[asterisk-users] need help

[Антон Сацкий]

please help me to solve the problem
if U can solve it for a chocolate :) it is also ok

https://issues.asterisk.org/jira/browse/ASTERISK-26073

-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
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Re: [asterisk-users] Need help with my dial plan - general logic flaws

[Steve Edwards]

On Sat, 12 Mar 2016, Ivan Demkovitch wrote:


I created very simple automated attendand (with a help of book), below is code.
But logic is simple:

Depending on time - I want:
If during business hours - give them menu and handle extensions
If after hours - give them message and take to voicemail.

This dial plan accomplishes just that.

What I don’t like is side behavior of this plan
When after hours - I can still press 1/2/3 and go into Sales/Support 
queue or to operator, etc.
It’s probably OK to allow dialing of extensions of actual users (101, 102, etc)
But I don’t want them to dial operator (0) or 1,2,3 and such.


A new context is a great way to limit the scope of trouble after hours 
callers can get into.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST-- 
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[asterisk-users] Need help with my dial plan - general logic flaws

[Ivan Demkovitch]

Hello group, 

I’m developer myself but creating dial plans is little bit different I guess :) 
I created very simple automated attendand (with a help of book), below is code. 
But logic is simple:

Depending on time - I want:
If during business hours - give them menu and handle extensions
If after hours - give them message and take to voicemail.

This dial plan accomplishes just that.

What I don’t like is side behavior of this plan
When after hours - I can still press 1/2/3 and go into Sales/Support 
queue or to operator, etc.
It’s probably OK to allow dialing of extensions of actual users (101, 102, etc)
But I don’t want them to dial operator (0) or 1,2,3 and such.

How do I fix it?

Thank you!
Ivan







[automated_attendant]

exten => fax,1,Goto(fax_incoming,fax,1)

exten => s,1,Verbose(1, Caller ${CALLERID(all)} has entered the auto attendant)
   same => n,Answer()
   same => n,Set(TIMEOUT(digit)=2) ;this sets the inter-digit timer
   same => n,Wait(1)   ;wait 1 sec to establish audio
   same => 
n(menustart),GotoIfTime(9:00-17:00,mon-fri,*,*?daygreeting:afterhoursgreeting)  
   ;depending if it's work time or not go to labels

   same => 
n(afterhoursgreeting),Background(/etc/asterisk/automated-attendant-prompts/ditat_afterhour_greeting)
 ; after hours greeting
   same => n,VoiceMail(99@default,u)
   same => n,Hangup()

   same => 
n(daygreeting),Background(/etc/asterisk/automated-attendant-prompts/ditat_main_greeting)
  ; day greeting
   same => n,WaitExten(4)  ; wait 4 
sec max before give up
   same => n,Goto(0,1) ; treat 
as caller pressed '0'

exten => 1,1,Verbose(1, Caller ${CALLERID(all)} has entered the sales queue)
   same => n,Goto(Queues,7001,1)   ; Sales 
Queue

exten => 2,1,Verbose(1, Caller ${CALLERID(all)} has entered the support queue)
   same => n,Goto(Queues,7002,1)   ; 
Support Queue

exten => 3,1,Verbose(1, Caller ${CALLERID(all)} has entered the support queue 
via Other extension)
   same => n,Goto(Queues,7002,1)   ; 
Support Queue

exten => 0,1,Verbose(1, Caller ${CALLERID(all)} wants an operator)
   same => n,Goto(internal,101,1)

exten => i,1,Verbose(1, Caller ${CALLERID(all)} has entered an invalid 
selection)
   same => n,Playback(invalid)
   same => n,Goto(s,menustart)

exten => t,1,Verbose(1, Caller ${CALLERID(all)} has timed out)
   same => n,Goto(0,1)

; -- CALLING EXTENSIONS DIRECTLY -
exten => _1XX,1,Verbose(1,Call to an extension starting with '1')
   same => n,Goto(internal,${EXTEN},1)
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[asterisk-users] Need help interpreting SDP on failing WebRTC connection

[Antonio Gómez Soto]

Hi,

I am trying to setup a WebRTC connection to asterisk 1.13.0.
Using Bria a regular SIP connection works, but using sipml5 on chrome, I
got nothing.

My network setup by the way: I am working behind a comcast cable modem, the
test setup is at digital ocean, and from my laptop I also have a direct VPN
connection
to the asterisk server my laptop being 192.168.241.10 and asterisk being
192.168.241.30

I do not understand several things:

1. asterisk seems to be telling sipml5 to send audio to it's public ip
addres, but * sends to 192.168.241.10
2. the asterisk output shows one way RTP flow. There's no sound from chrome.

I am trying to debug, but need some explanation about the SDP with respect
to WebRTC and ICE,
I hope someone can intersperse the output with comments?

Thanks,
Antonio

Below are the asterisk log, and the Javascript console output:

http://pastebin.com/dTFTrzg6
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Re: [asterisk-users] Need help troubleshooting Asterisk Auto dial out problem

[Jeremy Kister]

On 4/30/2014 7:24 PM, Jesse Thompson wrote:

impacted. However new files introduced into /var/spool/asterisk/outgoing/
folder get ignored. No messages spring up on asterisk -rvv console, nothing
shows up in the logs, the .call files just get snubbed. We're at a loss to


Are the new files being named uniquely ?

there are bugs (e.g., jira# 11291) that have to do with files having the 
same name.


my solution was to add .$$ on the end of the filename to ensure it was 
unique.



--

Jeremy Kister
http://jeremy.kister.net./


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Re: [asterisk-users] Need help troubleshooting Asterisk Auto dial out problem

[Jesse Thompson]

 Are the new files being named uniquely ?

 there are bugs (e.g., jira# 11291) that have to do with files having the
 same name.

 my solution was to add .$$ on the end of the filename to ensure it was
 unique.

 Yep, the files get a -MM-DD_HH:ii:ss- timestamp prefix in their names
before being mv'ed into the spool directory (same filesystem) and are never
realistically fired of more than once per second. Logic behind this was
that after they get moved automatically into the outgoing_done/ folder by
asterisk, we've got a rough log in the filenames of which alarms got
tripped at what times. :)

- - Jesse
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[asterisk-users] Need help troubleshooting Asterisk Auto dial out problem

[Jesse Thompson]

We've built an alert system at our company so that if our monitoring
software notices anything very bad happening, and we don't react to a text
message after a few minutes, then it will begin to call our telephones
directly. This seems to help a lot with staff who are asleep, or who might
not be near enough to the phone to hear text message notifications.

We do this by having a folder full of .call files, and programatically
moving a copy of the appropriate call file
into /var/spool/asterisk/outgoing/ when an outbound shout is required.

This has worked well for a couple of years, but recently we are
experiencing a problem where sometimes Asterisk will simply ignore this
folder. When we notice the logjam of ignored call files
in /var/spool/asterisk/outgoing/, then we restart the asterisk daemon and
things run smoothly again.

When wedged, the Asterisk daemon will function normally in every other
way that we use it. The PBX still functions, outbound manual calls still
function, inbound calls, voicemail, nothing else appears to be negatively
impacted. However new files introduced into /var/spool/asterisk/outgoing/
folder get ignored. No messages spring up on asterisk -rvv console, nothing
shows up in the logs, the .call files just get snubbed. We're at a loss to
determine what other debugging avenues may be available, and we have
googled for every applicable keyword we can think of including asterisk
auto dial and pbx_spool to no avail.

We are running:
Asterisk 1.8.13.1~dfsg1-3+deb7u3 built by pbuilder @ pungenday on a x86_64
running Linux on 2014-01-04 01:03:48 UTC
on Debian Linux stable (wheezy). It is possible that this symptom began a
few months ago after a system update (apt-get update; apt-get upgrade).

Any advise to further track down what is going on would be appreciated,
thank you!

- - Jesse Thompson
Webformix, Bend OR
www.webformix.com
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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

thanks a lot i will test and i will update you as soon as i have any
problem

2013/3/22 Asghar Mohammad asghar...@gmail.com

 your dialplan nothing to do with bandwidth it dial out to digium card what
 ever come in.
 1.
 if your providers calls come in via digium card and you want send out
 using sip or any other tech. then use context defined in group 1 for
 provider 1 and context defined in group 2 for provider 2.
 2.
 if your providers come in using sip just give him deferent ips, provider 1
 send to wimax ip and provider to FH.
 or explain if you are using other scenario.


 On Fri, Mar 22, 2013 at 7:14 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 yes i want to use the burden-sharing between Wimax and FH using a diguim
 cards


 2013/3/22 Asghar Mohammad asghar...@gmail.com

 hi,
 i think we miss understood you Question?
 you need round robin on tdm trunk or on 2 internet connections?
 what are you asking about   burden-sharing between Wimax and FH?


 On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 ok thank you so much i use dial(zap/r2) instead of g2 and it works
 without problem



 now my question i have 2 providers i use g1 for the first and g2 for
 the second



 if i understand i must use r1 instead of g1 for the first provider and
 r2 instead of g2 for the second provider in order to use the burden-sharing
 between Wimax and FH


 thanks and regards

 2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten =
 _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i
 use the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the
 leading slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice:
 +1-760-468-3867 PST
 Newline  Fax:
 +1-760-731-3000

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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

ok thank you so much i use dial(zap/r2) instead of g2 and it works without
problem



now my question i have 2 providers i use g1 for the first and g2 for the
second



if i understand i must use r1 instead of g1 for the first provider and r2
instead of g2 for the second provider in order to use the burden-sharing
between Wimax and FH


thanks and regards

2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the leading
 slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
 Newline  Fax:
 +1-760-731-3000

 --
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Re: [asterisk-users] Need help about round-robin

[Bharat Lalcheta]

Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:

 ok thank you so much i use dial(zap/r2) instead of g2 and it works without
 problem



 now my question i have 2 providers i use g1 for the first and g2 for the
 second



 if i understand i must use r1 instead of g1 for the first provider and r2
 instead of g2 for the second provider in order to use the burden-sharing
 between Wimax and FH


 thanks and regards

 2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i
 use the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the
 leading slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
 Newline  Fax:
 +1-760-731-3000

 --
 __**__**
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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

Hello bharat,

ok thank you so much for your help and support now i understand :)

2013/3/22 Bharat Lalcheta bharatlalch...@gmail.com

 Ya u r right. Value of 1 in r1 or g1 is group you mentioned in zapata.conf
 On Mar 22, 2013 8:54 PM, Salaheddine Elharit salah.elharit...@gmail.com
 wrote:

 ok thank you so much i use dial(zap/r2) instead of g2 and it works
 without problem



 now my question i have 2 providers i use g1 for the first and g2 for the
 second



 if i understand i must use r1 instead of g1 for the first provider and r2
 instead of g2 for the second provider in order to use the burden-sharing
 between Wimax and FH


 thanks and regards

 2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i
 use the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the
 leading slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
 Newline  Fax:
 +1-760-731-3000

 --
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Re: [asterisk-users] Need help about round-robin

[Asghar Mohammad]

hi,
i think we miss understood you Question?
you need round robin on tdm trunk or on 2 internet connections?
what are you asking about   burden-sharing between Wimax and FH?

On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit 
salah.elharit...@gmail.com wrote:

 ok thank you so much i use dial(zap/r2) instead of g2 and it works without
 problem



 now my question i have 2 providers i use g1 for the first and g2 for the
 second



 if i understand i must use r1 instead of g1 for the first provider and r2
 instead of g2 for the second provider in order to use the burden-sharing
 between Wimax and FH


 thanks and regards

 2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i
 use the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the
 leading slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

yes i want to use the burden-sharing between Wimax and FH using a diguim
cards

2013/3/22 Asghar Mohammad asghar...@gmail.com

 hi,
 i think we miss understood you Question?
 you need round robin on tdm trunk or on 2 internet connections?
 what are you asking about   burden-sharing between Wimax and FH?


 On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 ok thank you so much i use dial(zap/r2) instead of g2 and it works
 without problem



 now my question i have 2 providers i use g1 for the first and g2 for the
 second



 if i understand i must use r1 instead of g1 for the first provider and r2
 instead of g2 for the second provider in order to use the burden-sharing
 between Wimax and FH


 thanks and regards

 2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i
 use the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the
 leading slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users


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Re: [asterisk-users] Need help about round-robin

[Asghar Mohammad]

your dialplan nothing to do with bandwidth it dial out to digium card what
ever come in.
1.
if your providers calls come in via digium card and you want send out using
sip or any other tech. then use context defined in group 1 for provider 1
and context defined in group 2 for provider 2.
2.
if your providers come in using sip just give him deferent ips, provider 1
send to wimax ip and provider to FH.
or explain if you are using other scenario.

On Fri, Mar 22, 2013 at 7:14 PM, Salaheddine Elharit 
salah.elharit...@gmail.com wrote:

 yes i want to use the burden-sharing between Wimax and FH using a diguim
 cards


 2013/3/22 Asghar Mohammad asghar...@gmail.com

 hi,
 i think we miss understood you Question?
 you need round robin on tdm trunk or on 2 internet connections?
 what are you asking about   burden-sharing between Wimax and FH?


 On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 ok thank you so much i use dial(zap/r2) instead of g2 and it works
 without problem



 now my question i have 2 providers i use g1 for the first and g2 for the
 second



 if i understand i must use r1 instead of g1 for the first provider and
 r2 instead of g2 for the second provider in order to use the burden-sharing
 between Wimax and FH


 thanks and regards

 2013/3/21 Asghar Mohammad asghar...@gmail.com

 hi,

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup()

 Note r in Dial.
 you can use r for Ascending and R for Descending order

 On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i
 use the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the
 leading slash.

