Re: [asterisk-users] [Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1
On Mon, 21 Feb 2011 22:12:47 -0600, Shaun Ruffell sruff...@digium.com wrote: I don't have much direct experience with the cards supported by the wctdm driver, but based on what you show here, it appears more like either a fundamental PCI bus incompatibility, the card isn't seated properly in the slot, or failed hardware. Right on! Moving the PCI card to another slot solved the problem. 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] AddQueueMember and stateinterface question
Hi,
I have missed something so I wonder if someone could explain for me?
0424449647 desktop phone
0106024647 DECT phone
0424449630 Helsingborg queue
extensions.conf
---
[support]
exten = 0424449647,hint,SIP/0424449647SIP/0106024647
exten = 0424449647,1,Dial(SIP/0424449647SIP/0106024647,15,rtT)
[inputinterior.se]
exten = 0/0424449647,1,Answer()
exten = 0/0424449647,n,RemoveQueueMember(Helsinborg,Local/0424449647@support)
exten = 0/0424449647,n,Hangup()
;
exten = 1/0424449647,1,Answer()
exten = 1/0424449647,n,RemoveQueueMember(Helsinborg,Local/0424449647@support)
exten =
1/0424449647,n,AddQueueMember(Helsinborg,Local/0424449647@support,1,,Lisbeth
Mingert Nilsson,SIP/0424449647)
exten = 1/0424449647,n,Hangup()
;
exten = 0424449630,1,Answer()
exten =
0424449630,n,ExecIf($[${QUEUE_MEMBER(Helsingborg,logged)}=0]?Queue(Goteborg,rtT))
exten = 0424449630,n,Queue(Helsingborg,nrtT)
If i dial 0424449630 both desktop and DECT phone rings (if 0424449647 is logged
in ofc)
If desktop phone is answering, everything is fine:
Lisbeth Mingert Nilsson (Local/0424449647@support) with penalty 1 (dynamic) (In
use) has taken no calls yet
But if DECT phone is a answering:
Lisbeth Mingert Nilsson (Local/0424449647@support) with penalty 1 (dynamic)
(Not in use) has taken 1 calls (last was 136 secs ago)
I am looking for a way to monitor both phones.
I hought i could do something like:
exten =
1/0424449647,n,AddQueueMember(Helsinborg,Local/0424449647@support,1,,Lisbeth
Mingert Nilsson,SIP/0424449647SIP/0106024647)
But that didn't work.
/Magnus--
_
-- 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] cmd MySQL
Try rrplacing MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
With MySQL(Query resultid ${conn_id} SELECT `ramal` FROM
`colaboradores` WHERE `ramal`='${EXTEN}');
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Sent: 18 February 2011 17:57
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] cmd MySQL
Hi guys,
I'm trying to connect Asterisk to the MySQL, but I can't execute it. It
returns an error, as below:
-- Executing [200@teste:2] MYSQL(Console/dsp, Query resultid 1
SELECT/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200) in new stack
[Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query:
aMYSQL_query: mysql_query failed. Error: You have an error in your SQL
syntax; check the manual that corresponds to your MySQL server version
for the right syntax to use near '/ ramal/ FROM/ colaboradores/ WHERE/
ramal=200' at line 1
Its seems it can connect to mysql
My extension (AEL) is:
MySQL(Connect conn_id localhost root 123456 crm);
MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
MySQL(Fetch fetchid ${resultid} RAMAL);
MySQL(Clear ${fetchid});
MySQL(Disconnect ${connid});
MySQL(Clear ${connid});
NoOp(${RAMAL});
Where is the error? Thanks!!
The MySQL server is in the same server where Asterisk is running.
Thanks!!!
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
--
_
-- 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] lua -asterisk manual
Hi again
Could anybody pls share some thoughts about dialplan in lua? I mean some say
it works faster...I have tested my dialplan with pbx_config
(extensions.conf) , then with ael. Dialplan is not very complex (just some
selects in mysql, then based on select some if, then...etc) I think it is
just easier to use some script language to program it.
Does anyone know any drawbacks for using lua?
Does it work stable with asterisk?
And for example why lua and not just ael? Ael seams also being convenient.
On Fri, Feb 18, 2011 at 3:51 PM, Borin katerin.bo...@gmail.com wrote:
Pls could you share some lua config which contains mysql quires
On Fri, Feb 18, 2011 at 3:38 PM, Faisal Hanif fai...@vopium.com wrote:
The only specific you need to do in extensions.lua is create a table to
put your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
*Sent:* Friday, February 18, 2011 4:33 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] lua -asterisk manual
Please could someone advise good manual for using lua for asterisk
dialplan. There is not much docu about it.
--
_
-- 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] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
Hello
Incoming calls from the FXO trigger an AGI script which simply NOOP
data sent by Asterisk through stdin.
The first two NOOP work fine, but after this, Asterisk isn't happy:
= extensions.conf
...
