1, SIP/1000) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 1000 -- SIP/1000-000b is ringing == Spawn extension (from-pstn-4, s, 7) exited non-zero on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' This is what's supposed to happen for non-fax calls. Fax calls still don't route to the DAHDI port correctly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] Problem routing call to fax machine on DAHDI FXS port
,Answer exten = s,n,Ringing exten = s,n,Wait(6) exten = s,n,Dial(SIP/1000) exten = s,n,Voicemail(1000,u) exten = s,n,Hangup My chan_dahdi.conf file looks like: [trunkgroups] ;trunkgroup = 1,1 ;trunkgroup = 2,2 ;trunkgroup = 3,3 ;trunkgroup = 4,4 ;spanmap = 1,1 ;spanmap = 2,2 ;spanmap = 3,3 ;spanmap = 4,4 [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes answeronpolarityswitch=no usecallerid=yes cidsignalling=bell cidstart=ring ;hidecallerid=yes ;hidecalleridname=yes ;waitfordialtone=yes ;mwimonitor=no ;mwilevel=512 ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ;mwisendtype=rpas,lrev callwaiting=yes ;restrictcid=no usecallingpres=yes sendcalleridafter = 1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2.0 txgain=8.0 group=1 callgroup=1 pickupgroup=1 ;immediate=yes immediate=no callerid = asreceived useincomingcalleridondahditransfer = yes callprogress=no progzone=us ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no faxbuffers=6,full ;callerid=XXX1919XXX ;channel = 3 ;callerid=XXX1919XXX ;channel = 4 #include dahdi-channels.conf My dahdi-channels.conf file look like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Nov 30 19:08:07 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) ;;; line=1 WCTDM/4/0 FXOKS (In use) (SWEC: MG2) signalling=fxo_ks callerid=Channel 14001 mailbox=5000 group=5 context=from-fax-machine channel = 1 callerid= mailbox= group= context=default ;;; line=2 WCTDM/4/1 FXOKS (In use) (SWEC: MG2) signalling=fxo_ks callerid=Channel 24002 mailbox=6000 group=5 context=from-internal channel = 2 callerid= mailbox= group= context=default ;;; line=3 WCTDM/4/2 FXSKS (In use) (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-3 channel = 3 callerid= group= context=default ;;; line=4 WCTDM/4/3 FXSKS (In use) (SWEC: MG2) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn-4 channel = 4 callerid= group= context=default -- _ -- Bandwidth and Colocation Provided byhttp://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 What happens when you call the fax machine from a cell phone, for instance ? Can you ear the fax machine answering ? -- next part -- An HTML attachment was scrubbed... URL:http://lists.digium.com/pipermail/asterisk-users/attachments/20110318/3c81d121/attachment-0001.htm If I can the fax machine from the outside the call is routed to the expected voice extension as it is not a fax call: Starting simple switch on 'DAHDI/4-1' -- Executing [s@from-pstn-4:1] Wait(DAHDI/4-1, 1) in new stack -- Executing [s@from-pstn-4:2] Verbose(DAHDI/4-1, CALLERID is 919XXX) in new stack CALLERID is 919XXX -- Executing [s@from-pstn-4:3] Verbose(DAHDI/4-1, Time is 20110318-210123) in new stack Time is 20110318-210123 -- Executing [s@from-pstn-4:4] Answer(DAHDI/4-1, ) in new stack -- Executing [s@from-pstn-4:5] Ringing(DAHDI/4-1, ) in new stack -- Executing [s@from-pstn-4:6] Wait(DAHDI/4-1, 6) in new stack -- Executing [s@from-pstn-4:7] Dial(DAHDI/4-1, SIP/1000) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 1000 -- SIP/1000-000b is ringing == Spawn extension (from-pstn-4, s, 7) exited non-zero on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' This is what's supposed to happen for non-fax calls. Fax calls still don't route to the DAHDI port correctly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