 --
 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:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   
 http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users


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[asterisk-users] Need help about round-robin

[Salaheddine Elharit]

hello list,

i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
old version with zapata.conf and zaptel.conf)

i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
want to active the round-robin for span 2 and 6) in order to activate the
WIMAX and FH

please see the configuration below and tell me if there is anything  wrong

question 2: what is difference between etc\zapataa.conf and
etc\asterisk\zapata.conf

i make this configuration just in etc\asterisk\zapata.conf i don't know if
i must do this configuration also in etc\zapata.conf

etc\asterisk\zapata.conf


[channels]
context=default
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
rxgain=0.0
txgain=0.0

group=1
switchtype=euroisdn
signalling=pri_cpe
callgroup=1
pickupgroup=1
immediate=no
channel = 1-15,17-31

group=2
callgroup=2
switchtype=qsig
signalling=pri_net
callerid=X(my callerID)
immediate=no
channel = 156-170
channel = 172-176
channel = 32-46
channel = 48-62


etc\zaptel.conf

# Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not
hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-15,17-31
dchan=16

# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED
span=2,2,0,ccs,hdb3
# termtype: te
bchan=32-46,48-62
dchan=47

# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
# span=3,3,0,ccs,hdb3
# termtype: te
# bchan=63-77,79-93
# dchan=78

# Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
# span=4,4,0,ccs,hdb3
# termtype: te
# bchan=94-108,110-124
# dchan=109

# Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1
#span=5,5,0,ccs,hdb3
# termtype: te
#bchan=125-139,141-155
#dchan=140

# Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2
span=6,6,0,ccs,hdb3
# termtype: te
bchan=156-170,172-186
dchan=171

# Global data

loadzone = us
defaultzone = us

thank you so much
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Re: [asterisk-users] Need help about round-robin

[Bharat Lalcheta]

What do you mean by roundrobin here
On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:

 hello list,

 i have installed 2 diguim cards in my server using asterisk 1.4 (i use the
 old version with zapata.conf and zaptel.conf)

 i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
 want to active the round-robin for span 2 and 6) in order to activate the
 WIMAX and FH

 please see the configuration below and tell me if there is anything  wrong

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf

 i make this configuration just in etc\asterisk\zapata.conf i don't know if
 i must do this configuration also in etc\zapata.conf

 etc\asterisk\zapata.conf


 [channels]
 context=default
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 rxgain=0.0
 txgain=0.0

 group=1
 switchtype=euroisdn
 signalling=pri_cpe
 callgroup=1
 pickupgroup=1
 immediate=no
 channel = 1-15,17-31

 group=2
 callgroup=2
 switchtype=qsig
 signalling=pri_net
 callerid=X(my callerID)
 immediate=no
 channel = 156-170
 channel = 172-176
 channel = 32-46
 channel = 48-62


 etc\zaptel.conf

 # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not
 hand edit
 # Zaptel Configuration File
 #
 # This file is parsed by the Zaptel Configurator, ztcfg
 #
 # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED
 span=1,1,0,ccs,hdb3
 # termtype: te
 bchan=1-15,17-31
 dchan=16

 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED
 span=2,2,0,ccs,hdb3
 # termtype: te
 bchan=32-46,48-62
 dchan=47

 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
 # span=3,3,0,ccs,hdb3
 # termtype: te
 # bchan=63-77,79-93
 # dchan=78

 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
 # span=4,4,0,ccs,hdb3
 # termtype: te
 # bchan=94-108,110-124
 # dchan=109

 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1
 #span=5,5,0,ccs,hdb3
 # termtype: te
 #bchan=125-139,141-155
 #dchan=140

 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2
 span=6,6,0,ccs,hdb3
 # termtype: te
 bchan=156-170,172-186
 dchan=171

 # Global data

 loadzone = us
 defaultzone = us

 thank you so much

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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

i mean the burden-sharing between Wimax and FH

2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 What do you mean by roundrobin here
 On Mar 21, 2013 8:27 PM, Salaheddine Elharit salah.elharit...@gmail.com
 wrote:

 hello list,

 i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i
 want to active the round-robin for span 2 and 6) in order to activate the
 WIMAX and FH

 please see the configuration below and tell me if there is anything  wrong

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf

 i make this configuration just in etc\asterisk\zapata.conf i don't know
 if i must do this configuration also in etc\zapata.conf

 etc\asterisk\zapata.conf


 [channels]
 context=default
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 rxgain=0.0
 txgain=0.0

 group=1
 switchtype=euroisdn
 signalling=pri_cpe
 callgroup=1
 pickupgroup=1
 immediate=no
 channel = 1-15,17-31

 group=2
 callgroup=2
 switchtype=qsig
 signalling=pri_net
 callerid=X(my callerID)
 immediate=no
 channel = 156-170
 channel = 172-176
 channel = 32-46
 channel = 48-62


 etc\zaptel.conf

 # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do
 not hand edit
 # Zaptel Configuration File
 #
 # This file is parsed by the Zaptel Configurator, ztcfg
 #
 # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED
 span=1,1,0,ccs,hdb3
 # termtype: te
 bchan=1-15,17-31
 dchan=16

 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED
 span=2,2,0,ccs,hdb3
  # termtype: te
 bchan=32-46,48-62
 dchan=47

 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
 # span=3,3,0,ccs,hdb3
 # termtype: te
 # bchan=63-77,79-93
 # dchan=78

 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
 # span=4,4,0,ccs,hdb3
 # termtype: te
 # bchan=94-108,110-124
 # dchan=109

 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1
 #span=5,5,0,ccs,hdb3
 # termtype: te
 #bchan=125-139,141-155
 #dchan=140

 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2
 span=6,6,0,ccs,hdb3
 # termtype: te
 bchan=156-170,172-186
 dchan=171

 # Global data

 loadzone = us
 defaultzone = us

 thank you so much

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Re: [asterisk-users] Need help about round-robin

[Bharat Lalcheta]

If u want to dial in round robin use Dial(zap/r2/2) . It dials using
channel in round robin
On Mar 21, 2013 9:37 PM, Salaheddine Elharit salah.elharit...@gmail.com
wrote:

 i mean the burden-sharing between Wimax and FH

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 What do you mean by roundrobin here
 On Mar 21, 2013 8:27 PM, Salaheddine Elharit 
 salah.elharit...@gmail.com wrote:

 hello list,

 i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 i want to use the span 1 for group 1 and span 2-span 6 for the group 2
 (i want to active the round-robin for span 2 and 6) in order to activate
 the WIMAX and FH

 please see the configuration below and tell me if there is anything
  wrong

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf

 i make this configuration just in etc\asterisk\zapata.conf i don't know
 if i must do this configuration also in etc\zapata.conf

 etc\asterisk\zapata.conf


 [channels]
 context=default
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 rxgain=0.0
 txgain=0.0

 group=1
 switchtype=euroisdn
 signalling=pri_cpe
 callgroup=1
 pickupgroup=1
 immediate=no
 channel = 1-15,17-31

 group=2
 callgroup=2
 switchtype=qsig
 signalling=pri_net
 callerid=X(my callerID)
 immediate=no
 channel = 156-170
 channel = 172-176
 channel = 32-46
 channel = 48-62


 etc\zaptel.conf

 # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do
 not hand edit
 # Zaptel Configuration File
 #
 # This file is parsed by the Zaptel Configurator, ztcfg
 #
 # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED
 span=1,1,0,ccs,hdb3
 # termtype: te
 bchan=1-15,17-31
 dchan=16

 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED
 span=2,2,0,ccs,hdb3
  # termtype: te
 bchan=32-46,48-62
 dchan=47

 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
 # span=3,3,0,ccs,hdb3
 # termtype: te
 # bchan=63-77,79-93
 # dchan=78

 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
 # span=4,4,0,ccs,hdb3
 # termtype: te
 # bchan=94-108,110-124
 # dchan=109

 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1
 #span=5,5,0,ccs,hdb3
 # termtype: te
 #bchan=125-139,141-155
 #dchan=140

 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2
 span=6,6,0,ccs,hdb3
 # termtype: te
 bchan=156-170,172-186
 dchan=171

 # Global data

 loadzone = us
 defaultzone = us

 thank you so much

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Re: [asterisk-users] Need help about round-robin

[Steve Edwards]

On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

i have installed 2 diguim cards in my server using asterisk 1.4 (i use 
the old version with zapata.conf and zaptel.conf)


question 2: what is difference between etc\zapataa.conf and 
etc\asterisk\zapata.conf


There is no /etc/zapata.conf.

The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

Note that the direction of the 'slash' is significant as is the leading 
slash.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help about round-robin

[Bharat Lalcheta]

File is ok there is no etc/zapata file.
On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the leading
 slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help about round-robin

[Salaheddine Elharit]

how can i use Dial(zap/r2/2)

below an exemple from my extensions.conf

exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup();

thanks and regards.

2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the leading
 slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help about round-robin

[Bharat Lalcheta]

Use r2 instead of g2 in dial

Dial(Zap/r2/${EXTEN}
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Re: [asterisk-users] Need help about round-robin

[Asghar Mohammad]

hi,

exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten =
_0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
exten = _0612.,n,Hangup()

Note r in Dial.
you can use r for Ascending and R for Descending order

On Thu, Mar 21, 2013 at 6:00 PM, Salaheddine Elharit 
salah.elharit...@gmail.com wrote:

 how can i use Dial(zap/r2/2)

 below an exemple from my extensions.conf

 exten = _0612.,1,Set(CALLERID(number)=520460587)
 exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
 exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
 exten =
 _0612.,n,Dial(Zap/g2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)
 exten = _0612.,n,Hangup();

 thanks and regards.

 2013/3/21 Bharat Lalcheta bharatlalch...@gmail.com

 File is ok there is no etc/zapata file.
 On Mar 21, 2013 9:42 PM, Steve Edwards asterisk@sedwards.com
 wrote:

 On Thu, 21 Mar 2013, Salaheddine Elharit wrote:

  i have installed 2 diguim cards in my server using asterisk 1.4 (i use
 the old version with zapata.conf and zaptel.conf)

 question 2: what is difference between etc\zapataa.conf and
 etc\asterisk\zapata.conf


 There is no /etc/zapata.conf.

 The 2 files are /etc/zaptel.conf and /etc/asterisk/zapata.conf.

 Note that the direction of the 'slash' is significant as is the leading
 slash.

 --
 Thanks in advance,
 --**--**
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867PST
 Newline  Fax:
 +1-760-731-3000

 --
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Re: [asterisk-users] Need help understanding CDR

[Leandro Dardini]

Top replying ...

In the CDR you have two fields, duration and billed. Duration is
the total time from Dial command to end of calls. It is the time the
Dial command is running. Billed is the time from when the other
party answered and the end of the call.

In your example, duration and billsec will differ for just a second,
the time from the Call Connected to asterisk and the Welcome
greeting starts.

Leandro

2013/3/18 RSCL Mumbai rscl.mum...@gmail.com:
 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.com
 wrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
 correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] Need help understanding CDR

[Bharat Lalcheta]

Answered means your call answered by answer application or by ivr or moh
kind of dialplan.

Here, call connected to asterisk means your calls starts ringing and its
start duration field counter.

As soon as you answer the call, i,e, start playing file or moh its start
counter of billsec field.

In below scenario, answer time and billed time both is same i.e 10 (wav) +
60 (agent). Also duration field in CDR is same i.e 60 or 61 sec as you
directly start playing greetings.

What i understand from below scenario and if i do not misunderstood, you
want time of call answered by Agent i.e. 60 sec time in you billsec.

But you are getting confused on answered word, it does not mean answer by
agent.

If you want agent talk time with your customer, then i think CDR is not
provided the same.

Regards,

Bharat Lalcheta
On Mon, Mar 18, 2013 at 10:59 AM, RSCL Mumbai rscl.mum...@gmail.com wrote:

 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
 correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] Need help understanding CDR

[Asghar Mohammad]

hi,

00:00 -- Call Connected to asterisk - duration start here
00:01 -- welcome greeting starts  billisec start here
00:11 -- welcome greeting ends (10 sec wav file)
00:12 -- Call enters queue and at the same time rings on first available
extension
00:15 -- Call is answered by an agent
01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec
--- both end here

duration = 01:15
bilsec = 01:14

duration start as soon as call arrived in asterisk.
bilsec start as soon as call answered.

exten s,1,Answer()  duration and bilsec start at same time because
you answered the call immidataly
exten s,n,Plaback(something)
exten s,n,Dial(agent)
exten s,n,Hangup  duration and billsec are same

exten s,1,Ringing(10) -- duration start here
exten s,n,Answer()  bilsec start here
exten s,n,Plaback(something)
exten s,n,Dial(agent)
exten s,n,Hangup  duration and billsec end here

so billsec is 10 seconds less then duration

hope this will help you.

On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.com wrote:

 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
 correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] Need help understanding CDR

[RSCL Mumbai]

Thank you every one.
Now I understand why I was confused.
I have always been using Asterisk in an Inbound environment.
Hence my thought were misaligned wrt answered  billed.
Now I understand. Thank you all!!

Is there anyway to capture the time for conversation, IVR, hold etc etc.
If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or any 3rd
party application, more suitable for an Inbound environment.

It would help a lot if I could capture fragmented distribution of time per
call -- time in IVR, Queue, Call etc.

Regards,
Sans









On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.comwrote:

 hi,

 00:00 -- Call Connected to asterisk - duration start here
 00:01 -- welcome greeting starts  billisec start here

 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec
 --- both end here

 duration = 01:15
 bilsec = 01:14

 duration start as soon as call arrived in asterisk.
 bilsec start as soon as call answered.

 exten s,1,Answer()  duration and bilsec start at same time
 because you answered the call immidataly
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec are same

 exten s,1,Ringing(10) -- duration start here
 exten s,n,Answer()  bilsec start here
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec end here

 so billsec is 10 seconds less then duration

 hope this will help you.

 On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.comwrote:

 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.comwrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2
 sec correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

 --
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Re: [asterisk-users] Need help understanding CDR

[Leandro Dardini]

You can add custom fields in the CDR, so your dialplan can store start
time, end time and duration whenever you like.

Just use something like the

Set(CDR(customfield)=100);

Leandro

2013/3/18 RSCL Mumbai rscl.mum...@gmail.com:
 Thank you every one.
 Now I understand why I was confused.
 I have always been using Asterisk in an Inbound environment.
 Hence my thought were misaligned wrt answered  billed.
 Now I understand. Thank you all!!

 Is there anyway to capture the time for conversation, IVR, hold etc etc.
 If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or any 3rd
 party application, more suitable for an Inbound environment.

 It would help a lot if I could capture fragmented distribution of time per
 call -- time in IVR, Queue, Call etc.

 Regards,
 Sans









 On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.com
 wrote:

 hi,

 00:00 -- Call Connected to asterisk - duration start here
 00:01 -- welcome greeting starts  billisec start here

 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec
 --- both end here

 duration = 01:15
 bilsec = 01:14

 duration start as soon as call arrived in asterisk.
 bilsec start as soon as call answered.

 exten s,1,Answer()  duration and bilsec start at same time
 because you answered the call immidataly
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec are same

 exten s,1,Ringing(10) -- duration start here
 exten s,n,Answer()  bilsec start here
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec end here

 so billsec is 10 seconds less then duration

 hope this will help you.

 On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.com
 wrote:

 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed
 time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.com
 wrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com
 wrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2
 sec correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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 _
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 _
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http

Re: [asterisk-users] Need help understanding CDR

[Asghar Mohammad]

hi,
try Asterisk manager or AGI.

On Mon, Mar 18, 2013 at 12:36 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:

 Thank you every one.
 Now I understand why I was confused.
 I have always been using Asterisk in an Inbound environment.
 Hence my thought were misaligned wrt answered  billed.
 Now I understand. Thank you all!!