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(CID=${CALLERID(num)})
exten = s,n,AGI(/var/tmp/test.lua)
exten = s,n,Wait(5)
exten = s,n,Hangup
= /var/tmp/test.lua
#!/usr/bin/lua
--Must empty stdin for CHANNEL STATUS to work
while true do
local line = io.read()
if line == then break end
io.write(NOOP ,line,\n)
end
= Console
centos*CLI
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from_fxo:1] Wait(DAHDI/1-1, 2) in new stack
-- Executing [s@from_fxo:2] Set(DAHDI/1-1, CID=123465) in new
stack
-- Executing [s@from_fxo:3] AGI(DAHDI/1-1, /var/tmp/test.lua)
in new stack
-- Launched AGI Script /var/tmp/test.lua
AGI Tx agi_request: /var/tmp/test.lua
AGI Tx agi_channel: DAHDI/1-1
AGI Tx agi_language: en
AGI Tx agi_type: DAHDI
AGI Tx agi_uniqueid: 1298367207.9
AGI Tx agi_callerid: 0177628460
AGI Tx agi_calleridname: unknown
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: unknown
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from_fxo
AGI Tx agi_extension: s
AGI Tx agi_priority: 3
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx NOOP agi_request: /var/tmp/test.lua
AGI Tx 200 result=0
AGI Rx NOOP agi_channel: DAHDI/1-1
AGI Tx 200 result=0
[Feb 22 10:33:30] ERROR[5444]: utils.c:967 ast_carefulwrite: write()
returned error: Broken pipe
AGI Rx NOOP agi_language: en
AGI Tx 200 result=0
[Feb 22 10:33:30] ERROR[5444]: utils.c:967 ast_carefulwrite: write()
returned error: Broken pipe
=
Has someone experienced the same thing? Am I doing it wrong, or is
1.4.39.1 broken and I should downgrade to a known, good build?
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
Re: [asterisk-users] calls are not going thru e1 line
This is very strange. Everything matches mine except Asterisk itself
(I'm using 1.6.2.16.1).
I did notice that you set the loadzone(s) for UK use - yet your e-mail
address in in Poland. Are you setting this up in the UK?
BTW - you have a typo in chan_dahdi.conf (busydetec=yes is missing the
't' [I wonder if this is causing your problem - as the 'include' is
after this]) and I'd cetainly remove pulsedial=yes ;).
Anyway, here's the part of my chan_dahdi.conf that is working for me
(I've changed the context to match yours):
;chan_dahdi.conf
[trunkgroups]
[channels]
language = en
context = incoming_calls
switchtype = euroisdn
pridialplan = unknown
prilocaldialplan = unknown
internationalprefix = 00
nationalprefix = 0
localprefix =
unknownprefix =
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
sendcalleridafter = 1
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no
faxdetect = no
echocancel = yes
echocancelwhenbridged = no
echotraining = yes
signalling = pri_cpe
channel = 1-15,17-31
Maybe drop mine in as a replacement and see what happens then (remember
to back yours up).
BTW - you don't need to include dahdi-channels.conf in the above - as
it's already included.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Sent: 21 February 2011 13:53
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] calls are not going thru e1 line
Hi Andrew,
I am using current versions of software, find below versions:
1.) asterisk
voice:~# asterisk -V
Asterisk 1.8.2.3
2.)dahdi
*CLI dahdi show version
DAHDI Version: 2.4.0 Echo Canceller: MG2
3.) lipri
*CLI pri show version
libpri version: 1.4.11.5
I've already tried to call over each channel from 1 to 15 (i have only
15B channels)
exten = _X.,n,Dial(DAHDI/1/${EXTEN})
exten = _X.,n,Dial(DAHDI/2/${EXTEN})
exten = _X.,n,Dial(DAHDI/15/${EXTEN})
but everytime i am getting the same DIALSTATUS
snip
-- Channel 0/1, span 1 got hangup request, cause 31
...
-- Auto fallthrough, channel 'SIP/2000-0002' status is 'CHANUNAVAIL'
/snip
Regards,
Robert
On 21.02.2011 12:13, Andrew Thomas wrote:
I'm curious as to what versions of everything you are using. Reason
being this line -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-.
It states DAHDI/i1/00256312261627-1... and I don't recall seeing that
before (my 2.4.0 says -- DAHDI/1-1 is proceeding passing it to
SIP/801-000c [1-1 being the span and channel numbers]).
What happens if you change exten = _X.,n,Dial(DAHDI/g1/${EXTEN}) to
exten = _X.,n,Dial(DAHDI/1/${EXTEN})?
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
--
_
-- 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] Assigning an extension to a roaming phone
exten = 3001,n,playback(vm-youhave)
I do have the file in /usr/share/asterisk/sounds:
-rw-r--r-- 1 root root 1452 2008-03-06 00:39 vm-youhave.gsm
but still it does not play it ?!
The goodbye at the end does play correctly.
** vm-goodbye is in /usr/share/asterisk/sounds?
Yes it is.
$ ls -al /usr/share/asterisk/sounds/vm-goodb*
-rw-r--r-- 1 root root 1683 2008-03-06 00:39
/usr/share/asterisk/sounds/vm-goodbye.gsm
Part two
exten = _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten = _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)
line 1 user dials 4001 and gets ${ROAM} set from ASTDB
line 2 attempts to dial SIP extension based on ${ROAM} value.
I dialed 3001, then 001. It does say 001 back.
But then 4001 does not work.
[Feb 21 17:53:06] WARNING[26195]: chan_sip.c:2921 create_addr: No such host:
001
[Feb 21 17:53:06] WARNING[26195]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
-- Axelle
I doubt you have an extension 001 in your list (the number 4001 is trying to
dial).
Well, the 001 is supposed to be created by this line, isn't it?
exten = 3001,n,Set(DB(roam/ext)=${digito})
But obviously, yes, it is not working :(
Is the ${ROAM} trying to reach an in-house extension or an outside
number?
an in-house extension.
-- Axelle
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Assigning an extension to a roaming phone
[roaming-ext]
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(you-entered)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito} to call
${CALLERID(num)})
exten = 3001,n,Set(DB(roam/${digito})=${CALLERID(num)})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
Good idea the Verbose commands, at least I see a bit better what is
happening. I should have thought about that one. Thanks.
But I don't understand the CALLERID part: the roaming user is unknown
on my network, so how could he have a correct CALLERID?
