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Gopalakrishnan N | 24 May 2013 22:29
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Asterisk 11 dtmf not recognised

Hi

I have a dialplan as per the following,

extensions.conf
[avgtest]
exten = 100,n,Playback(avgtest/message1)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,,5,3) ; Attempts for 5 times with 3 seconds of timeout
exten = 100,n,GotoIf($["${inPIN}" = "${rightPIN}"]?pin-accepted,1)
exten = 100,n,Hangup() ; Didn't go to pin-accepted, so play badPIN and hangup
exten=pinaccepted,1,Playback(avgtest/message2) ; correct pin, play

sipconf
[1001]
uername=1001
secret=1001
context=avgtest
disallow=all
allow=ulaw
allow=alaw
dtmfmode=auto
type=friend
host=dynamic
canreinvite=yes
relaxdtmf=yes

This looks very simple but dtmf is not recognised.

Am using asterisk 11.

Any suggestions is much appreciated.

Regards

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luke devon | 24 May 2013 19:46
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asterisk-gui-2.1.0-rc1

Hi 

I have installed asterisk-gui-2.1.0-rc1 . After I logged in to the GUI , it was continuously refreshing the web browser and trying to load the configurations. 

Can I know where is gone wrong ?

Thanks in advance
Luke 
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Thorsten Göllner | 24 May 2013 15:34
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Pri-Debug-Log / Is Early Media supported by provider?

Hi,

I tried to use Early Media:

exten => 1,1,Playback(demo-thanks,noanswer)
  same => n,Hangup()

But when calling my extension I do not hear the voicefile - I only hear 
the ring tone. In the Asterisk-Log I can see, that the voicefile is played.

I got the same result when using "Progress()" in the first priority.

I tried "pri set debug on span 1" and got the following:
(I replaced originating caller id by 123456)