 Is there anyway to capture the time for conversation, IVR, hold etc etc.
 If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or any
 3rd party application, more suitable for an Inbound environment.

 It would help a lot if I could capture fragmented distribution of time per
 call -- time in IVR, Queue, Call etc.

 Regards,
 Sans









 On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.comwrote:

 hi,

 00:00 -- Call Connected to asterisk - duration start here
 00:01 -- welcome greeting starts  billisec start here

 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec
 --- both end here

 duration = 01:15
 bilsec = 01:14

 duration start as soon as call arrived in asterisk.
 bilsec start as soon as call answered.

 exten s,1,Answer()  duration and bilsec start at same time
 because you answered the call immidataly
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec are same

 exten s,1,Ringing(10) -- duration start here
 exten s,n,Answer()  bilsec start here
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec end here

 so billsec is 10 seconds less then duration

 hope this will help you.

 On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai rscl.mum...@gmail.comwrote:

 I am using SIP.

 I am still a bit confused about answered  billed time.

 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first available
 extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

 In the given schematic what will be the Answered time and billed
 time.

 Thank you for the help in advance!!









 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad 
 asghar...@gmail.comwrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.comwrote:

 If you have analog FXO ports then the call is considered answered as
 soon as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2
 sec correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

 --
 _
 -- 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



 --
 _
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Re: [asterisk-users] Need help understanding CDR

[Ishfaq Malik]

If you use application Queue to pass the calls to the agents you will
have the advantage of having the queue log available which will give you
lots of detailed information.

Regards

Ish

On Mon, 2013-03-18 at 17:06 +0530, RSCL Mumbai wrote:
 Thank you every one.
 Now I understand why I was confused.
 I have always been using Asterisk in an Inbound environment.
 Hence my thought were misaligned wrt answered  billed.
 Now I understand. Thank you all!!
 
 Is there anyway to capture the time for conversation, IVR, hold etc
 etc.
 If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or
 any 3rd party application, more suitable for an Inbound environment.
 
 It would help a lot if I could capture fragmented distribution of time
 per call -- time in IVR, Queue, Call etc.
 
 Regards,
 Sans
 
 
 
 
 
 
 
 
 
 On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.com
 wrote:
 hi,
 
 
 00:00 -- Call Connected to asterisk - duration start here
 00:01 -- welcome greeting starts  billisec start here
 
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings on first
 available extension
 00:15 -- Call is answered by an agent
 
 01:15 -- Conversation over, Call disconnected -- agents spoke
 for 60 sec --- both end here
 
 
 duration = 01:15
 bilsec = 01:14
 
 
 duration start as soon as call arrived in asterisk.
 bilsec start as soon as call answered.
 
 
 exten s,1,Answer()  duration and bilsec start at same
 time because you answered the call immidataly
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec are same
 
 
 exten s,1,Ringing(10) -- duration start here
 exten s,n,Answer()  bilsec start here
 exten s,n,Plaback(something)
 exten s,n,Dial(agent)
 exten s,n,Hangup  duration and billsec end here
 
 
 so billsec is 10 seconds less then duration
 
 
 hope this will help you.
 
 On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai
 rscl.mum...@gmail.com wrote:
 I am using SIP.
 
 I am still a bit confused about answered  billed
 time.
 
 For example:
 00:00 -- Call Connected to asterisk
 00:01 -- welcome greeting starts
 00:11 -- welcome greeting ends (10 sec wav file)
 00:12 -- Call enters queue and at the same time rings
 on first available extension
 00:15 -- Call is answered by an agent
 01:15 -- Conversation over, Call disconnected --
 agents spoke for 60 sec.
 
 In the given schematic what will be the Answered
 time and billed time.
 
 Thank you for the help in advance!!
 
 
 
 
 
 
 
 
 
 On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad
 asghar...@gmail.com wrote:
 If you have analog FXO ports then the call is
 considered answered as soon as dialing is
 completed not always true if FXO configured
 properly it should not send back answered as
 soon as dialed.
 
 
 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling
 ewiel...@nyigc.com wrote:
 If you have analog FXO ports then the
 call is considered answered as soon as
 dialing is completed.   This does not
 apply to SIP, PRI, or other
 technologies which support far end
 answer detection.
 
 -Original Message-
 From:
 asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: [asterisk-users] Need help
 understanding CDR

[asterisk-users] Need help understanding CDR

[RSCL Mumbai]

Hi,

Attached is a sample CDR.

I need some help to understand the billsec column.
PS: the time value in billsec  duration is same.

With reference to the attached log, what does the 10 sec / 6 sec / 2
sec correspond
to:

(a) Time between call connection to asterisk and disconnection from
asterisk?
(b) Time after welcome greeting and before hangup -- the time the call rang
on the extension?
(c) Or any other scenario

Thank you in advance.

Best regards,
Sans
+-+--+--+-++-+---+---+-+-+-+
| calldate| clid | src  | dst   
  | dcontext   | dstchannel 
 | lastapp   | lastdata 
 | billsec | disposition | dnid|
+-+--+--+-++-+---+---+-+-+-+
| 2013-03-15 17:52:53 | 19170002018 8130006555   | 8130006555   | s 
  | app-announcement-4 |
 | Playback  | custom/Welcome,noanswer  
 |  10 | ANSWERED| 19170002018 |
| 2013-03-12 16:32:05 | 19170002018 2810007178   | 2810007178   | s 
  | app-announcement-4 |
 | Playback  | custom/Welcome,noanswer  
 |   6 | ANSWERED| 19170002018 |
| 2013-03-12 16:31:55 | 19170002018 2810007178   | 2810007178   | s 
  | app-announcement-4 |
 | Playback  | custom/Welcome,noanswer  
 |   2 | ANSWERED| 19170002018 |
+-+--+--+-++-+---+---+-+-+-+--
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Re: [asterisk-users] Need help understanding CDR

[Eric Wieling]

If you have analog FXO ports then the call is considered answered as soon as 
dialing is completed.   This does not apply to SIP, PRI, or other technologies 
which support far end answer detection.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Sunday, March 17, 2013 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Need help understanding CDR

Hi,

Attached is a sample CDR.

I need some help to understand the billsec column.
PS: the time value in billsec  duration is same.

With reference to the attached log, what does the 10 sec / 6 sec / 2 sec 
correspond to:

(a) Time between call connection to asterisk and disconnection from asterisk?
(b) Time after welcome greeting and before hangup -- the time the call rang on 
the extension?
(c) Or any other scenario

Thank you in advance.

Best regards,
Sans

--
_
-- 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
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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need help understanding CDR

[Asghar Mohammad]

hi,
billsec is time in seconds after call has answered, duration is total time
in seconds of call.
as your calls answered imidiatly your billsec and duration is almost same.

On Sun, Mar 17, 2013 at 5:14 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec 
 correspond
 to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] Need help understanding CDR

[Asghar Mohammad]

If you have analog FXO ports then the call is considered answered as soon
as dialing is completed not always true if FXO configured properly it
should not send back answered as soon as dialed.

On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
 correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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Re: [asterisk-users] Need help understanding CDR

[RSCL Mumbai]

I am using SIP.

I am still a bit confused about answered  billed time.

For example:
00:00 -- Call Connected to asterisk
00:01 -- welcome greeting starts
00:11 -- welcome greeting ends (10 sec wav file)
00:12 -- Call enters queue and at the same time rings on first available
extension
00:15 -- Call is answered by an agent
01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.

In the given schematic what will be the Answered time and billed time.

Thank you for the help in advance!!









On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed not always true if FXO configured properly it
 should not send back answered as soon as dialed.


 On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote:

 If you have analog FXO ports then the call is considered answered as soon
 as dialing is completed.   This does not apply to SIP, PRI, or other
 technologies which support far end answer detection.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
 Sent: Sunday, March 17, 2013 12:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Need help understanding CDR

 Hi,

 Attached is a sample CDR.

 I need some help to understand the billsec column.
 PS: the time value in billsec  duration is same.

 With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
 correspond to:

 (a) Time between call connection to asterisk and disconnection from
 asterisk?
 (b) Time after welcome greeting and before hangup -- the time the call
 rang on the extension?
 (c) Or any other scenario

 Thank you in advance.

 Best regards,
 Sans

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[asterisk-users] Need Help

[Joe Ruffolo]

Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 
2u server for our small business's phones system.

We are using some Polycom Soundpoint IP phones. The whole thing came crashing 
down over the Holidays and as of right now that's about

all we have working right now are the phones. The reason I joined this list is 
because I was hoping to get our external paging  intercom system back up and 
running

(it runs off of a sound card but cant get it all configured correctly) and to 
be honest I have no clue where to start. I've tried reading some online guides 
but nothing.


[MRKlogoblkemail]

Joe Ruffolo
Director of Operations
801 N State St Unit C
Elgin, Il. 60123
847-468-1700v
847-468-0717f
j...@mrkgroup.commailto:j...@mrkgroup.com
www.mrkgroupltd.comhttp://www.mrkgroupltd.com/

[mrk r2 logo]



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Re: [asterisk-users] Need Help

[Andrew Latham]

On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo j...@mrkgroup.com wrote:

 Hi all! In need of some serious help. We currently run Trixbox and Cent Os
 on a 2u server for our small business’s phones system.



 We are using some Polycom Soundpoint IP phones. The whole thing came
 crashing down over the Holidays and as of right now that’s about



 all we have working right now are the phones. The reason I joined this
 list is because I was hoping to get our external paging  intercom system
 back up and running



 (it runs off of a sound card but cant get it all configured correctly) and
 to be honest I have no clue where to start. I’ve tried reading some online
 guides but nothing.







 Joe Ruffolo

 Director of Operations

 801 N State St Unit C

 Elgin, Il. 60123

 847-468-1700v

 847-468-0717f

 j...@mrkgroup.com

 www.mrkgroupltd.com

Trixbox forums are here http://fonality.com/trixbox/ as this list will
not have the help you might need.


--
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Re: [asterisk-users] Need Help

[Patrick Lists]

On 01/17/2013 09:05 PM, Joe Ruffolo wrote:

Hi all! In need of some serious help. We currently run Trixbox and Cent
Os on a 2u server for our small business’s phones system.


Afaik Trixbox is no longer maintained and their forum are hardly active 
anymore so it may be a bit of a challenge to get support. If you really 
need a GUI like Trixbox then I suggest you have a look at Elastix which 
is very much alive and has a large community and professional services 
to help you out. See http://www.elastix.org/ Or have a look at Digium's 
Switchvox (payware). Whatever you do/choose, make sure that your box is 
secure if you open it up to the Internet.


Regards,
Patrick


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Re: [asterisk-users] Need Help

[Duncan Turnbull]

Hi Joe
On 18/01/2013, at 9:05 AM, Joe Ruffolo j...@mrkgroup.com wrote:

 Hi all! In need of some serious help. We currently run Trixbox and Cent Os on 
 a 2u server for our small business’s phones system.
  
 We are using some Polycom Soundpoint IP phones. The whole thing came crashing 
 down over the Holidays and as of right now that’s about
  
 all we have working right now are the phones. The reason I joined this list 
 is because I was hoping to get our external paging  intercom system back up 
 and running
  
It is useful to distinguish between whether asterisk is working (which this 
list can help with) or whether its trixbox functionality thats not working

When you say phones are working I am assuming you can call in and out and thus 
asterisk is working fine. The issue is configuration for other parts of the 
system

Trixbox is a lot based around the Freepbx web interface http://www.freepbx.org/ 
and if its configuration errors their forums or paid support can probably help 
you

However you said it came crashing down, which is concerning as that sounds like 
hardware issues, generally the platforms are pretty stable unless you get 
failure or corruption somewhere, often caused by power loss. 

I would think someone can provide (sell) you remote sys admin / asterisk 
support to check your hardware, but you probably need to ask for that 
specifically, or you can buy it via digium or freepbx sites

 (it runs off of a sound card but cant get it all configured correctly) and to 
 be honest I have no clue where to start. I’ve tried reading some online 
 guides but nothing.
  
My guess is your trixbox paging system sends messages out of your sound card 
into your physical paging system so two issues you may have are upset configs 
or hardware failure. 

Is this your only issue or are other things such as queues and ring groups 
suffering? Breaking it down into specific pieces will help others understand 
whether they can help or not

I use freepbx a lot but not trixbox or its version of trixbox and alterations, 
and I am not familiar with sound card setups anymore or Centos (ubuntu is my 
preference) but there are bound to be others who can help solve specific 
problems for you

It won't be particularly hard, but picking your priority issue and focussing on 
that is a good first step

Good luck

Cheers Duncan

  
 image001.jpg
  
 Joe Ruffolo
 Director of Operations
 801 N State St Unit C
 Elgin, Il. 60123
 847-468-1700v
 847-468-0717f
 j...@mrkgroup.com  
 www.mrkgroupltd.com
   
 image002.jpg
  
   
  
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[asterisk-users] Need help designing implementation

[larry lin]

 Hi,
 
 I'd like to replace my current VOIP provider with an Asterisk based 
 solution.  I have some ideas I want to run by the list to see if they 
 are possible, and get answers to a couple questions.

Take a look at gafachi (https://www.gafachi.com/), good voice quality and 
stable. 

Larry
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[asterisk-users] Need help designing implementation

[Dyweni - Asterisk-Users]

Hi,

I'd like to replace my current VOIP provider with an Asterisk based 
solution.  I have some ideas I want to run by the list to see if they 
are possible, and get answers to a couple questions.


I want to setup two Asterisk servers that are linked to each other:
- The first server would be my external (public) server and would 
live in a real data center.  The second server would be my internal 
(private) server and would live in my house.
- The external server would receive all incoming calls and handle the 
voice mail stuff.
- The internal server would run all the phones in my house (VOIP or 
Analog-via-FXS).  All outgoing calls would be routed out through the 
external server.


I also want to add the following additional functionality:
- If the external server looses connectivity to the internal server 
while a call is in progress, the external server should place the call 
on hold while it tries to reach us via our cell phones.  A message 
should be played informing the remote party that the connection had been 
lost and it is trying to re-establish it now.  If it can't reach us, it 
should inform the remote party that the connection could not be 
re-established and allow the remote party to leave some closing remarks 
on the voice mail system.


- If a call comes in and no one is at home to take the call (or if all 
lines at home are busy), it should ring all of our cell phones and 
whoever answers the call first gets the call.  If no one answers the 
call via the cell phones after 3 rings, it should route the call to the 
voice mail system.  I say 3 rings on the cell phone because I do not 
want the cell phone voice mail to take the call.