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
I tried it and I get the following logs:
[Feb 22 11:57:44] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from 'IMSI20830'
sip:IMSI20830@127.0.0.1' failed for '127.0.0.1' - No matching
peer found
= this line appears when the roaming user comes in
Create roaming extension
Setting roaming extension 4001 to call 2103
= those two lines occur when the roaming user dials 3001. Why is it
returning a callerid 2103?? 2103 corresponds to another registered
IMSI !!
Calling roaming extension 4001
[Feb 22 12:05:07] WARNING[27577]: chan_sip.c:2921 create_addr: No such
host: 2103
[Feb 22 12:05:07] WARNING[27577]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
= this is displayed when the roaming user dials 4001. It's quite
normal it can't route to 2103 because that phone is off.
Then you need to include the [roaming-ext] context in whatever context your
phones dial from. The basic idea behind this is that you need to store the
extension where your roamer is currently sitting in your DB, which you were
doing. By adding the ${CALLERID(num)} to the database, you give it an idea
of where the calls should go.
Not sure to understand.
The goal here is to assign an extension to the roaming user. He calls 3001.
Where 'the calls should go' is to the roaming user.
Now, this means your ${CALLERID(num)}
variable needs to match your SIP endpoint's name, of course, but if these
don't currently match, I'm pretty sure there is a variable you can use to
achieve the same effect.
-- Axelle
--
_
-- 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 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.22 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.4 Security advisory AST-2011-002 is available at: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf 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] Assigning an extension to a roaming phone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Tuesday, February 22, 2011 5:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to a roaming phone
[roaming-ext]
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(you-entered)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito} to call
${CALLERID(num)})
exten = 3001,n,Set(DB(roam/${digito})=${CALLERID(num)})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
Good idea the Verbose commands, at least I see a bit better what is
happening. I should have thought about that one. Thanks.
But I don't understand the CALLERID part: the roaming user is unknown
on my network, so how could he have a correct CALLERID?
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
I tried it and I get the following logs:
[Feb 22 11:57:44] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from 'IMSI20830'
sip:IMSI20830@127.0.0.1' failed for '127.0.0.1' - No matching
peer found
= this line appears when the roaming user comes in
snip
Axelle, please post the CLI output from the 3001 call and I'll put up a
dialplan that should work for 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
Re: [asterisk-users] Assigning an extension to a roaming phone
H Danny,
Axelle, please post the CLI output from the 3001 call and I'll put up a
dialplan that should work for you.
So this is the output I get:
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on
openbts (pid = 20597)
[Feb 22 15:18:02] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from
'IMSI20830061sip:IMSI20830061@127.0.0.1' failed for
'127.0.0.1' - No matching peer found
Create roaming extension
Caller IMSI is
Setting roaming extension 4001
Calling roaming extension 4001
[Feb 22 15:22:42] WARNING[9512]: chan_sip.c:2921 create_addr: No such host: 4001
[Feb 22 15:22:42] WARNING[9512]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
openbts*CLI database show
/SIP/Registry/IMSI20810 :
127.0.0.1:5062:3600:IMSI20810:sip:IMSI20810@127.0.0.1:5062
/SIP/Registry/IMSI20830061 :
127.0.0.1:5062:3600:IMSI20830061:sip:IMSI20830061@127.0.0.1:5062
/SIP/Registry/IMSI2083044xxx :
127.0.0.1:5062:3600:IMSI2083044:sip:IMSI2083044@127.0.0.1:5062
/roam/001 : 4001
/roam/002 : 2103
/roam/003 : 4003
/roam/007 : 4007
/roam/ext : 001
openbts*CLI
My current extensions.conf is
[globals]
; This is the extensions file used in the Burning Man 2008
; site test, with private information removed.
; Jump to the end for handset examples.
[macro-dialSIP]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(3000)
exten = s-CONGESTION,1,Congestion(3000)
exten = s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,2,Hangup
[from-trunk]
; route incoming calls from the PSTN
[sip-external]
include = sip-local
; roaming users
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Verbose(Caller IMSI is ${IMSI})
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito})
exten = 3001,n,Set(DB(roam/${digito})=4${digito})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
; outgoing trunk access
; NANP
[sip-local]
; removing full IMSI value for the cut and paste in the list
exten = 2102,1,Macro(dialSIP,IMSI20810)
exten = 2103,1,Macro(dialSIP,IMSI2083044)
; Note 2111 is commented:
;exten = 2111,1,Macro(dialSIP,IMSI20830061)
exten = 6123,1,SayNumber(${EXTEN})
My sip.conf is:
[general]
bindaddr=0.0.0.0
bindport=5060
; Comment these out if no backhaul is available.
; This is a GSM handset entry.
; You need one for each SIM.
; The IMSI is a 15-digit code in the SIM.
; You can see it in the Control log whenever a phone tries to register.
[IMSI20810]
callerid=2102
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
[IMSI2083044]
callerid=2103
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
;[IMSI20830061]
;callerid=2111
;canreinvite=no
;type=friend
;context=sip-external
;allow=gsm
;host=dynamic
-- Axelle
--
_
-- 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] Weird Inbound Problem.