PRI Span: 1 < Protocol Discriminator: Q.931 (8)  len=48
PRI Span: 1 < TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent from 
originator)
PRI Span: 1 < Message Type: SETUP (5)
PRI Span: 1 < [a1]
PRI Span: 1 < Sending Complete (len= 1)
PRI Span: 1 < [04 03 80 90 a3]
PRI Span: 1 < Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0 Info 
transfer capability: Speech (0)
PRI Span: 1 <                              Ext: 1  Trans mode/rate: 
64kbps, circuit-mode (16)
PRI Span: 1 <                                User information layer 1: 
A-Law (35)
PRI Span: 1 < [18 03 a9 83 8e]
PRI Span: 1 < Channel ID (len= 5) [ Ext: 1  IntID: Implicit Other(PRI)  
Spare: 0  Exclusive  Dchan: 0
PRI Span: 1 <                       ChanSel: As indicated in following 
octets
PRI Span: 1 <                       Ext: 1  Coding: 0  Number Specified  
Channel Type: 3
PRI Span: 1 <                       Ext: 1  Channel: 14 Type: CPE]
PRI Span: 1 < [6c 0c 21 83 31 37 38 31 34 38 34 31 34 32]
PRI Span: 1 < Calling Number (len=14) [ Ext: 0  TON: National Number 
(2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
PRI Span: 1 <                           Presentation: Presentation 
allowed of network provided number (3)  '123456' ]
PRI Span: 1 < [70 0c c1 36 30 32 31 32 35 30 30 30 33 30]
PRI Span: 1 < Called Number (len=14) [ Ext: 1  TON: Subscriber Number 
(4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1' ]
PRI Span: 1 < [7d 02 91 81]
PRI Span: 1 < IE: High-layer Compatibility (len = 4)
PRI Span: 1 -- Making new call for cref 14783
PRI Span: 1 Received message for call 0x7f48ec00a370 on link 
0x7f49201859a0 TEI/SAPI 0/0
PRI Span: 1 -- Processing Q.931 Call Setup
PRI Span: 1 -- Processing IE 161 (cs0, Sending Complete)
PRI Span: 1 -- Processing IE 4 (cs0, Bearer Capability)
PRI Span: 1 -- Processing IE 24 (cs0, Channel Identification)
PRI Span: 1 -- Processing IE 108 (cs0, Calling Party Number)
PRI Span: 1 -- Processing IE 112 (cs0, Called Party Number)
PRI Span: 1 -- Processing IE 125 (cs0, High-layer Compatibility)
PRI Span: 1 q931.c:8281 post_handle_q931_message: Call 14783 enters 
state 6 (Call Present).  Hold state: Idle
Span 1: Processing event PRI_EVENT_RING(5)
PRI Span: 1 q931.c:5477 q931_call_proceeding: Call 14783 enters state 9 
(Incoming Call Proceeding).  Hold state: Idle
PRI Span: 1
PRI Span: 1 > DL-DATA request
PRI Span: 1 > Protocol Discriminator: Q.931 (8)  len=10
PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to 
originator)
PRI Span: 1 > Message Type: CALL PROCEEDING (2)
PRI Span: 1 TEI=0 Transmitting N(S)=70, window is open V(A)=70 K=7
PRI Span: 1
PRI Span: 1 > Protocol Discriminator: Q.931 (8)  len=10
PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to 
originator)
PRI Span: 1 > Message Type: CALL PROCEEDING (2)
PRI Span: 1 > [18 03 a9 83 8e]
PRI Span: 1 > Channel ID (len= 5) [ Ext: 1  IntID: Implicit Other(PRI)  
Spare: 0  Exclusive  Dchan: 0
PRI Span: 1 >                       ChanSel: As indicated in following 
octets
PRI Span: 1 >                       Ext: 1  Coding: 0  Number Specified  
Channel Type: 3
PRI Span: 1 >                       Ext: 1  Channel: 14 Type: CPE]
     -- Accepting call from '123456' to '1' on channel 0/14, span 1
     -- Executing [1 <at> port1:1] NoOp("DAHDI/i1/123456-245", "") in new stack
     -- Executing [1 <at> port1:2] Playback("DAHDI/i1/123456-245", 
"demo-thanks,noanswer") in new stack
     -- <DAHDI/i1/123456-245> Playing 'demo-thanks.gsm' (language 
'de_female')
     -- Executing [1 <at> port1:3] Hangup("DAHDI/i1/123456-245", "") in new 
stack
   == Spawn extension (port1, 1, 3) exited non-zero on 
'DAHDI/i1/123456-245'
PRI Span: 1 q931.c:6837 q931_hangup: Hangup other cref:14783
PRI Span: 1 q931.c:6594 __q931_hangup: ourstate Incoming Call 
Proceeding, peerstate Outgoing Call Proceeding, hold-state Idle
PRI Span: 1 q931.c:5783 q931_disconnect: Call 14783 enters state 11 
(Disconnect Request).  Hold state: Idle
PRI Span: 1
PRI Span: 1 > DL-DATA request
PRI Span: 1 > Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to 
originator)
PRI Span: 1 > Message Type: DISCONNECT (69)
PRI Span: 1 TEI=0 Transmitting N(S)=71, window is open V(A)=71 K=7
PRI Span: 1
PRI Span: 1 > Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 14783/0x39BF) (Sent to 
originator)
PRI Span: 1 > Message Type: DISCONNECT (69)
PRI Span: 1 > [08 02 81 90]
PRI Span: 1 > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  
Spare: 0  Location: Private network serving the local user (1)
PRI Span: 1 >                  Ext: 1  Cause: Normal Clearing (16), 
class = Normal Event (1) ]
     -- Hungup 'DAHDI/i1/123456-245'

Can you see, if my provider supports Early Media? Or do I have to ask 
the tech department there?

Ubuntu 64 bit kernel: 3.2.0-38-generic
Asterisk: 11.2.1
DAHDI: 2.6.1
Libpri: 1.4.12
Sangoma A104 (4 port E1, germany)
Wanpipe Driver: 3.5.28

Best regards
-Thorsten-

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luke devon | 24 May 2013 12:32
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Registration timed out - for created users


Hi all , 

I have managed to install and configure the 

1. asterisk-1.8-current
2. dahdi-linux-complete-current


I did not faced any issues during the installation. After that I installed X-Lite soft phone in two different PCs and tested the setup. every thing was success. I was able make calls from each extensions.


But when I observe the log files , i could see some messages ......

chan_sip.c:    -- Registration for 'alphaUser <at> 192.168.1.12' timed out, trying again (Attempt #2)

Something is not right. I have double check the configurations. But I could not find where I have done the mistake.

following is my configurations,

sip.conf
-------
register => alpahaUser:1234 <at> 192.168.1.10

[alphaUser]
type=friend
username=alphaUser
secret=1234
context=tutorial
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=tutorial
mailbox=alphaUser <at> internal


extensions.conf
----------------
[tutorial]
exten => 5555,1,Dial(SIP/alphaUser)


Please help me to identify and resolve the issue .