- I also would like the system to automatically route all calls 
directly to voice mail depending on the time of day (say 10PM to 8AM).  
I would like specify in a white list specific phone numbers that are 
allowed to ring through regardless of time of day (i.e. her parents, my 
parents).


- I would like the VOIP phones to turn on the voice mail waiting 
indicator light if the external server has new voice messages.



Is all of this possible?  If not, which part's are not (and how much 
work do you think would be needed to make those parts work)?



--
Thanks,
Dyweni



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Re: [asterisk-users] Need help designing implementation

[Chris Bagnall]

On 29/11/12 6:33 pm, Dyweni - Asterisk-Users wrote:

I want to setup two Asterisk servers that are linked to each other:
- The first server would be my external (public) server and would live
in a real data center.  The second server would be my internal
(private) server and would live in my house.
- The external server would receive all incoming calls and handle the
voice mail stuff.
- The internal server would run all the phones in my house (VOIP or
Analog-via-FXS).  All outgoing calls would be routed out through the
external server.


That all seems perfectly doable.


- If the external server looses connectivity to the internal server
while a call is in progress, the external server should place the call
on hold while it tries to reach us via our cell phones.  A message
should be played informing the remote party that the connection had been
lost and it is trying to re-establish it now.  If it can't reach us, it
should inform the remote party that the connection could not be
re-established and allow the remote party to leave some closing remarks
on the voice mail system.


I don't think that's doable without quite a lot of work - but others may 
be able to advise further.
To elaborate a little, it's easy to detect whether a route is usable 
when a call is placed, but detecting a call failure *during* the call is 
much more difficult.



- If a call comes in and no one is at home to take the call (or if all
lines at home are busy), it should ring all of our cell phones and
whoever answers the call first gets the call.  If no one answers the
call via the cell phones after 3 rings, it should route the call to the
voice mail system.  I say 3 rings on the cell phone because I do not
want the cell phone voice mail to take the call.


That's easy, though remember asterisk does things in seconds rather than 
rings.
You should also remember there's a delay in processing the call through 
the mobile networks before the phone actually starts ringing - in the 
UK that averages around 7 seconds between the call being sent to the 
mobile network from your server, and the phone ringing.



- I also would like the system to automatically route all calls directly
to voice mail depending on the time of day (say 10PM to 8AM). I would
like specify in a white list specific phone numbers that are allowed
to ring through regardless of time of day (i.e. her parents, my parents).


Shouldn't be difficult.


- I would like the VOIP phones to turn on the voice mail waiting
indicator light if the external server has new voice messages.


I believe this is doable in the newer versions of asterisk, but not the 
older versions. Again, someone else will hopefully chip in here, since 
our stuff is still running 1.4 :-)



Is all of this possible?  If not, which part's are not (and how much
work do you think would be needed to make those parts work)?


As is so often the case, (almost) anything is possible if you're 
prepared to spend time doing it. How much is worth doing depends on your 
time, and what else you might prefer to be doing with it...


FWIW, you might want to think about whether you actually need a separate 
asterisk box at home. In my experience, unless you have many dozens of 
extensions, you're almost better off (and certainly no worse off) 
connecting your SIP devices at home (assuming you're using SIP) directly 
back to the * server in the datacentre. One less box to maintain, and 
things like MWI will just work without having to play with the 
messaging interfaces.


Kind regards,

Chris
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Re: [asterisk-users] Need help for Aculab Prosody X PCI card installation and configuration with Asterisk

[Mitul Limbani]

AFAIK its a propreitary card from Aculab and wont work on Asterisk unless
you buy software or support or both from them.

My advice is to dump it n get a digium card in same or lesser cost which
you need to pay aculab.

Mitul Limbani
On Nov 12, 2012 1:23 PM, RAJNI VANZA rajniva...@gmail.com wrote:

 Hi All,

 I need to install and configuration of Aculab prosody X PCI card with
 Asterisk-1.8.9.1 on Centos-5.7 system.

 I will try for that but not success. so, please suggest me way to achieve
 it.

 Thanks in Advance.

 --
 Best Regards,
 Rajni Vanza

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[asterisk-users] Need help for Aculab Prosody X PCI card installation and configuration with Asterisk

[RAJNI VANZA]

Hi All,

I need to install and configuration of Aculab prosody X PCI card with
Asterisk-1.8.9.1 on Centos-5.7 system.

I will try for that but not success. so, please suggest me way to achieve
it.

Thanks in Advance.

-- 
Best Regards,
Rajni Vanza
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Re: [asterisk-users] Need help defining a stackexchange (i.e. stackoverflow) for telephony

[Simon P. Ditner]

It's nearly there now, just need a few more votes in order for it to 
trigger the next phase. Please take a moment to vote if you're 
interested:


  http://area51.stackexchange.com/proposals/12932/telephony/

On Mon, 9 May 2011, Simon P. Ditner wrote:

For those of that are fans of stackoverflow.com, and stackexchange.com, 
there's an effort to define a telephony stackexchange site. It's still in the 
definition phase. What it needs to move forwards is more votes on on/off 
topic questions, and perhaps some better questions to vote for or against.


If you're interested in helping out, or following the progress, visit:
http://area51.stackexchange.com/proposals/12932/telephony/

Cheers,
spd


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[asterisk-users] Need help defining a stackexchange (i.e. stackoverflow) for telephony

[Simon P. Ditner]

For those of that are fans of stackoverflow.com, and stackexchange.com, 
there's an effort to define a telephony stackexchange site. It's still in 
the definition phase. What it needs to move forwards is more votes on 
on/off topic questions, and perhaps some better questions to vote for or 
against.


If you're interested in helping out, or following the progress, visit:
http://area51.stackexchange.com/proposals/12932/telephony/

Cheers,
spd


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[asterisk-users] need help with IVR dialplan

[haloha]

Hi list


i setup successfull asterisk version 1.4 + opensips,
Opensips is the Registrar Server, Asterisk is the IVR server


the call flow
IP phone ---INVITE 1001 opensips - ASterisk INVITE
5001---opensips --- Busy|cancel|404..---asterisk---wait 10s to bye ---IP
phone (5000)

my case  is:
1/ IP phone(5000) ---Opensips
2/ IVR number : 1001
3/ IP Phone calls 1001 to opensips -- asterisk, ASterisk will play IVR
4/ IP phone press 1, asterisk will Dial(SIP/to_opensips/5001,20)
5/ there are some cases when asterisk send call back to opensips
 5.1/ if extension 5001 does not exist, opensips send 404 message back
to asterisk, then asterisk wait 10s to hangup the IP phone 5000
 5.2/ if extension 5001is real, opensips send ring ring back to
asterisk, then 5001 does not want to answer call
 5.2.1/  the call  is time out - then asterisk wait 10s to hangup the IP
phone 5000
 5.2.2/ the call is cancel by 5001 - asterisk receives cancel then
asterisk wait 10s to hangup the IP phone 5000
 5.2.3/ the Phone 5001 is busy - asterisk receive busy then asterisk
wait 10s to hangup the IP phone 5000


how to i force asterisk hangup IP PHONE 5000 when asterisk receives time
out|Cancel|busy from opensips

Thank you
Ha`
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Re: [asterisk-users] Need help with a pika warp asterisk appliance problem.

[Rod Boileau]

Timothy,

The Warp analog modules version A.1 are not compatible with the PADS 2.x
software base.
Replacing the modules to A.3 or newer and upgrading the software to PADS
2.1 or newer should resolve these issues.

Best Regards,
Rod Boileau
Manager, Customer Care
PIKA Technologies
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Re: [asterisk-users] Need help with a pika warp asterisk appliance problem.

[Kevin P. Fleming]

Timothy C Litwiller wrote:

 This upgrade says it has a special procedure and changes the layout of 
 the files it uses - so I am not sure I can downgrade again. I've asked 
 on the Pikawarp.org forum but so far no answer.  if it goes a few more 
 days I will have to try something - the people in that building have 
 been using the 1 phone and thier cell phones for a few days now - but I 
 really need to find a fix for this.

Since you bought the product from Pika, you should call Pika.

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Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] Need help with a pika warp asterisk appliance problem.

[Timothy C Litwiller]

I've tried the forums at pikawarp.org and it seems no one is there anymore.

Is there someone here that can help.

last week I decided it was time to upgrade the rom on the machine to try 
to get the newest freepbx.

so I followed the instructions to upgrade. I resetup the extensions and 
all the other setting like it was before.  I did this over the network

so I go onsite to double check that it is working correctly.  It isn't 
all the fxs lines that go into the fxs card have bad static - some so 
bad you can't even tell it is a dial tone.

the fxo card seems ok and the fxs line that is on the main board is ok.  
So I figured the rom didn't falsh correctly so I downloaded it again. 
This time to my local machine put it the files on the sd card - deleted 
the file that keeps it from loading over and over and let it flash 
again. -- with the same results.  Something it not setting up properly 
for the fxs card in the system - or has damaged it.
fxs/1 is on the main board - it works dial tone clear and sound during a 
call clear
fxs/2 can't tell these is dial tone
fxs/3 can hear dial tone thru bad static
fxs/4 can hear dial tone thru bad static - not quite as bad as 3
fxs/5 dial tone sounds ok - but if you call you start hearing clicks and 
pop and bursts of static.

Does anyone have any ideas?
This upgrade says it has a special procedure and changes the layout of 
the files it uses - so I am not sure I can downgrade again. I've asked 
on the Pikawarp.org forum but so far no answer.  if it goes a few more 
days I will have to try something - the people in that building have 
been using the 1 phone and thier cell phones for a few days now - but I 
really need to find a fix for this.

Thanks for listening and if you have any ideas I would sure like to hear 
them.



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Re: [asterisk-users] need help on setup rtp directly between 2 sipclients

[Juan E. Rodríguez]

Try setting canreinvite and nat to no for those extensions.

Saludos,
Juan E. Rodríguez


-Original Message-
From: Alyed al...@vivoxie.com
Date: Fri, 26 Mar 2010 10:56:50 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] need help on setup rtp directly between 2 sip
clients

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[asterisk-users] need help on setup rtp directly between 2 sip clients

[haloha]

Hi all

my asterisk server, 2 sip client softphones are the same LAN

asterisk ip address : 192.168.1.5
sip client 1 : 192.168.1.4
sip client 2 : 192.168.1.2

asterisk starts ok with sip

setup the sip.conf
[test]
type=friend
username=test
secret=1000
host=dynamic
context=cucku
directmedia=yes
directrtpsetup=yes

[1000]
type=friend
username=1000
secret=1000
host=dynamic
context=cucku
directmedia=yes
directrtpsetup=yes

when make call between 2 sip clients and see the debug in asterisk console
the problem is asterisk setup the inital call for media = asterisk IP
address, when all things done, asterisk does re-invite to setup the rtp
directly between 2 sip clients

is there any way to setup rtp directly between 2 sip clients, no need to go
through asterisk server

here is my debug log:
--- SIP read from UDP://192.168.1.4:18341 ---
INVITE sip:1...@192.168.1.5 sip%3a1...@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:18341
;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport
Max-Forwards: 70
Contact: sip:t...@192.168.1.4:18341
To: 1000sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=f543a140
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 261
v=0
o=- 8 2 IN IP4 192.168.1.4
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.4
t=0 0
m=audio 50420 RTP/AVP 107 0 8 101

--- Transmitting (no NAT) to 192.168.1.4:18341 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:18341
;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341
From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=f543a140
To: 1000sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26
Supported: replaces, timer
Contact: sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
Content-Length: 0

Reliably Transmitting (no NAT) to 192.168.1.2:34312:
INVITE sip:1...@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport
Max-Forwards: 70
From: Do Nguyen Ha sip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=as2886cf30
To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176
Contact: sip:t...@192.168.1.5 sip%3at...@192.168.1.5
Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.26
Date: Thu, 25 Mar 2010 12:15:05 GMT
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 1983608375 1983608375 IN IP4 192.168.1.5
s=Asterisk PBX 1.6.0.26
c=IN IP4 192.168.1.5
t=0 0
m=audio 17580 RTP/AVP 0 3 8 101

--- SIP read from UDP://192.168.1.2:34312 ---
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060
Contact: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176
To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176;tag=403c255c
From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=as2886cf30
Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


--- Transmitting (no NAT) to 192.168.1.4:18341 ---
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.4:18341
;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341
From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=f543a140
To: 1000sip:1...@192.168.1.5 sip%3a1...@192.168.1.5;tag=as0307d0b3
Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26
Supported: replaces, timer
ontact: sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
Content-Length: 0


--- SIP read from UDP://192.168.1.2:34312 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060
Contact: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176
To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176;tag=403c255c
From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=as2886cf30
Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
CSeq: 102 INVITE
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183
v=0
o=- 8 2 IN IP4 192.168.1.2
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.2
t=0 0
m=audio 53062 RTP/AVP 0 8 101

-
ACK sip:1...@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport
Max-Forwards: 70
From: Do Nguyen Ha sip:t...@192.168.1.5 sip%3at...@192.168.1.5
;tag=as2886cf30
To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176;tag=403c255c
Contact: sip:t...@192.168.1.5 sip%3at...@192.168.1.5
Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.26
Content-Length: 0

--- Reliably Transmitting (no NAT) to 192.168.1.4:18341 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:18341

Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

I guess to do what you want you need to dial directly between the phones.
Can't do it with xlite but you can with SJphones

Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:5060

Alyed



2010/3/26 haloha haloha...@gmail.com

 Hi all

 my asterisk server, 2 sip client softphones are the same LAN

 asterisk ip address : 192.168.1.5
 sip client 1 : 192.168.1.4
 sip client 2 : 192.168.1.2

 asterisk starts ok with sip

 setup the sip.conf
 [test]
 type=friend
 username=test
 secret=1000
 host=dynamic
 context=cucku
 directmedia=yes
 directrtpsetup=yes

 [1000]
 type=friend
 username=1000
 secret=1000
 host=dynamic
 context=cucku
 directmedia=yes
 directrtpsetup=yes

 when make call between 2 sip clients and see the debug in asterisk console
 the problem is asterisk setup the inital call for media = asterisk IP
 address, when all things done, asterisk does re-invite to setup the rtp
 directly between 2 sip clients

 is there any way to setup rtp directly between 2 sip clients, no need to go
 through asterisk server

 here is my debug log:
 --- SIP read from UDP://192.168.1.4:18341 ---
 INVITE sip:1...@192.168.1.5 sip%3a1...@192.168.1.5 SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.4:18341
 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport
 Max-Forwards: 70
 Contact: sip:t...@192.168.1.4:18341
 To: 1000sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
 From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=f543a140
 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
 CSeq: 2 INVITE
 Content-Type: application/sdp
 User-Agent: X-Lite release 1104o stamp 56125
 Content-Length: 261
 v=0
 o=- 8 2 IN IP4 192.168.1.4
 s=CounterPath X-Lite 3.0
 c=IN IP4 192.168.1.4
 t=0 0
 m=audio 50420 RTP/AVP 107 0 8 101