Hi List, Someone may have run into this problem. Very strange. I have a customer running 1.422. They use a digium ISDN card connected to an primary rate for their inbound currently. We have tested inbound SIP from one of our trunks. We use these trunks with all our asterisk customers without an issue. With this Asterisk box when we answer an inbound SIP call to an extension literally after .5 seconds (500ms) the audio just dies going from the extension to the callee... The extension call still hear the caller. If we point the DID at a conference audio works perfectly. If we point it to an IVR which then points to the extension the audio is perfect both directions. The SIP traces look perfect, identical SDP if going to an extension or a IVR. Any clues? How would I go about debugging this, the CLI output looks fine.. Asterisk 1.4.22 built by root @ asterisk on a i686 running Linux on 2010-05-13 16:19:20 UTC Thanks in advance. Brian. -- _ -- 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] Assigning an extension to a roaming phone
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle Sent: Tuesday, February 22, 2011 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Assigning an extension to a roaming phone H Danny, Axelle, please post the CLI output from the 3001 call and I'll put up a dialplan that should work for you. snip Not what I asked for, but here's what I can tell you. From what you posted, you can dial and outside number and from in-house you can dial 2102 or 2103. The way the dialplan works is that you set up specific numbers that will be valid like you have done with 2102, 2103 and 3001 or a range of numbers that will be valid like 4000-4999. For the 4XXX magic number snippet to ever work correctly, it has to dial an outside number or a pre-defined in-house extension. From what you posted, if you dial 4002, the call should properly connect to 2103. I'm going to play with this a little and post back how you can use these two snippets for Asterisk Russian Roulette. -- _ -- 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 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Thanks Ish On Tue, 2011-02-22 at 08:02 -0500, Asterisk Development Team wrote: The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-002, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.22 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.4 Security advisory AST-2011-002 is available at: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Thanks Ish snip -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 Yes, you can take the two minutes to search for Must release lock in http://svn.asterisk.org/svn/asterisk/tags/1.8.2.4/channels/chan_sip.c ~~~ 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] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
On 11-02-22 10:16 AM, Ishfaq Malik wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 No. The ChangeLog would give you the information you're looking for. http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3 Leif. -- _ -- 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] Assigning an extension to a roaming phone
Le 22/02/2011 12:32, Axelle a écrit : Good idea the Verbose commands, at least I see a bit better what is happening. Maybe a core set verbose 3 too ? I should have thought about that one. Thanks. But I don't understand the CALLERID part: the roaming user is unknown on my network, so how could he have a correct CALLERID? Well i think the problem lies here somehow. Usual roaming setup are using a fixed number of phone, with users beeing able to register themself on some location. So you have an IP network, with SIP agents (cell phones ?), some of those are manually setup in you sip.conf file, but you want to allow unknown cell phones users to self register in your system ? Someone enter your network, dial 3001@your ipbx and get/set a temporary internal number. Then other phone can dial his ? I don't think it's possible, although ... What you need is to mimic the SIP registration process, by fetching the following informations from during the setup call: * IP of the phone * UDP/TCP Port of the SIP process * Some SIP user ID Then you store thoses in your DB in the form SIP/user@IP:port and then you could be able to Dial this string, (if the phone is ok to be dialed by an unknown party this way) Regards, benoit -- _ -- 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] Assigning an extension to a roaming phone
Axelle, please post the CLI output from the 3001 call and I'll put up a dialplan that should work for you. snip Not what I asked for, but here's what I can tell you. Oh I'm sorry but then what are you asking for? I thought it was the console messages on Asterisk. From what you posted, you can dial and outside number and from in-house you can dial 2102 or 2103. The way the dialplan works is that you set up specific numbers that will be valid like you have done with 2102, 2103 and 3001 or a range of numbers that will be valid like 4000-4999. For the 4XXX magic number snippet to ever work correctly, it has to dial an outside number or a pre-defined in-house extension. From what you posted, if you dial 4002, the call should properly connect to 2103. Yes, indeed, but that's not what I want it to do. 2103 does not correspond to anyone. Yeah, by the way, just to make that clear: the roaming phone does not have *any phone number*. I need the dialplan to assign one. Re-routing to another number won't work, as there is no other number... Thanks Axelle -- _ -- 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] Assigning an extension to a roaming phone
On Fri, 2011-02-18 at 12:36 +0100, Axelle wrote:
Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,vm-youhave,3)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Set(ROAM=${digito})
exten = 3001,n,Set(DB(roam/ext)=${digito})
exten = 3001,n,playback(vm-goodbye)
exten = 3001,n,hangup
exten = _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten = _4XXX,n,dial(SIP/${ROAM})
The idea was that the roaming phone first dials 3001, sets a 3 digits
extension (eg 123) and then I supposed that 4123 would work. But it
does not.
I am unsure about the 2 Set lines.
Can anyone help?
Regards
I think you are confusing extensions with SIP usernames. In order to
dial anything on your system you already need to know the name of the
SIP phone (unless you are not doing authentication which is a bad idea).
That way you can use Dial(SIP/sipphone). If you want to assign an an
extension to this sipphone you need to save the username to your
database to a key maybe like roam/123. That way when you lookup
roam/123 from the database you get the username of the phone and insert
that into your dial command.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Description: This is a digitally signed message part
--
_
-- 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] Assigning an extension to a roaming phone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Tuesday, February 22, 2011 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to a roaming phone
Axelle, please post the CLI output from the 3001 call and I'll put up a
dialplan that should work for you.
snip
Not what I asked for, but here's what I can tell you.
Oh I'm sorry but then what are you asking for? I thought it was the
console messages on Asterisk.
From what you posted,
you can dial and outside number and from in-house you can dial 2102 or
2103.
The way the dialplan works is that you set up specific numbers that will
be
valid like you have done with 2102, 2103 and 3001 or a range of numbers
that
will be valid like 4000-4999. For the 4XXX magic number snippet to ever
work correctly, it has to dial an outside number or a pre-defined in-house
extension.
From what you posted, if you dial 4002, the call should properly connect
to
2103.