Thanks in Advance
Luke.


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Nick Khamis | 23 May 2013 17:08
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Asterisk on Solaris

Hello Everyone,

I have bumped into the thralling penguin page on linux vs solaris for
asterisk. Does the benchmark still hold with the newer versions of
kernels? Curious to know of your thoughts. Also, they mentioned
running it on Sun Fire x2100, but no benchmarks were given for that.

Can increased performance be accomplished simply by changing to
Solaris or OpenSolaris?

Kind Regards,

Nick.

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Gopalakrishnan N | 23 May 2013 15:49
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GotoIf function

Hi,

Actually i would like to get the input from the user and he should not try more than 3 times, he can try more than 3 times, if yes it will get routed to the next priority and if not it goes to the loopback again from the beginning.

And following is the one I created, I just want to know whether this will validate the input and will allow for 3 times.... 

exten => s,1,GotoIfTime(08:00-09:00,mon-fri,*,*?2:avgtech,1)
exten => s,n,Background(voicemessage_1)
exten => s,n(voicemessage2),Background(voicemessage_2)

exten => s,n(begin),Set(wait=2)
exten => s,n,Set(gottries=0)
exten => s,n,Read(get,"silence/1",,,,${wait})

exten => s,n(gotnothing),Set(gottries=$[${gottries}+1]
exten => s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit)
exten => s,n(reallynothing),GotoIf($[${gottries}>3]?done:voicemessage5)
exten => s,n(done),Background(voicemessage3)
exten => s,n,Background(voicemessage4)
exten => s,n,Playback(moh)
exten => s,n, ; Addittional messageing
exten => s,n,Queue(general technical team)

exten => s,n(voicemessage5),Goto(voicemessage2) 

exten => s,n(gotdigit),Set(got=${get})
exten => s,n,GotoIf( $[ "${got}" = "1"]?doneinstall)
exten => s,n(doneinstall),Background(voicemessage3)
exten => s,n,Background(voicemessage4)
exten => s,n,Playback(moh)
exten => s,n, ; Addittional messageing
exten => s,n,Queue(installation technical skill)

exten => s,n,GotoIf( $[ "${got}" = "2"]?done2)
exten => s,n(done2),Background(voicemessage6)
exten => s,n,Goto(begin2)
exten => s,n(begin2),Set(wait=2)
exten => s,n,Set(gottries=0)
exten => s,n,Read(get,"silence/1",,,,${wait})
exten => s,n(gotnothing),Set(gottries=$[${gottries}+1]
exten => s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit2)
exten => s,n(reallynothing),GotoIf($[${gottries}>3]?done:option2)
exten => s,n(done),Background(voicemessage3)
exten => s,n,Background(voicemessage4)
exten => s,n,Playback(moh)
exten => s,n, ; Addittional messageing
exten => s,n,Queue(general technical skill)

exten => s,n(option2),Background(voicemessage5)
exten => s,n,Goto(done2)

and so on... for digit 3...

Thanks in advance... 

Regards. 
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Andrey Utkin | 23 May 2013 15:41
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Seeking for TTS engine supporting Hebrew

I am not aware of any alive project on this field.
My client is interested in it. He told me that Google Translate web
service used to provide pronunciation of hebrew texts, but now speech
button is disabled for Hebrew, and also direct requesting for speech
generation (by  Lefteris Zafiris's awsome scripts) fails.
I can work on Asterisk integration by myself, i'd be happy to know of
such engine(s) at all.
--
Andrey Utkin

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bilal ghayyad | 23 May 2013 11:57
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Integration with skype

Hello;

There is no free channel to be used to have integration between asterisk and skype? What is the software that
I can use to send and receive chat messages on skype network?

Regards
Bilal

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bilal ghayyad | 23 May 2013 11:49
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Jabber

Hello;

Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to communicate with them. But, how much
jabber channel in asterisk is stable and updated?

Regards
Bilal

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Positively Optimistic | 23 May 2013 02:04
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Diversion vs. P-Asserted-Id vs. Remote-Party-Id vs. P-Charge-Info vs. From Fields

We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number.   Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating?   

Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the lerg...  i.e., toll-free...

Does anyone have any experience with this?

Thanks,
Optimistic...


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Andrew Colin | 22 May 2013 16:39
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Error 488 Not Acceptable Here

Hi guys,

Any idea why I am getting this error when someone tries to send me a T38 
Fax?

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Gmane