 --- Transmitting (no NAT) to 192.168.1.4:18341 ---
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 192.168.1.4:18341
 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341
 From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=f543a140
 To: 1000sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
 CSeq: 2 INVITE
 User-Agent: Asterisk PBX 1.6.0.26
 Supported: replaces, timer
 Contact: sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
 Content-Length: 0

 Reliably Transmitting (no NAT) to 192.168.1.2:34312:
 INVITE sip:1...@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport
 Max-Forwards: 70
 From: Do Nguyen Ha sip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=as2886cf30
 To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176
 Contact: sip:t...@192.168.1.5 sip%3at...@192.168.1.5
 Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.0.26
 Date: Thu, 25 Mar 2010 12:15:05 GMT
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 309
 v=0
 o=root 1983608375 1983608375 IN IP4 192.168.1.5
 s=Asterisk PBX 1.6.0.26
 c=IN IP4 192.168.1.5
 t=0 0
 m=audio 17580 RTP/AVP 0 3 8 101

 --- SIP read from UDP://192.168.1.2:34312 ---
 SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060
 Contact: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176
 To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176;tag=403c255c
 From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=as2886cf30
 Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
 CSeq: 102 INVITE
 User-Agent: X-Lite release 1104o stamp 56125
 Content-Length: 0


 --- Transmitting (no NAT) to 192.168.1.4:18341 ---
 SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP 192.168.1.4:18341
 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341
 From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=f543a140
 To: 1000sip:1...@192.168.1.5 sip%3a1...@192.168.1.5;tag=as0307d0b3
 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg.
 CSeq: 2 INVITE
 User-Agent: Asterisk PBX 1.6.0.26
 Supported: replaces, timer
 ontact: sip:1...@192.168.1.5 sip%3a1...@192.168.1.5
 Content-Length: 0


 --- SIP read from UDP://192.168.1.2:34312 ---
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060
 Contact: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176
 To: sip:1...@192.168.1.2:34312;rinstance=862211afcf483176;tag=403c255c
 From: Do Nguyen Hasip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=as2886cf30
 Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d...@192.168.1.5
 CSeq: 102 INVITE
 Content-Type: application/sdp
 User-Agent: X-Lite release 1104o stamp 56125
 Content-Length: 183
 v=0
 o=- 8 2 IN IP4 192.168.1.2
 s=CounterPath X-Lite 3.0
 c=IN IP4 192.168.1.2
 t=0 0
 m=audio 53062 RTP/AVP 0 8 101

 -
 ACK sip:1...@192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport
 Max-Forwards: 70
 From: Do Nguyen Ha sip:t...@192.168.1.5 sip%3at...@192.168.1.5
 ;tag=as2886cf30
 

Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[haloha]

Hi Alyed

xilte softphone work perfectly on other sip server(opensips server)

Don't remember the exact syntax but guess it's something like
sip:usern...@the.phones.ip:

 5060


you mean i config the extension.conf look like exten =
1000,1,Dial(SIP/1...@ip address:5060), is it right?

the problem i got here is the asterisk server to stay middle of  media
first, then redirect the media later, how to fix it,asterisk no need stay in
middle of media because all devices are in the same LAN

is there another hint

Thank you


On Fri, Mar 26, 2010 at 11:56 PM, Alyed al...@vivoxie.com wrote:

 I guess to do what you want you need to dial directly between the phones.
 Can't do it with xlite but you can with SJphones

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:5060

 Alyed



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Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

If your sofphones are registering to the asterisk, then asterisk needs to be
in the middle, otherwise there's no way your 101 sofpthone user can actually
know where (by where I mean which IP) is the 102 softphone user.

UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
well dialing directly to 102's IP.That's where Xlite doesn't work, but
SJphone does.

SJphone supports the advanced SIP URI syntax which for a user is:
sip:usern...@the.user.ip

Nevertheless.. if you are inside a LAN, why wouldn't you want those
calls to go through asterisk??? If you have collision problems I suggest you
fix them instead of asking everyone to call using SIP uri.

Alyed


2010/3/26 haloha haloha...@gmail.com

 Hi Alyed

 xilte softphone work perfectly on other sip server(opensips server)

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:

 5060


 you mean i config the extension.conf look like exten =
 1000,1,Dial(SIP/1...@ip address:5060), is it right?

 the problem i got here is the asterisk server to stay middle of  media
 first, then redirect the media later, how to fix it,asterisk no need stay in
 middle of media because all devices are in the same LAN

 is there another hint

 Thank you


 On Fri, Mar 26, 2010 at 11:56 PM, Alyed al...@vivoxie.com wrote:

 I guess to do what you want you need to dial directly between the phones.
 Can't do it with xlite but you can with SJphones

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:5060

 Alyed




 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[haloha]

Hi Alyed

so the asterisk is in middle in all version, right? thank you for your
explanation
all devices i mean are asterisk + softphones
my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE function that is why i ask the media
should be go directly between softphones and no need go through the asterisk


will check the SJphone feature, thank you for your suggestion


Thank you

On Sat, Mar 27, 2010 at 9:04 AM, Alyed al...@vivoxie.com wrote:

 If your sofphones are registering to the asterisk, then asterisk needs to
 be in the middle, otherwise there's no way your 101 sofpthone user can
 actually know where (by where I mean which IP) is the 102 softphone user.

 UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
 well dialing directly to 102's IP.That's where Xlite doesn't work, but
 SJphone does.

 SJphone supports the advanced SIP URI syntax which for a user is:
 sip:usern...@the.user.ip

 Nevertheless.. if you are inside a LAN, why wouldn't you want those
 calls to go through asterisk??? If you have collision problems I suggest you
 fix them instead of asking everyone to call using SIP uri.

 Alyed


 2010/3/26 haloha haloha...@gmail.com

 Hi Alyed


 xilte softphone work perfectly on other sip server(opensips server)

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:

 5060


 you mean i config the extension.conf look like exten =
 1000,1,Dial(SIP/1...@ip address:5060), is it right?

 the problem i got here is the asterisk server to stay middle of  media
 first, then redirect the media later, how to fix it,asterisk no need stay in
 middle of media because all devices are in the same LAN

 is there another hint

 Thank you


 On Fri, Mar 26, 2010 at 11:56 PM, Alyed al...@vivoxie.com wrote:

 I guess to do what you want you need to dial directly between the phones.
 Can't do it with xlite but you can with SJphones

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:5060

 Alyed




 --
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 New to Asterisk? Join us for a live introductory webinar every Thurs:
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   http://lists.digium.com/mailman/listinfo/asterisk-users



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 _
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Re: [asterisk-users] need help on setup rtp directly between 2 sip clients

[Alyed]

so the asterisk is in middle in all version, right? thank you for your
explanation
is the one whom everyone goes and says hey I'm 101 and live downstairs can
I play with you guys?

my goal is asterisk is on internet - WAN IP address and the softphones are
in NAT but the xlite supports the ICE function that is why i ask the media
should be go directly between softphones and no need go through the asterisk


I guess you still don't fully get it :)

The scenario you mention is similar (at least for the direct call thingy)
ICE doesn't mean you don't need to know where the callee is it just means it
will play a little with the SDP part of the SIP. Have a look at
http://www.voiptraversal.com/ice_methodology.htm to better understand what's
ICE about.

Alyed


2010/3/26 haloha haloha...@gmail.com

 Hi Alyed

 so the asterisk is in middle in all version, right? thank you for your
 explanation
 all devices i mean are asterisk + softphones
 my goal is asterisk is on internet - WAN IP address and the softphones are
 in NAT but the xlite supports the ICE function that is why i ask the media
 should be go directly between softphones and no need go through the asterisk


 will check the SJphone feature, thank you for your suggestion


 Thank you

 On Sat, Mar 27, 2010 at 9:04 AM, Alyed al...@vivoxie.com wrote:

 If your sofphones are registering to the asterisk, then asterisk needs to
 be in the middle, otherwise there's no way your 101 sofpthone user can
 actually know where (by where I mean which IP) is the 102 softphone user.

 UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How?
 well dialing directly to 102's IP.That's where Xlite doesn't work, but
 SJphone does.

 SJphone supports the advanced SIP URI syntax which for a user is:
 sip:usern...@the.user.ip

 Nevertheless.. if you are inside a LAN, why wouldn't you want those
 calls to go through asterisk??? If you have collision problems I suggest you
 fix them instead of asking everyone to call using SIP uri.

 Alyed


 2010/3/26 haloha haloha...@gmail.com

 Hi Alyed


 xilte softphone work perfectly on other sip server(opensips server)

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:

 5060


 you mean i config the extension.conf look like exten =
 1000,1,Dial(SIP/1...@ip address:5060), is it right?

 the problem i got here is the asterisk server to stay middle of  media
 first, then redirect the media later, how to fix it,asterisk no need stay in
 middle of media because all devices are in the same LAN

 is there another hint

 Thank you


 On Fri, Mar 26, 2010 at 11:56 PM, Alyed al...@vivoxie.com wrote:

 I guess to do what you want you need to dial directly between the
 phones. Can't do it with xlite but you can with SJphones

 Don't remember the exact syntax but guess it's something like
 sip:usern...@the.phones.ip:5060

 Alyed




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[asterisk-users] Need help with auto-forwarding virtual extensions (Asterisk 1.4/GUI 2.0)

[Cory Andrews]

I have a server running Asterisk 1.4 with Asterisk GUI 2.0. I have (2)
extensions that do not have physical IP phones registered to them, and
I want these extensions to auto-forward to external cellphone
numbers.

Anyone have an easy methodology for doing this, via the GUI or via conf
file adjustment?

Thanks in Advance

-- 
Cory J Andrews






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Re: [asterisk-users] Need help/suggestions for DialPlan

[Myles Wakeham]

Steve wrote:
 Patterns and wildcards are your friend.
 Maybe something like:
 [example]
  exten = _!,1,   verbose(1,[${CONTEXT}:${EXTEN}])
  exten = _!,2,   answer()
  exten = 1,3,goto(sales,s,1)
  exten = 2,3,goto(support,s,1)
  exten = 3,3,goto(customer-service,s,1)
  exten = 8,3,goto(directory,s,1)
  exten = _x,3,   hangup()
  exten = _6xx,3, goto(dial-by-extension,s,1)

First, thanks to everyone for the pointers.

This looks like it might fit my ACD requirements.  What I'm still 
confused about here is that it seems to get the extension in line 1, but 
answers in line 2?  Wouldn't it have to answer the call before it could 
get an extension?

And what I'm also not sure about is how this would work based on the 
number of digits entered.  Is it that by not specifying how many digits 
to get, it just handles processing 3 digits entered if the first digit 
is not 1,2,3 or 8?

Myles



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[asterisk-users] need help to setup a sip trunk between a Nortel CS1000 and asterisk

[James Hopwood]

I'm completely new to asterisk and while we have access to Nortel experts none 
of them know asterisk and since I'm the network guy I've been lumped with this.

This is what I'm trying to accomplish

We have a CS1000 that's sip capable.
I want to be able to connect an Asterisk box to the cs1000 via sip as an IVR, 
so I can pass a call off to asterisk, go through some IVR menus and have the 
asterisk box send the calls back to the CS1000 to the correct extensions.

Now I have a clean Asterisk install with nothing on it but a single sip phone 
so I could test it was working.

What I need to find out is how do I setup the sip trunks between the CS1000 and 
the Asterisk server. A how to would be great, or a pointer in the right 
direction as to what I need to learn / know to figure this out.

As I mentioned above I'm a complete noob
An will probably need a lot of help on this

Thanks in advance

James Hopwood






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[asterisk-users] Need help/suggestions for DialPlan

[Myles Wakeham]

I am revising our DialPlan strategy for our Asterisk system (1.4.2) and 
looking for some info on 'best practices' for this.  Here's what I'm 
trying to do:

I have an ACD menu that gives the caller the options as follows:

- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list

or enter the extension number at anytime to directly dial that extension.

I am setting up extensions to be 3 digits long, all starting with 6 (ie. 
601, 602, 603, etc.)

My current setup requires a single digit to be pressed before going to 
that context in the Dialplan.  Its all working fine with the exception 
of direct employee extension dialing.

What I have done is to create a context for '6'.  In other words, I wait 
for a single keypress at the main menu and if I get a 6, I then transfer 
to get the next 2 digits which represents the employee ID and then 
transfer to that person.

But what is happening is that the speed in which Asterisk is picking up 
the digits from the caller isn't fast enough for some reason, and is 
often missing the first digit (6) and then processing on some other 
digit that it picks up on.  I don't want to have to change our recorded 
audio menu since that was done professionally and was expensive, but 
just want it so that the incoming keys pressed are processed instantly.

I'm using a TDM400 from an analog line for the incoming menu, and then 
transferring to our internal LAN once the extension is chosen via SIP.

This is the context of my 'main menu':

; Calls during business hours
exten = s,1,Set(TIMEOUT(digit)=1)
exten = s,n,Wait(1)
exten = s,n,Background(1-MainMenu)
exten = s,n,WaitExten(3)
exten = s,n,Goto(ts_operator,s,1)

I have the context for extensions handled like this:

; User pressed 6 - start of extensions
[ts_extensions]
exten = s,1,Set(TIMEOUT(digit)=2)
exten = s,n,WaitExten(10)
exten = s,n,Hangup()
exten = t,1,Hangup()
exten = i,1,Goto(ts_start,s,1)

exten = 01,1,VoiceMail(6...@edgeneering)
exten = 01,n,Hangup()
exten = 02,1,VoiceMail(6...@edgeneering)
exten = 02,n,Hangup()
exten = 03,1,VoiceMail(6...@edgeneering)
exten = 03,n,Hangup()


(I'm dummying out the transfer to SIP phone parts with VoiceMails here 
for demo purposes)

What am I doing wrong that is stopping correct identification of the 
digits entered by the caller?

Thanks in advance for any assistance.

Myles


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Re: [asterisk-users] Need help/suggestions for DialPlan

[meetmecall]

If you add option 6 to the menu for the first position and use the  
read command for the 2 last position and use a second line that looks  
something like:

exten 6,n,Dial(SIP,6${ENTERED_NUMBER},20,t)


it should work.

The {ENTERED_NUMBER} should be the variable filled with the read  
command.