Yes, indeed, but that's not what I want it to do. 2103 does not
correspond to anyone.
Yeah, by the way, just to make that clear: the roaming phone does not
have *any phone number*. I need the dialplan to assign one. Re-routing
to another number won't work, as there is no other number...
Thanks
Axelle
Okay. I modified the dialplan you posted to look like this
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Verbose(Caller IMSI is ${IMSI})
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito})
exten = 3001,n,Set(DB(roam/${digito})=${digito})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
Here's the output from my 3001 call -
Verbosity is at least 10
-- Executing [3001@default:1] Verbose(SIP/sipuser-006f, Create
roaming extension) in new stack
Create roaming extension
-- Executing [3001@default:2] Verbose(SIP/sipuser-006f, Caller
IMSI is ) in new stack
Caller IMSI is
-- Executing [3001@default:3] Read(SIP/sipuser-006f,
digito|beep|3) in new stack
-- Accepting a maximum of 3 digits.
-- SIP/sipuser-006f Playing 'beep' (language 'en')
-- User entered '144'
-- Executing [3001@default:4] SayDigits(SIP/sipuser-006f, 144)
in new stack
-- SIP/sipuser-006f Playing 'digits/1' (language 'en')
-- SIP/sipuser-006f Playing 'digits/4' (language 'en')
-- SIP/sipuser-006f Playing 'digits/4' (language 'en')
-- Executing [3001@default:5] Verbose(SIP/sipuser-006f, Setting
roaming extension 4144) in new stack
Setting roaming extension 4144
-- Executing [3001@default:6] Set(SIP/sipuser-006f,
DB(roam/144)=144) in new stack
-- Executing [3001@default:7] Playback(SIP/sipuser-006f,
vm-goodbye) in new stack
-- SIP/sipuser-006f Playing 'vm-goodbye' (language 'en')
-- Executing [3001@default:8] Hangup(SIP/sipuser-006f, ) in new
stack
== Spawn extension (default, 3001, 8) exited non-zero on
'SIP/sipuser-006f'
-- Executing [h@default:1] Goto(SIP/sipuser-006f, end-call|h|1)
in new stack
-- Goto (end-call,h,1)
-- Executing [h@end-call:1] Hangup(SIP/sipuser-006f, ) in new
stack
== Spawn extension (end-call, h, 1) exited non-zero on 'SIP/sipuser-
006f'
* and my 4144 call *
-- Executing [4144@default:1] Verbose(SIP/sipuser-0070, Calling
roaming extension 4144) in new stack
Calling roaming extension 4144
-- Executing [4144@default:2] Set(SIP/sipuser-0070, ROAMEXT=144)
in new stack
-- Executing [4144@default:3] Dial(SIP/sipuser-0070, SIP/144|30)
in new stack
-- Called 144
-- SIP/144-0071 is ringing
-- SIP/144-0071 answered SIP/sipuser-0070
-- Native bridging SIP/sipuser-0070 and SIP/144-0071
-- Executing [h@default:1] Goto(SIP/sipuser-0070, end-call|h|1)
in new stack
-- Goto (end-call,h,1)
-- Executing [h@end-call:1] Hangup(SIP/sipuser-0070, ) in new
stack
Still not probably what you are looking for, but maybe it will steer you
along.
--
_
-- 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] calls between iax and sip
Hello, i have asterisk installed and i have configured a client iax and sip without any issue, when i call a internal extension sip from iax there is no problem but when i call a iax extension from sip extension the result is KO(wrong number) any help please 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] calls between iax and sip
On Tue, 22 Feb 2011, salaheddine elharit wrote: i have asterisk installed and i have configured a client iax and sip without any issue, when i call a internal extension sip from iax there is no problem but when i call a iax extension from sip extension the result is KO(wrong number) any help please No details, no help. Crank up verbosity on the CLI and see if the messages yield a clue. If not, please post the console messages. -- 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] calls between iax and sip
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, February 22, 2011 12:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] calls between iax and sip On Tue, 22 Feb 2011, salaheddine elharit wrote: i have asterisk installed and i have configured a client iax and sip without any issue, when i call a internal extension sip from iax there is no problem but when i call a iax extension from sip extension the result is KO(wrong number) any help please No details, no help. Crank up verbosity on the CLI and see if the messages yield a clue. If not, please post the console messages. Isn't Dionne Warrick a poster on this list? :) -- _ -- 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] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
You're not properly reading in the response after each NoOp you send out. Each
time you send something to asterisk in AGI, you must read the response in your
script.
Thanks,
--Warren Selby, dCAP
On Feb 22, 2011, at 4:39 AM, Gilles codecompl...@free.fr wrote:
Hello
Incoming calls from the FXO trigger an AGI script which simply NOOP
data sent by Asterisk through stdin.
The first two NOOP work fine, but after this, Asterisk isn't happy:
= extensions.conf
...
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(CID=${CALLERID(num)})
exten = s,n,AGI(/var/tmp/test.lua)
exten = s,n,Wait(5)
exten = s,n,Hangup
= /var/tmp/test.lua
#!/usr/bin/lua
--Must empty stdin for CHANNEL STATUS to work
while true do
local line = io.read()
if line == then break end
io.write(NOOP ,line,\n)
end
= Console
centos*CLI
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from_fxo:1] Wait(DAHDI/1-1, 2) in new stack
-- Executing [s@from_fxo:2] Set(DAHDI/1-1, CID=123465) in new
stack
-- Executing [s@from_fxo:3] AGI(DAHDI/1-1, /var/tmp/test.lua)
in new stack
-- Launched AGI Script /var/tmp/test.lua
AGI Tx agi_request: /var/tmp/test.lua
AGI Tx agi_channel: DAHDI/1-1
AGI Tx agi_language: en
AGI Tx agi_type: DAHDI
AGI Tx agi_uniqueid: 1298367207.9
AGI Tx agi_callerid: 0177628460
AGI Tx agi_calleridname: unknown
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: unknown
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from_fxo
AGI Tx agi_extension: s
AGI Tx agi_priority: 3
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx NOOP agi_request: /var/tmp/test.lua
AGI Tx 200 result=0
AGI Rx NOOP agi_channel: DAHDI/1-1
AGI Tx 200 result=0
[Feb 22 10:33:30] ERROR[5444]: utils.c:967 ast_carefulwrite: write()
returned error: Broken pipe
AGI Rx NOOP agi_language: en
AGI Tx 200 result=0
[Feb 22 10:33:30] ERROR[5444]: utils.c:967 ast_carefulwrite: write()
returned error: Broken pipe
=
Has someone experienced the same thing? Am I doing it wrong, or is
1.4.39.1 broken and I should downgrade to a known, good build?
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
--
_
-- 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] Multiple public address to one Asterisk server behind NAT?
I have a situation where an Asterisk server is NATted, sitting behind a PIX. One public IP is used for one purpose, now a second public IP is required for another. Is there a way to have Asterisk use more than one public IP when behind NAT? (I already use the externalIP setting)... If not, any suggestions for a SIMPLE way to do this? Thanks MD -- _ -- 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] Multiple public address to one Asterisk serverbehind NAT?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, February 22, 2011 3:34 PM To: Asterisk Users List Subject: [asterisk-users] Multiple public address to one Asterisk serverbehind NAT? I have a situation where an Asterisk server is NATted, sitting behind a PIX. One public IP is used for one purpose, now a second public IP is required for another. Is there a way to have Asterisk use more than one public IP when behind NAT? (I already use the externalIP setting)... If not, any suggestions for a SIMPLE way to do this? Thanks MD Dumb answer - bind to 0.0.0.0 - the externalIP setting probably trashes this idea. -- _ -- 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] Multiple public address to one Asterisk server behind NAT?
You could run two copies of asterisk on different private IP addresses. Have your current install bound to the first private IP with the externalIP set to the first public and the second install running on the other IP with the other externalIP set. On Tue, Feb 22, 2011 at 2:34 PM, Michelle Dupuis mdup...@ocg.ca wrote: I have a situation where an Asterisk server is NATted, sitting behind a PIX. One public IP is used for one purpose, now a second public IP is required for another. Is there a way to have Asterisk use more than one public IP when behind NAT? (I already use the externalIP setting)... If not, any suggestions for a SIMPLE way to do this? Thanks MD -- _ -- 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] About maxlen parameter in queues
Finally I could get it to work by running a shell script which parsed results from 'queue show' CLI command in dearch of 'Not in Use' members. It was done with an AGI. Regards, Daniel On Tue, Feb 8, 2011 at 11:52 AM, Carlos Chavez cur...@telecomabmex.comwrote: On Mon, 2011-02-07 at 10:44 -0500, Daniel - Asterisk wrote: Hi Danny, Could you please let me know what function do I use to get if the queue is full? Elder On Mon, Feb 7, 2011 at 10:42 AM, Danny Nicholas da...@debsinc.com wrote: __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel - Asterisk Sent: Monday, February 07, 2011 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] About maxlen parameter in queues Dear list, I want to avoid sending calls to a queue when it is full. From the fact that 'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like to know if there's a way to do it. Setting the Queue() timeout to a little value is not the most suitable option. I'm using asterisk 1.4.21 but I don't know if there are some options available on release 1.8 Thanks, Elder Arohuanca Lagos t. 992728100 This is a bit “hackish”, but why don’t you just make a context that uses AGI to query the queue and only let the call proceed if not full? Maybe it would be easier to use the GROUP and GROUP_COUNT functions to see how many users are in the queue and decide on that. Although this really defeats the purpose of having a Queue. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] Multiple public address to one Asterisk serverbehind NAT?
There is only one NIC internally (only 1 internal IP) so binding to 0.0.0.0 won't do anything. Asterisk uses the externIP setting to publish a different address when behind NAT, that's what externIP does. But there is only one externIP settings. I'm thinking about openSER/proxy/etc type solutions but need to keep it as simple as possible. MD From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Tuesday, February 22, 2011 4:40 PM To: Asterisk Users List Subject: Re: [asterisk-users] Multiple public address to one Asterisk serverbehind NAT? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, February 22, 2011 3:34 PM To: Asterisk Users List Subject: [asterisk-users] Multiple public address to one Asterisk serverbehind NAT? I have a situation where an Asterisk server is NATted, sitting behind a PIX. One public IP is used for one purpose, now a second public IP is required for another. Is there a way to have Asterisk use more than one public IP when behind NAT? (I already use the externalIP setting)... If not, any suggestions for a SIMPLE way to do this? Thanks MD “Dumb” answer – bind to 0.0.0.0 – the externalIP setting probably trashes this idea. -- _ -- 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] AstriEurope coference
Yeah, this is messages which i saw before. Weird is that its hidden somewhere under registration form and there was no notification about cancellation for registered users. Anyway, its a pity that AstriEurope is cancelled. Are there other similar conference in Europe in 2011 ? Regards, Albert On 22.02.2011 07:56, randulo wrote: On Mon, Feb 21, 2011 at 11:56 PM, Albert alber...@wp.pl wrote: does anyone know is AstriEurope coference is still on ? http://www.astrieurop.com/fr/cloture.php Cancelled. Hello, It is with regret that we announce you the cancellation of the AstriEurop exhibition on May, 3rd and 4th 2011 in Paris. We thank all the companies/partners having supported this project. /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 -- _ -- 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] calls are not going thru e1 line
Hi Andrew,
thanks for your answer. I haven't notice this typo before, i was
replacing this config so many times ;-)
I did as you suggested, replaced with your config but result is still
the same.