Don't forget to add some lines to catch wrong numbers and time outs

Hope this helps :-)

Erik


On 9 dec 2009, at 18:53, Myles Wakeham wrote:

 I am revising our DialPlan strategy for our Asterisk system (1.4.2)  
 and
 looking for some info on 'best practices' for this.  Here's what I'm
 trying to do:

 I have an ACD menu that gives the caller the options as follows:

 - Press 1 for sales
 - Press 2 for support
 - Press 3 for customer service
 - Press 8 for a 'Dial by Name' list

 or enter the extension number at anytime to directly dial that  
 extension.

 I am setting up extensions to be 3 digits long, all starting with 6  
 (ie.
 601, 602, 603, etc.)

 My current setup requires a single digit to be pressed before going to
 that context in the Dialplan.  Its all working fine with the exception
 of direct employee extension dialing.

 What I have done is to create a context for '6'.  In other words, I  
 wait
 for a single keypress at the main menu and if I get a 6, I then  
 transfer
 to get the next 2 digits which represents the employee ID and then
 transfer to that person.

 But what is happening is that the speed in which Asterisk is picking  
 up
 the digits from the caller isn't fast enough for some reason, and is
 often missing the first digit (6) and then processing on some other
 digit that it picks up on.  I don't want to have to change our  
 recorded
 audio menu since that was done professionally and was expensive, but
 just want it so that the incoming keys pressed are processed  
 instantly.

 I'm using a TDM400 from an analog line for the incoming menu, and then
 transferring to our internal LAN once the extension is chosen via SIP.

 This is the context of my 'main menu':

 ; Calls during business hours
 exten = s,1,Set(TIMEOUT(digit)=1)
 exten = s,n,Wait(1)
 exten = s,n,Background(1-MainMenu)
 exten = s,n,WaitExten(3)
 exten = s,n,Goto(ts_operator,s,1)

 I have the context for extensions handled like this:

 ; User pressed 6 - start of extensions
 [ts_extensions]
 exten = s,1,Set(TIMEOUT(digit)=2)
 exten = s,n,WaitExten(10)
 exten = s,n,Hangup()
 exten = t,1,Hangup()
 exten = i,1,Goto(ts_start,s,1)

 exten = 01,1,VoiceMail(6...@edgeneering)
 exten = 01,n,Hangup()
 exten = 02,1,VoiceMail(6...@edgeneering)
 exten = 02,n,Hangup()
 exten = 03,1,VoiceMail(6...@edgeneering)
 exten = 03,n,Hangup()


 (I'm dummying out the transfer to SIP phone parts with VoiceMails here
 for demo purposes)

 What am I doing wrong that is stopping correct identification of the
 digits entered by the caller?

 Thanks in advance for any assistance.

 Myles


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Re: [asterisk-users] Need help/suggestions for DialPlan

[Jarrod Lash]

Have you tried...

exten = 6XX,1,Dial(SIP,${EXTEN},20,t)


-- 
Jarrod Lash, jar...@fed-com.com
Federated Communications, LLC.
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368


On Wed, Dec 9, 2009 at 12:53 PM, Myles Wakeham my...@techsol.org wrote:
 I am revising our DialPlan strategy for our Asterisk system (1.4.2) and
 looking for some info on 'best practices' for this.  Here's what I'm
 trying to do:

 I have an ACD menu that gives the caller the options as follows:

 - Press 1 for sales
 - Press 2 for support
 - Press 3 for customer service
 - Press 8 for a 'Dial by Name' list

 or enter the extension number at anytime to directly dial that extension.

 I am setting up extensions to be 3 digits long, all starting with 6 (ie.
 601, 602, 603, etc.)

 My current setup requires a single digit to be pressed before going to
 that context in the Dialplan.  Its all working fine with the exception
 of direct employee extension dialing.

 What I have done is to create a context for '6'.  In other words, I wait
 for a single keypress at the main menu and if I get a 6, I then transfer
 to get the next 2 digits which represents the employee ID and then
 transfer to that person.

 But what is happening is that the speed in which Asterisk is picking up
 the digits from the caller isn't fast enough for some reason, and is
 often missing the first digit (6) and then processing on some other
 digit that it picks up on.  I don't want to have to change our recorded
 audio menu since that was done professionally and was expensive, but
 just want it so that the incoming keys pressed are processed instantly.

 I'm using a TDM400 from an analog line for the incoming menu, and then
 transferring to our internal LAN once the extension is chosen via SIP.

 This is the context of my 'main menu':

 ; Calls during business hours
 exten = s,1,Set(TIMEOUT(digit)=1)
 exten = s,n,Wait(1)
 exten = s,n,Background(1-MainMenu)
 exten = s,n,WaitExten(3)
 exten = s,n,Goto(ts_operator,s,1)

 I have the context for extensions handled like this:

 ; User pressed 6 - start of extensions
 [ts_extensions]
 exten = s,1,Set(TIMEOUT(digit)=2)
 exten = s,n,WaitExten(10)
 exten = s,n,Hangup()
 exten = t,1,Hangup()
 exten = i,1,Goto(ts_start,s,1)

 exten = 01,1,VoiceMail(6...@edgeneering)
 exten = 01,n,Hangup()
 exten = 02,1,VoiceMail(6...@edgeneering)
 exten = 02,n,Hangup()
 exten = 03,1,VoiceMail(6...@edgeneering)
 exten = 03,n,Hangup()


 (I'm dummying out the transfer to SIP phone parts with VoiceMails here
 for demo purposes)

 What am I doing wrong that is stopping correct identification of the
 digits entered by the caller?

 Thanks in advance for any assistance.

 Myles


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Re: [asterisk-users] Need help/suggestions for DialPlan

[Steve Edwards]

On Wed, 9 Dec 2009, Myles Wakeham wrote:

 I have an ACD menu that gives the caller the options as follows:

 - Press 1 for sales
 - Press 2 for support
 - Press 3 for customer service
 - Press 8 for a 'Dial by Name' list

 or enter the extension number at anytime to directly dial that extension.

 I am setting up extensions to be 3 digits long, all starting with 6 (ie.
 601, 602, 603, etc.)

Patterns and wildcards are your friend.

Maybe something like:

[example]
 exten = _!,1,   verbose(1,[${CONTEXT}:${EXTEN}])
 exten = _!,2,   answer()
 exten = 1,3,goto(sales,s,1)
 exten = 2,3,goto(support,s,1)
 exten = 3,3,goto(customer-service,s,1)
 exten = 8,3,goto(directory,s,1)
 exten = _x,3,   hangup()
 exten = _6xx,3, goto(dial-by-extension,s,1)

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help/suggestions for DialPlan

[Alec Davis]

Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.

Not sure of what release you're testing on.

See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad... announcement 
 
Alec Davis

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need help/suggestions for DialPlan

On Wed, 9 Dec 2009, Myles Wakeham wrote:

 I have an ACD menu that gives the caller the options as follows:

 - Press 1 for sales
 - Press 2 for support
 - Press 3 for customer service
 - Press 8 for a 'Dial by Name' list

 or enter the extension number at anytime to directly dial that extension.

 I am setting up extensions to be 3 digits long, all starting with 6 (ie.
 601, 602, 603, etc.)

Patterns and wildcards are your friend.

Maybe something like:

[example]
 exten = _!,1,   verbose(1,[${CONTEXT}:${EXTEN}])
 exten = _!,2,   answer()
 exten = 1,3,goto(sales,s,1)
 exten = 2,3,goto(support,s,1)
 exten = 3,3,goto(customer-service,s,1)
 exten = 8,3,goto(directory,s,1)
 exten = _x,3,   hangup()
 exten = _6xx,3, goto(dial-by-extension,s,1)

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help/suggestions for DialPlan

[Alec Davis]

Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.

Not sure of what release you're testing on.

See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad... announcement 
 
Alec Davis

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need help/suggestions for DialPlan

On Wed, 9 Dec 2009, Myles Wakeham wrote:

 I have an ACD menu that gives the caller the options as follows:

 - Press 1 for sales
 - Press 2 for support
 - Press 3 for customer service
 - Press 8 for a 'Dial by Name' list

 or enter the extension number at anytime to directly dial that extension.

 I am setting up extensions to be 3 digits long, all starting with 6 (ie.
 601, 602, 603, etc.)

Patterns and wildcards are your friend.

Maybe something like:

[example]
 exten = _!,1,   verbose(1,[${CONTEXT}:${EXTEN}])
 exten = _!,2,   answer()
 exten = 1,3,goto(sales,s,1)
 exten = 2,3,goto(support,s,1)
 exten = 3,3,goto(customer-service,s,1)
 exten = 8,3,goto(directory,s,1)
 exten = _x,3,   hangup()
 exten = _6xx,3, goto(dial-by-extension,s,1)

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need help/suggestions for DialPlan

[Alec Davis]

Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.

Not sure of what release you're testing on.

See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad... announcement 
 
Alec Davis

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need help/suggestions for DialPlan

On Wed, 9 Dec 2009, Myles Wakeham wrote:

 I have an ACD menu that gives the caller the options as follows:

 - Press 1 for sales
 - Press 2 for support
 - Press 3 for customer service
 - Press 8 for a 'Dial by Name' list

 or enter the extension number at anytime to directly dial that extension.

 I am setting up extensions to be 3 digits long, all starting with 6 (ie.
 601, 602, 603, etc.)

Patterns and wildcards are your friend.

Maybe something like:

[example]
 exten = _!,1,   verbose(1,[${CONTEXT}:${EXTEN}])
 exten = _!,2,   answer()
 exten = 1,3,goto(sales,s,1)
 exten = 2,3,goto(support,s,1)
 exten = 3,3,goto(customer-service,s,1)
 exten = 8,3,goto(directory,s,1)
 exten = _x,3,   hangup()
 exten = _6xx,3, goto(dial-by-extension,s,1)

--
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] Need help with this conf

[B.Masoud @ SH]

Hello, I would appreciate if someone can give some help on what I want:

 

When someone call my box (from outside), to a certain ZAP port, it will put
him on hold, and immediately the box calls to outside SIP trunk to a
preconfigured certain number, then when the other party picks up the phone,
both calls connected and CDR starts counting.

 

Any idea?

 

Thanks.

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[asterisk-users] need help debug asterisk-1.6 sip connection

[Joseph]

I have a DID but for some reason is not working in asterisk-1.6
The same sip connection in asterisk-1.4 is working OK, but it doesn't work with 
asterisk-1.6

Here is my sip.conf section:
...
[actio-out]
type=friend
secret=password
user=48746612254
username=48746612254
fromuser=48746612254
authname=48746612254
callerpage=48746612254
fromdomain=sip.actio.pl
host=sip.actio.pl
insecure=very
nat=yes
qualify=yes
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
context=internal
canreinvite=no

Here is relevant section from asterisk-1.6 (failed connection) and asterisk-1.4 
(working connection)

== start asterisk-1.6 (not working) ==

-
--- (17 headers 18 lines) ---
   == Using SIP RTP CoS mark 5
Sending to 81.15.150.20 : 5060 (no NAT)
Using INVITE request as basis request - 
ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o
Found peer 'actio-out' for '17804791270' from 81.15.150.20:5060

--- Reliably Transmitting (NAT) to 81.15.150.20:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
81.15.150.20;branch=z9hG4bK4c28.d397a70c58c5c983c7d85bb171d8e3b2.0;received=81.15.150.20
Via: SIP/2.0/UDP 
81.15.150.20:5061;branch=z9hG4bKba07785b184a5f79266bde33dccc8212;rport=5061
From: sip:17804791...@81.15.150.20;tag=26a9eb26114a01c9f4d1f64b72cc1d9e
To: sip:48746612...@81.15.150.20;tag=as52ab0bbb
Call-ID: ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o
CSeq: 200 INVITE
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0da18b05
Content-Length: 0



Scheduling destruction of SIP dialog 
'ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o' in 13632 ms (Method: 
INVITE)
syscon2*CLI
--- SIP read from UDP://81.15.150.20:5060 ---
ACK sip:s...@68.148.245.78:61454 SIP/2.0
Via: SIP/2.0/UDP 
81.15.150.20;branch=z9hG4bK4c28.d397a70c58c5c983c7d85bb171d8e3b2.0
From: sip:17804791...@81.15.150.20;tag=26a9eb26114a01c9f4d1f64b72cc1d9e
Call-ID: ffc94f46-c5d211de-9310e4a5-81fb2...@82.177.2.12~1o
To: sip:48746612...@81.15.150.20;tag=as52ab0bbb
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


-
--- (8 headers 0 lines) ---
syscon2*CLI
--- SIP read from UDP://81.15.150.20:5060 ---

= end asterisk-1.6 (not working) =



== start asterisk-1.4 (working) ==
-
--- (17 headers 18 lines) ---
Sending to 81.15.150.20 : 5060 (no NAT)
Using INVITE request as basis request - 
f203cdef-c5d411de-932ae4a5-81fb2...@82.177.2.12~1o
Found peer 'actio-out'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 2
Found RTP audio format 100
Peer audio RTP is at port 81.15.150.20:46648
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G723 for ID 4
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 99
Found audio description format G726-32 for ID 2
Found unknown media description format X-NSE for ID 100
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x90f 
(g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
Peer audio RTP is at port 81.15.150.20:46648
Looking for s in from_poland (domain 68.148.245.78)
list_route: hop: sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr
syscon4*CLI
--- Transmitting (NAT) to 81.15.150.20:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
81.15.150.20;branch=z9hG4bK5978.8397fd91b29a224fb6158a2eb64d4489.0;received=81.15.150.20
Via: SIP/2.0/UDP 
81.15.150.20:5061;branch=z9hG4bKa185fc54438defa99101bdc43db8e8c7;rport=5061
Record-Route: sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr
From: sip:17804791...@81.15.150.20;tag=1f5a641fc6ffb42064d4123781f0e7bb
To: sip:48746612...@81.15.150.20
Call-ID: f203cdef-c5d411de-932ae4a5-81fb2...@82.177.2.12~1o
CSeq: 200 INVITE
User-Agent: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:s...@10.0.0.109
Content-Length: 0



 -- Executing [...@from_poland:1] Answer(SIP/48746612254-00789120, ) in 
new stack
Audio is at 10.0.0.109 port 13414
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

--- Reliably Transmitting (NAT) to 81.15.150.20:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
81.15.150.20;branch=z9hG4bK5978.8397fd91b29a224fb6158a2eb64d4489.0;received=81.15.150.20
Via: SIP/2.0/UDP 
81.15.150.20:5061;branch=z9hG4bKa185fc54438defa99101bdc43db8e8c7;rport=5061
Record-Route: sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr

[asterisk-users] Need Help

[kiran.re...@mpowerglobal.in]

Hi list,

I am new to asterisk. I need help for installing and configure Asterisk 
IVR,OBD,IBD Server.


We have a PRI line,I need to know what are the system requirements and 
hardware requirement for Asterisk *IVR*,*OBD*(Outbound 
dialer),*IBD*(Inbound dialer).