Some technicians from telco came yesterday to investigate and confirmed
that something is wrong at they end, now i am waiting for them to clear
this issue.
I am not setting this up in UK, but in Uganda. That's why i am using
loadzone from UK.
I will keep you posted if my issue was solved.
Thanks,
Albert
On 22.02.2011 11:55, Andrew Thomas wrote:
This is very strange. Everything matches mine except Asterisk itself
(I'm using 1.6.2.16.1).
I did notice that you set the loadzone(s) for UK use - yet your e-mail
address in in Poland. Are you setting this up in the UK?
BTW - you have a typo in chan_dahdi.conf (busydetec=yes is missing the
't' [I wonder if this is causing your problem - as the 'include' is
after this]) and I'd cetainly remove pulsedial=yes ;).
Anyway, here's the part of my chan_dahdi.conf that is working for me
(I've changed the context to match yours):
;chan_dahdi.conf
[trunkgroups]
[channels]
language = en
context = incoming_calls
switchtype = euroisdn
pridialplan = unknown
prilocaldialplan = unknown
internationalprefix = 00
nationalprefix = 0
localprefix =
unknownprefix =
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
sendcalleridafter = 1
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no
faxdetect = no
echocancel = yes
echocancelwhenbridged = no
echotraining = yes
signalling = pri_cpe
channel = 1-15,17-31
Maybe drop mine in as a replacement and see what happens then (remember
to back yours up).
BTW - you don't need to include dahdi-channels.conf in the above - as
it's already included.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Sent: 21 February 2011 13:53
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] calls are not going thru e1 line
Hi Andrew,
I am using current versions of software, find below versions:
1.) asterisk
voice:~# asterisk -V
Asterisk 1.8.2.3
2.)dahdi
*CLI dahdi show version
DAHDI Version: 2.4.0 Echo Canceller: MG2
3.) lipri
*CLI pri show version
libpri version: 1.4.11.5
I've already tried to call over each channel from 1 to 15 (i have only
15B channels)
exten = _X.,n,Dial(DAHDI/1/${EXTEN})
exten = _X.,n,Dial(DAHDI/2/${EXTEN})
exten = _X.,n,Dial(DAHDI/15/${EXTEN})
but everytime i am getting the same DIALSTATUS
snip
-- Channel 0/1, span 1 got hangup request, cause 31
...
-- Auto fallthrough, channel 'SIP/2000-0002' status is 'CHANUNAVAIL'
/snip
Regards,
Robert
On 21.02.2011 12:13, Andrew Thomas wrote:
I'm curious as to what versions of everything you are using. Reason
being this line -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-.
It states DAHDI/i1/00256312261627-1... and I don't recall seeing that
before (my 2.4.0 says -- DAHDI/1-1 is proceeding passing it to
SIP/801-000c [1-1 being the span and channel numbers]).
What happens if you change exten = _X.,n,Dial(DAHDI/g1/${EXTEN}) to
exten = _X.,n,Dial(DAHDI/1/${EXTEN})?
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
--
_
-- 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] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
On Tue, 22 Feb 2011 14:54:38 -0600, Warren Selby wcse...@selbytech.com wrote: You're not properly reading in the response after each NoOp you send out. Each time you send something to asterisk in AGI, you must read the response in your script. Thanks for the tip. It's working now. -- _ -- 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] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
On Tue, 22 Feb 2011 14:54:38 -0600, Warren Selby You're not properly reading in the response after each NoOp you send out. Each time you send something to asterisk in AGI, you must read the response in your script. On Wed, 23 Feb 2011, Gilles wrote: Thanks for the tip. It's working now. While the documentation on the protocol is clear, nobody gets it right the first time -- which is why I always suggest using an established library for the language of your choice. -- 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 installing FXS module in old digium 4 channel tdm card
How/Where would I do that? TIA CF On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote: On 2/21/11 4:46 PM, C F wrote: I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work. Running dmesg I can see the following error: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Module 0: FAILED FXS (FCC) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules) Does this have to do with the fact that the module is way newer than the card? Not having much direct experience with the wctdm.c driver, that would be my guess. You might be able to edit the wctdm_proslic_insane() function to force the FLAG_3215 on for the card and see if that gives you a different result. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Problem installing FXS module in old digium 4 channel tdm card
On Tue, Feb 22, 2011 at 11:00:22PM -0500, C F wrote: On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote: On 2/21/11 4:46 PM, C F wrote: I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work. Running dmesg I can see the following error: [snip] ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Module 0: FAILED FXS (FCC) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules) Does this have to do with the fact that the module is way newer than the card? Not having much direct experience with the wctdm.c driver, that would be my guess. You might be able to edit the wctdm_proslic_insane() function to force the FLAG_3215 on for the card and see if that gives you a different result. How/Where would I do that? Around line 1297 of drivers/dahdi/wctdm.c you could change: if (wctdm_getreg(wc, card, 1) 0x80) /* ProSLIC 3215, not a 3210 */ wc-flags[card] |= FLAG_3215; to wc-flags[card] |= FLAG_3215; and just skip the read of register 1. I don't know if this will work in your case, but it's something to try. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Dialplan execution stops on app call even with TryExec (Am I missing something simple?)
Figured it out...