Thanks and Regards,
Kiran Reddy
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[asterisk-users] Need Help

[kiran.re...@mpowerglobal.in]

Hi list,

I am new to asterisk. I need help for installing and configure Asterisk 
IVR,OBD,IBD Server.


We have a PRI line,I need to know what are the system requirements and 
hardware requirement for Asterisk *IVR*,*OBD*(Outbound 
dialer),*IBD*(Inbound dialer).


Thanks and Regards,
Kiran Reddy
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[asterisk-users] Need Help

[kiran.re...@mpowerglobal.in]

Hi list,

I am new to asterisk. I need help for installing and configure Asterisk 
IVR,OBD,IBD Server.


We have a PRI line,I need to know what are the system requirements and 
hardware requirement for Asterisk *IVR*,*OBD*(Outbound 
dialer),*IBD*(Inbound dialer).


Thanks and Regards,
Kiran Reddy
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[asterisk-users] Need Help

[kiran.re...@mpowerglobal.in]

Hi list,

I am new to asterisk. I need help for installing and configure Asterisk 
IVR,OBD,IBD Server.


We have a PRI line,I need to know what are the system requirements and 
hardware requirement for Asterisk *IVR*,*OBD*(Outbound 
dialer),*IBD*(Inbound dialer).


Thanks and Regards,
Kiran Reddy
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Re: [asterisk-users] Need Help

[Steve Edwards]

On Wed, 21 Oct 2009, kiran.re...@mpowerglobal.in wrote:

 I am new to asterisk. I need help for installing and configure Asterisk 
 IVR,OBD,IBD Server.

4 posts in 3 hours?

1) Don't repost, you just annoy people that may have helped you.

2) Ask specific questions, not I know nothing, please tell me 
everything.

3) Use a meaningful question. You may attract the interest of someone who 
can help you.

4) Help yourself. Use google. Read TFOT.pdf.

-- 
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] need help with card for IVR

[Roshan Singh]

Hi,
I have to implement asterisk to get a IVR system so that simultaneously 4-8
users can call and redirect them to other phones preferably GSM mobiles.
I will use PSTN lines. Also I need to know if these calls can be redirected
some how to GSM mobiles.

I do not have any pre-requisite in this field so please consider me as a
novice in your reply.

I need to know the cards as well as their prices.
Are intel dailogic cards supported by asterisk?

-- 
Roshan Kumar Singh
http://roshansingh.wordpress.com
http://sourceforge.net/projects/gscribble
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[asterisk-users] Need help - CDR MySQL

[Cyprus VoIP]

Hello all,

I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but 
without success.

Is there a proper online manual that describes all the steps to follow 
and debugging/monitoring information?

When I type in the CLI module show, cdr_addon_mysql.so is not listed, 
although in modules.conf, I added the line load = cdr_addon_mysql.so. 
I also tried preload, but it didn't change anything.

I also checked res_mysql.conf and cdr_mysql.conf, and the entire 
necessary data for the mysql server is there.

In cdr_manager.conf and cdr.conf, I set enabled = yes in [general].

I would appreciate any help I can get at this point, as I'm clueless as 
to what can be wrong.

Thanks.



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Re: [asterisk-users] Need help - CDR MySQL

[Tilghman Lesher]

On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
 Hello all,

 I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
 without success.

 Is there a proper online manual that describes all the steps to follow
 and debugging/monitoring information?

 When I type in the CLI module show, cdr_addon_mysql.so is not listed,
 although in modules.conf, I added the line load = cdr_addon_mysql.so.
 I also tried preload, but it didn't change anything.

 I also checked res_mysql.conf and cdr_mysql.conf, and the entire
 necessary data for the mysql server is there.

 In cdr_manager.conf and cdr.conf, I set enabled = yes in [general].

 I would appreciate any help I can get at this point, as I'm clueless as
 to what can be wrong.

Type:  'module load cdr_addon_mysql.so' and correct any errors you see.

-- 
Tilghman  Teryl
with Peter, Cottontail, Midnight, Thumper,  Johnny (bunnies)
and Harry, BB,  George (dogs)

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Re: [asterisk-users] Need help - CDR MySQL

[Cyprus VoIP]

Thanks. I found out that the module didn't load:

[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module: 
Error loading module 'cdr_addon_mysql.so': 
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared object 
file: No such file or directory

When I checked, I saw that it doesn't exist. It seems that when I 
installed the addons, I didn't realize that there were some issues to 
resolve first, as in the menuselect, I see that both app_addon_sql_mysql 
and cdr_addon_mysql have dependencies problems.

What should I do to resolve that?

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59

 On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
 Hello all,

 I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
 without success.

 Is there a proper online manual that describes all the steps to follow
 and debugging/monitoring information?

 When I type in the CLI module show, cdr_addon_mysql.so is not listed,
 although in modules.conf, I added the line load = cdr_addon_mysql.so.
 I also tried preload, but it didn't change anything.

 I also checked res_mysql.conf and cdr_mysql.conf, and the entire
 necessary data for the mysql server is there.

 In cdr_manager.conf and cdr.conf, I set enabled = yes in [general].

 I would appreciate any help I can get at this point, as I'm clueless as
 to what can be wrong.
 
 Type:  'module load cdr_addon_mysql.so' and correct any errors you see.
 

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Re: [asterisk-users] Need help - CDR MySQL

[Pascal Bruno]

You have to fix the dependency issues, which means install the stuff  
you are missing that cdrmysql depends on so u can recompile it.

Sent from my iPod

On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:

 Thanks. I found out that the module didn't load:

 [Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
 Error loading module 'cdr_addon_mysql.so':
 /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
 object
 file: No such file or directory

 When I checked, I saw that it doesn't exist. It seems that when I
 installed the addons, I didn't realize that there were some issues to
 resolve first, as in the menuselect, I see that both  
 app_addon_sql_mysql
 and cdr_addon_mysql have dependencies problems.

 What should I do to resolve that?

 Thanks.

  Original Message  
 Subject: Re: [asterisk-users] Need help - CDR MySQL
 From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sunday, 30 August, 2009 17:17:59

 On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
 Hello all,

 I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
 but
 without success.

 Is there a proper online manual that describes all the steps to  
 follow
 and debugging/monitoring information?

 When I type in the CLI module show, cdr_addon_mysql.so is not  
 listed,
 although in modules.conf, I added the line load =  
 cdr_addon_mysql.so.
 I also tried preload, but it didn't change anything.

 I also checked res_mysql.conf and cdr_mysql.conf, and the entire
 necessary data for the mysql server is there.

 In cdr_manager.conf and cdr.conf, I set enabled = yes in  
 [general].

 I would appreciate any help I can get at this point, as I'm  
 clueless as
 to what can be wrong.

 Type:  'module load cdr_addon_mysql.so' and correct any errors you  
 see.


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Re: [asterisk-users] Need help - CDR MySQL

[Cyprus VoIP]

I think that the missing component is mysqlclient, but when i yum 
update mysql, it does nothing.

Anyone know how to download the RPM? I'm using CentOS 5.3.

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:28:35

 You have to fix the dependency issues, which means install the stuff  
 you are missing that cdrmysql depends on so u can recompile it.
 
 Sent from my iPod
 
 On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:
 
 Thanks. I found out that the module didn't load:

 [Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
 Error loading module 'cdr_addon_mysql.so':
 /usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
 object
 file: No such file or directory

 When I checked, I saw that it doesn't exist. It seems that when I
 installed the addons, I didn't realize that there were some issues to
 resolve first, as in the menuselect, I see that both  
 app_addon_sql_mysql
 and cdr_addon_mysql have dependencies problems.

 What should I do to resolve that?

 Thanks.

  Original Message  
 Subject: Re: [asterisk-users] Need help - CDR MySQL
 From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Sunday, 30 August, 2009 17:17:59

 On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:
 Hello all,

 I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
 but
 without success.

 Is there a proper online manual that describes all the steps to  
 follow
 and debugging/monitoring information?

 When I type in the CLI module show, cdr_addon_mysql.so is not  
 listed,
 although in modules.conf, I added the line load =  
 cdr_addon_mysql.so.
 I also tried preload, but it didn't change anything.

 I also checked res_mysql.conf and cdr_mysql.conf, and the entire
 necessary data for the mysql server is there.

 In cdr_manager.conf and cdr.conf, I set enabled = yes in  
 [general].

 I would appreciate any help I can get at this point, as I'm  
 clueless as
 to what can be wrong.
 Type:  'module load cdr_addon_mysql.so' and correct any errors you  
 see.

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Re: [asterisk-users] Need help - CDR MySQL

[hh174]

yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'

Olivier

Cyprus VoIP a crit:

  I think that the missing component is mysqlclient, but when i "yum 
update mysql", it does nothing.

Anyone know how to download the RPM? I'm using CentOS 5.3.

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:28:35

  
  
You have to fix the dependency issues, which means install the stuff  
you are missing that cdrmysql depends on so u can recompile it.

Sent from my iPod

On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:



  Thanks. I found out that the module didn't load:

[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
Error loading module 'cdr_addon_mysql.so':
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
object
file: No such file or directory

When I checked, I saw that it doesn't exist. It seems that when I
installed the addons, I didn't realize that there were some issues to
resolve first, as in the menuselect, I see that both  
app_addon_sql_mysql
and cdr_addon_mysql have dependencies problems.

What should I do to resolve that?

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59

  
  
On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:


  Hello all,

I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
but
without success.

Is there a proper online manual that describes all the steps to  
follow
and debugging/monitoring information?

When I type in the CLI "module show", cdr_addon_mysql.so is not  
listed,
although in modules.conf, I added the line "load =  
cdr_addon_mysql.so".
I also tried "preload", but it didn't change anything.

I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire
necessary data for the mysql server is there.

In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in  
"[general]".

I would appreciate any help I can get at this point, as I'm  
clueless as
to what can be wrong.
  

Type:  'module load cdr_addon_mysql.so' and correct any errors you  
see.


  
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Re: [asterisk-users] Need help - CDR MySQL

[Cyprus VoIP]

OK, Installed all the dependencies and recompiled everything.

My cdr_addon_mysql.so loads now, but still nothing is logged in the
database.

I ran this on the CLI:
localhost*CLI cdr show status
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: yes
CDR registered backend: cdr_manager
CDR registered backend: cdr-custom
CDR registered backend: Adaptive ODBC
CDR registered backend: csv

Should there be a reference here to MySQL?

Anything else I should set?

Thanks.

 Original Message 
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: hh174 oliv...@hh174.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:58:48 

  
  
yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'
  
Olivier
  
Cyprus VoIP a crit:
  
I think that the missing component is mysqlclient, but when i "yum 
update mysql", it does nothing.

Anyone know how to download the RPM? I'm using CentOS 5.3.

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Pascal Bruno tipas...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:28:35

  

  You have to fix the dependency issues, which means install the stuff  
you are missing that cdrmysql depends on so u can recompile it.

Sent from my iPod

On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:


  
Thanks. I found out that the module didn't load:

[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
Error loading module 'cdr_addon_mysql.so':
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared  
object
file: No such file or directory

When I checked, I saw that it doesn't exist. It seems that when I
installed the addons, I didn't realize that there were some issues to
resolve first, as in the menuselect, I see that both  
app_addon_sql_mysql
and cdr_addon_mysql have dependencies problems.

What should I do to resolve that?

Thanks.

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59

  

  On Sunday 30 August 2009 08:30:54 Cyprus VoIP wrote:

  
Hello all,

I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon,  
but
without success.

Is there a proper online manual that describes all the steps to  
follow
and debugging/monitoring information?

When I type in the CLI "module show", cdr_addon_mysql.so is not  
listed,
although in modules.conf, I added the line "load =  
cdr_addon_mysql.so".
I also tried "preload", but it didn't change anything.

I also checked "res_mysql.conf" and "cdr_mysql.conf", and the entire
necessary data for the mysql server is there.

In "cdr_manager.conf" and "cdr.conf", I set "enabled = yes" in  
"[general]".

I would appreciate any help I can get at this point, as I'm  
clueless as
to what can be wrong.
  
  
  Type:  'module load cdr_addon_mysql.so' and correct any errors you  
see.



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Re: [asterisk-users] Need help - CDR MySQL

[Doug Lytle]

Cyprus VoIP wrote:
 I think that the missing component is mysqlclient, but when i yum 
 update mysql, it does nothing.

   

You need to make sure that mysql-devel is installed  and then re-compile 
add-ons

Doug


-- 
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Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Need help - CDR MySQL

[Cyprus VoIP]

I already did that and now cdr_addon_mysql.so is loaded, but I still 
don't get anything into the database. How can I debug it?

 Original Message  
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Doug Lytle supp...@drdos.info
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 19:27:46

 Cyprus VoIP wrote:
 I think that the missing component is mysqlclient, but when i yum 
 update mysql, it does nothing.

   
 
 You need to make sure that mysql-devel is installed  and then re-compile 
 add-ons
 
 Doug
 
 

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Re: [asterisk-users] Need help - CDR MySQL

[Doug Lytle]

Cyprus VoIP wrote:
 I already did that and now cdr_addon_mysql.so is loaded, but I still 
 don't get anything into the database. How can I debug it?

   

I'm guessing that you've already set the database up correctly and that 
you've entered the correct info in the cdr_mysql.conf?

Doug


-- 
Ben Franklin quote:

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Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Need help - CDR MySQL

[Matt Riddell]

On 31/08/09 9:28 AM, Cyprus VoIP wrote:
 I already did that and now cdr_addon_mysql.so is loaded, but I still
 don't get anything into the database. How can I debug it?

To start with do cdr mysql status.

If that shows as connected, turn debug on (inside 
/etc/asterisk/logger.conf) and do a logger reload in the Asterisk console.

Make another call and see what it says.

I suspect though that you'll find that cdr mysql status shows it as not 
connected.

Did you edit /etc/asterisk/cdr_mysql.conf?