1) Incoming SIP immediately routed out a Dahdi PRI trunk is answered
just before it dials the trunk
2) CNG detected after call is bridged
3) Call redirected to fax extension AFTER the bridge is torn down and
the hangup extension is run in the original Dial context
4) Fax extension executes and ReceiveFax detects T.38 hangup when fax is
completed
5) pbx WON'T run hangup extension (again) in fax ext. context so no more
dialplan execution after the ReceiveFax (so no fax delivery capabilities)
Not sure about impact but I modified chan_sip.c after CNG is detected
and just before the redirect (to the fax extension) and told it not to
run the hangup extension in the original Dial context by setting
AST_FLAG_BRIDGE_HANGUP_DONT. This appears to have fixed the issue.
-- Jay
On 2/21/2011 10:47 AM, Jay Reeder wrote:
We're having an issue where we call ReceiveFax in a context that
includes a hangup extension and half the time dialplan execution
doesn't continue after the fax is received successfully. Am I missing
something simple here? Below is a sample call where this happened:
The last log line for this channel/call is:
[Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel
'SIP/Level3_sip_peer_mcqueen-2c3d' FAX session '228' is complete, result:
'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 8, resolution: '204x196',
transfer rate: '9600', remoteSID: 'TIME'
The context it's executing in is:
[ext-fax-voicenation]
exten = s,1,Noop(Receiving Fax for: ${FROM_DID} From: ${CALLERID(all)})
exten = s,n(receivefax),StopPlaytones
exten =
s,n,Set(FAX_FILE_NAME=/var/www/html/vncake/fax_temp/${FROM_DID}-${CALLERID(number)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.tif)
; Gafachi is known to have a broken ecm implementation - disable on receive -
also send with 'z' option
exten = s,n,Set(trunk_name=${CUT(CHANNEL,-,1)})
exten = s,n,Noop(trunk name is ${trunk_name:4})
exten = s,n,ExecIf($[ ${trunk_name:4:7} = gafachi]?Set(FAXOPT(ecm)=no))
;--
; Level3 V17/V34 modems were unreliable and V17 (14400) wasn't working so we
downgrade to slower fax-modems
exten = s,n,Set(FAXOPT(modem)=V27,V29)
;--
exten = s,n,Set(FAXDELIVERED=no)
exten = s,n,TryExec(ReceiveFAX(${FAX_FILE_NAME},f))
exten = s,n,System(/var/www/html/vncake/cake/console/cake -app
/var/www/html/vncake/app email_fax ${FAX_FILE_NAME} ${FAXPAGES}
err:${FAXOPT(error)})
exten = s,n,Set(FAXDELIVERED=yes)
exten = s,n,ExecIf($[${FAXOPT(error)}=]?Set(FAXSTATUS=FAILED LICENSE
EXCEEDED))
exten = s,n,ExecIf($[${FAXOPT(error)}!=
${FAXOPT(error)}!=NO_ERROR]?Set(FAXSTATUS=FAILED FAXOPT: error:
${FAXOPT(error)} status: ${FAXOPT(status)} statusstr: ${FAXOPT(statusstr)}))
exten = s,n,Hangup
exten = h,1,Noop(*** process fax now ***)
exten = h,n,GotoIf($[${FAXDELIVERED} = yes]?end)
; if hangup while processing script above(before flag set =yes) then will
jump to hangup and double process - need to pause here so script can make
adjustments
exten = h,n,System(/bin/sleep 5)
exten = h,n,System(/var/www/html/vncake/cake/console/cake -app
/var/www/html/vncake/app email_fax ${FAX_FILE_NAME} ${FAXPAGES}
err:${FAXOPT(error)})
exten = h,n,Set(FAXDELIVERED=yes)
exten = h,n(end),Macro(hangupcall,)
exten = h,process+101(failed),Noop(FAX ${FAXSTATUS} for:${FAX_RX_EMAIL} ,
From: ${CALLERID(all)})
; email to notify instability in the fax module
exten = h,n,ExecIf($[${FAXOPT(error)} = FILE_IO_FAIL]?System(echo \**
Asterisk Fax FILE_IO_FAIL - will reload. Thank you. Asterisk :)\ | mail -s
\** Asterisk Fax FILE_IO_FAIL\ **email address was here**))
; Restart Asterisk if FILE IO FAILURE on fax - indicates instability in fax
module
; accomplished by cron job that will restart asterisk as root when this file
is found
exten = h,n,ExecIf($[${FAXOPT(error)} = FILE_IO_FAIL]?System(echo
\FAX_IO_FAILURE\ /tmp/FAX_IO_FAILURE))
exten = h,n,Macro(hangupcall,)
; end of [ext-fax]
What am I missing here? Half the time we don't get back into the
dialplan from the ReceiveFax even though wrapped in TryExec. Before
wrapping the call in TryExec, I would get a log entry about ReceiveFax
exiting non-zero (after successful fax receipt) and no other log entry
for the call.
This is running on 1.6.2.17 rc3 ... we upgraded because the same thing
was happening with 1.6.2.6.
Any help would be appreciated.
Thanks,
Jay
--
_
-- 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] AstriEurope coference
On Tue, Feb 22, 2011 at 11:49 PM, Albert alber...@wp.pl wrote: Yeah, this is messages which i saw before. Weird is that its hidden somewhere under registration form and there was no notification about cancellation for registered users. Yes, it's in a popup when you try to register. I imagine they didn't want the people they will pitch for other events to see that not enough sponsor support came on board to have it. I was pretty surprised that it worked well enough last year. True, it's nice to have such events in Europe, but apparently the bottom line is that not enough business was generated last year and the majors backed out. I saw that Aastra is doing it's own tour of cities. /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