-- 
Cheers,

Matt Riddell
Director
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[asterisk-users] need help, service unavailable, registered but call does not get through

[tom]

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get
thorugh: here is my sip debug outout: thx for ur help!!
asterisk-users@lists.digium.com

--- (13 headers 16 lines) ---
Sending to AA.BBB.CCC.DD : 28127 (NAT)
Using INVITE request as basis request -
Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk.
Found user '701' for '701'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.73:40958
Found unknown media description format BV32 for ID 107
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.73:40958
Looking for 702 in from-internal (domain ABC.dyndns.org)
list_route: hop: sip:7...@123.456.789.000:37587
acerdebian*CLI
--- Transmitting (NAT) to 123.456.789.000:28127 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.73:15158
;branch=z9hG4bK-d8754z-0a540c5d3439c271-1---d8754z-;received=123.456.789.000;rport=28127
From: mesip:7...@abc.dyndns.org sip%3a...@abc.dyndns.org;tag=3c08d834
To: 702sip:7...@abc.dyndns.org sip%3a...@abc.dyndns.org
Call-ID: Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: sip:7...@192.168.1.4 sip%3a...@192.168.1.4
Content-Length: 0



-- Executing [...@from-internal:1] ResetCDR(SIP/701-0864f1b8, ) in
new stack
-- Executing [...@from-internal:2] NoCDR(SIP/701-0864f1b8, ) in new
stack
-- Executing [...@from-internal:3] Wait(SIP/701-0864f1b8, 1) in new
stack
Retransmitting #1 (NAT) to 123.456.789.000:9855:
OPTIONS sip:7...@123.456.789.000:37587;rinstance=9428b8620cd7a907 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK782c5851;rport
Max-Forwards: 70
From: Unknown sip:unkn...@192.168.1.4 sip%3aunkn...@192.168.1.4
;tag=as43db5836
To: sip:7...@123.456.789.000:37587;rinstance=9428b8620cd7a907
Contact: sip:unkn...@192.168.1.4 sip%3aunkn...@192.168.1.4
Call-ID: 564e1f392a3b289f3b655a7200a67...@192.168.1.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.6
Date: Thu, 02 Jul 2009 18:51:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
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Re: [asterisk-users] need help on asterisk call forwarding

[Oguzhan Kayhan]

 On Thursday 30 April 2009 09:37:50 Oguzhan Kayhan wrote:
 Ok I found an example script that said to be work..
 but i have some errors.
 Here is the script and then the error msgs.
 exten = *666*,2,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)

 The DB returns nothing, so it evaluates $[ = 1] which doesn't parse.  Try:

 exten = *666*,2,GotoIf($[0${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)

 When the value doesn't exist, it will evaluate $[0 = 1].  If it does
 exist, it
 evaluates $[01 = 1], which is true.

Still getting
ast_expr2.fl:440 ast_yyerror: ast_yyerror():  syntax error: syntax error,
unexpected '=', expecting $end; Input: = 1

Smthing wrong with the syntax???






 exten = *22*,1,DB_DELETE(CFIM/${CALLERID(NUM)})

 DB_DELETE is a function, not an application.  You can do one of the
 following:

 exten = *22*,1,NoOp(${DB_DELETE(CFIM/${CALLERID(NUM)})})
 exten = *22*,1,DBDel(CFIM/${CALLERID(NUM)})

 --
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[asterisk-users] need help on asterisk call forwarding

[Oguzhan Kayhan]

Hello,
I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
asterisk-gui. Sure there is no menu for that on gui but, when i try to
write some example scripts to extensions.conf to make it work. I totally
failed.

I dont wanna install smthing like freepbx etc on the system so, i need
help to add unconditional etc call forwarding feature for 1.6


Thanks




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Re: [asterisk-users] need help on asterisk call forwarding

[Oguzhan Kayhan]

 Hello,
 I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
 asterisk-gui. Sure there is no menu for that on gui but, when i try to
 write some example scripts to extensions.conf to make it work. I totally
 failed.

 I dont wanna install smthing like freepbx etc on the system so, i need
 help to add unconditional etc call forwarding feature for 1.6


 Thanks


Ok I found an example script that said to be work..
but i have some errors.
Here is the script and then the error msgs.




[apps]
; Toggle Call Forward
exten = *666*,1,Answer
exten = *666*,2,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)
exten = *666*,3,Set(DB(CFBOOLEAN/${CALLERID(NUM)})=0) ; toggle fwd off
exten = *666*,4,Playback(call-forward)
exten = *666*,5,Playback(disabled)
exten = *666*,6,Hangup
; CFBOOLEAN not exist means need to turn it on.
; first check to see if a FWD number is set
exten = *666*,102,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})}= 1]?103:203)
exten = *666*,103,Set(DB(CFBOOLEAN/${CALLERID(NUM)}=1)) ; set FWD toggle on.
exten = *666*,104,Playback(call-forward)
exten = *666*,105,Playback(enabled)
exten = *666*,106,Hangup
exten = *666*,203,Playback(invalid)
exten = *666*,204,Hangup
; Unconditional Call Forward
exten = _*21*X.,1,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,2,Playback(call-forward)
exten = _*21*X.,3,Hangup
exten = *22*,1,DB_DELETE(CFIM/${CALLERID(NUM)})
exten = *22*,2,Hangup



When i dial *666*  i got the msg
syntax error: syntax error, unexpected '=', expecting $end; Input: = 1

and when i dial *22* i got the error


No application 'DB_DELETE' for extension (DLPN_international, *22*, 1)


CAn somebody help me fix the errors and explains what is the error about it?












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Re: [asterisk-users] need help on asterisk call forwarding

[Tilghman Lesher]

On Thursday 30 April 2009 09:37:50 Oguzhan Kayhan wrote:
 Ok I found an example script that said to be work..
 but i have some errors.
 Here is the script and then the error msgs.
 exten = *666*,2,GotoIf($[${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)

The DB returns nothing, so it evaluates $[ = 1] which doesn't parse.  Try:

exten = *666*,2,GotoIf($[0${DB(CFBOOLEAN/${CALLERID(NUM)})} = 1]?3:102)

When the value doesn't exist, it will evaluate $[0 = 1].  If it does exist, it
evaluates $[01 = 1], which is true.

 exten = *22*,1,DB_DELETE(CFIM/${CALLERID(NUM)})

DB_DELETE is a function, not an application.  You can do one of the following:

exten = *22*,1,NoOp(${DB_DELETE(CFIM/${CALLERID(NUM)})})
exten = *22*,1,DBDel(CFIM/${CALLERID(NUM)})

-- 
Tilghman

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Re: [asterisk-users] Need help on how to programmatically call an extension test call state

[eric weaver]

On Thu, Mar 26, 2009 at 10:22 PM, David fire ddf...@gmail.com wrote:

 you can use the asterisk Manager or AMI.
 there is a very good java project asterisk-java but there are librarys for
 almost every languaje.
 look for Asterisk Manager and AMI www.voip-info.org is a good place to
 start


Thanks!
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[asterisk-users] Need help on how to programmatically call an extension test call state

[eric weaver]

I would be grateful if someone could tell me where to find the docs to get
started on the following problem:

A program needs to be written to place a SIP call to a certain extension on
another Asterisk system, and see whether the call state ratchets up to
ringing, then drop, and take action on the results.

Can anyone refer me to the appropriate starting point to read up on this?
Thanks.
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Re: [asterisk-users] Need help on how to programmatically call an extension test call state

[David fire]

you can use the asterisk Manager or AMI.
there is a very good java project asterisk-java but there are librarys for
almost every languaje.
look for Asterisk Manager and AMI www.voip-info.org is a good place to start
David


2009/3/27 eric weaver ecwea...@gmail.com

 I would be grateful if someone could tell me where to find the docs to get
 started on the following problem:

 A program needs to be written to place a SIP call to a certain extension on
 another Asterisk system, and see whether the call state ratchets up to
 ringing, then drop, and take action on the results.

 Can anyone refer me to the appropriate starting point to read up on this?
 Thanks.



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Re: [asterisk-users] Need help on Forwarding

[Philipp Kempgen]

Please don't cross-post.

Max Alex schrieb:
 I am using asterisk 1.4.19,
 I have setup the dialplans to get the incoming call and that will be sent to
 another context by local channel,
 In another context i have setup the ring group, that portion is working
 fine.
 I have noticed that when i have set one of the extension in call forwarding
 in phone (linksys)
 then it says to me 302 Moved Temporarily and call is forwarded to that
 number.
 In this i need to disable the forwarding from dialplan or any configuration
 method, so when the ring group is in process then no call will be forwarded.

In Asterisk 1.6 Queue() has an i option.
---cut---
'i' -- ignore call forward requests from queue members and do nothing
   when they are requested.
---cut---
which might do what you need but obviously that doesn't help you
in Asterisk 1.4.

I guess SIP users connected to a PBX should not do any kind of
forwarding etc. as that's a PBX feature.


Philipp Kempgen
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] Need help on Forwarding

[Max Alex]

Hi All,
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have set one of the extension in call forwarding
in phone (linksys)
then it says to me 302 Moved Temporarily and call is forwarded to that
number.
In this i need to disable the forwarding from dialplan or any configuration
method, so when the ring group is in process then no call will be forwarded.
Please provide help regarding this!!
Thanks in Advance!!
Thanks,
Max Alex
Voip Developer
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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[Zeeshan Zakaria]

D Tuncy, if you don't mind, can you show me your config files which you are
using to successfully register phones on two servers. I tried various
different things, and once it got registered on two servers, but couldn't
dialout on any. Now it is again back on only one server and I don't remember
changing any proxy setting. I must be doing something wrong which I don't
know what it is.

Zeeshan

On Tue, Jan 20, 2009 at 1:58 AM, D Tucny d...@tucny.com wrote:

 That's not my experience...
 e.g.

 SIP Phone show register

 LINE REGISTRATION TABLE
 Proxy Registration: ENABLED, state: REGISTERED
 line  APR  state  timer   expires proxy:port
   ---  -  --  --
  
 1 111  REGISTERED 115 98  192.168.1.1:5060
 2 111  REGISTERED 115 98  192.168.1.12:5060
 3 ...  NONE   0   0   undefined:0
 4 ...  NONE   0   0   undefined:0
 5 ...  NONE   0   0   undefined:0
 6 ...  NONE   0   0   undefined:0
 1-BU  111  REGISTERED 115 98  192.168.1.1:5060

 Note: APR is Authenticated, Provisioned, Registered

 I can see registers on both servers...

 d

 2009/1/20 Yehavi Bourvine yehavi.bourv...@gmail.com

  From my experience it won't register to the second box, only to the first
 one. Why? god knows...

 __Yehavi:

 2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me yet.

 I have universal setting in SIPDefault.cnf and phone specific settings
 in SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like
 this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug
 peer 224 to see if it's reaching the server but not authenticating etc...
 Also, you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[D Tucny]

Zeeshan,

I've put them on a server so that you should be able to access them...

http://astinfo.newportcoastsoftware.com/SIPDefault.cnf
http://astinfo.newportcoastsoftware.com/SIPmacaddress.cnf

d

2009/1/21 Zeeshan Zakaria zisha...@gmail.com

 D Tuncy, if you don't mind, can you show me your config files which you are
 using to successfully register phones on two servers. I tried various
 different things, and once it got registered on two servers, but couldn't
 dialout on any. Now it is again back on only one server and I don't remember
 changing any proxy setting. I must be doing something wrong which I don't
 know what it is.

 Zeeshan


 On Tue, Jan 20, 2009 at 1:58 AM, D Tucny d...@tucny.com wrote:

 That's not my experience...
 e.g.

 SIP Phone show register

 LINE REGISTRATION TABLE
 Proxy Registration: ENABLED, state: REGISTERED
 line  APR  state  timer   expires proxy:port
   ---  -  --  --
  
 1 111  REGISTERED 115 98  192.168.1.1:5060
 2 111  REGISTERED 115 98  192.168.1.12:5060
 3 ...  NONE   0   0   undefined:0
 4 ...  NONE   0   0   undefined:0
 5 ...  NONE   0   0   undefined:0
 6 ...  NONE   0   0   undefined:0
 1-BU  111  REGISTERED 115 98  192.168.1.1:5060

 Note: APR is Authenticated, Provisioned, Registered

 I can see registers on both servers...

 d

 2009/1/20 Yehavi Bourvine yehavi.bourv...@gmail.com

  From my experience it won't register to the second box, only to the
 first one. Why? god knows...

 __Yehavi:

 2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me
 yet.

 I have universal setting in SIPDefault.cnf and phone specific settings
 in SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like
 this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug
 peer 224 to see if it's reaching the server but not authenticating etc...
 Also, you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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 Zeeshan A Zakaria

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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[Zeeshan Zakaria]

Thanks for these scripts. Good thing is that now phone has got registered on
two servers.

Now a new weird problem I am having is that it is not loading all the
entries from SIPDefault.cnf. It is loading only proxy related settings and
ignoring the rest.

Also I have 6 lines on this phone, but it doesn't register if I set any
extension on lines 3 and above. It only registers first two lines.

Any idea what could be reason for this to happen?

Zeeshan


On Wed, Jan 21, 2009 at 9:01 AM, D Tucny d...@tucny.com wrote:

 Zeeshan,

 I've put them on a server so that you should be able to access them...

 http://astinfo.newportcoastsoftware.com/SIPDefault.cnf
 http://astinfo.newportcoastsoftware.com/SIPmacaddress.cnf

 d

 2009/1/21 Zeeshan Zakaria zisha...@gmail.com

 D Tuncy, if you don't mind, can you show me your config files which you are
 using to successfully register phones on two servers. I tried various
 different things, and once it got registered on two servers, but couldn't
 dialout on any. Now it is again back on only one server and I don't remember
 changing any proxy setting. I must be doing something wrong which I don't
 know what it is.

 Zeeshan


 On Tue, Jan 20, 2009 at 1:58 AM, D Tucny d...@tucny.com wrote:

 That's not my experience...
 e.g.

 SIP Phone show register

 LINE REGISTRATION TABLE
 Proxy Registration: ENABLED, state: REGISTERED
 line  APR  state  timer   expires proxy:port
   ---  -  --  --
  
 1 111  REGISTERED 115 98  192.168.1.1:5060
 2 111  REGISTERED 115 98  192.168.1.12:5060
 3 ...  NONE   0   0   undefined:0
 4 ...  NONE   0   0   undefined:0
 5 ...  NONE   0   0   undefined:0
 6 ...  NONE   0   0   undefined:0
 1-BU  111  REGISTERED 115 98  192.168.1.1:5060

 Note: APR is Authenticated, Provisioned, Registered

 I can see registers on both servers...

 d

 2009/1/20 Yehavi Bourvine yehavi.bourv...@gmail.com

  From my experience it won't register to the second box, only to the
 first one. Why? god knows...

 __Yehavi:

 2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me
 yet.

 I have universal setting in SIPDefault.cnf and phone specific settings
 in SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like
 this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug
 peer 224 to see if it's reaching the server but not authenticating etc...
 Also, you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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 --
 Zeeshan A Zakaria

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[asterisk-users] Need Help

[Bayardo Sanchez]

i need a program for monitorin my bandwitch of my asterisk server

-- 
Bayardo Sánchez García
Web Developer - Internet Portals
Linux User: #418392
Ubuntu User #14171
America Central - Managua, NI (505) 249-2853 -  4886876
IM msn messenger: bjsanch...@hotmail.com
Skype: bayardo.sanchez
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