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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.
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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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Asterisk Development Team | 22 May 2013 15:56
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Changes to the community service maintenance notifications

You may have noticed (or maybe not) that there have been several
maintenance notifications for the asterisk.org community services this
month. We are working hard to keep up the services running smoothly,
and those notices are sent whenever we think our maintenance may
interfere with the operation of any of the services.

So far, it's been our policy that we send out a maintenance
notification whenever we do anything other than the most minor
maintenance on the services. You can usually read "may have
intermittent availability" as "it should be available unless things go
horribly wrong".

We now realize that most of these notifications are just spam for most
of the community. It is also cumbersome for us to send out the
notifications every time we touch the services. Especially considering
that the services are typically unavailable for at most a few minutes,
if at all.

In an effort to reduce spam and make service availability more
predictable, we're changing the policy about when we send
notifications about community service availability.

Starting on Monday, May 27th, we will have a regular maintenance
window every Monday for one hour starting at 9:00 PM Central Time
(that's 02:00 UTC during daylight saving time in the summer, and 03:00
UTC during standard time). We will try to restrict the service
impacting maintenance to that weekly window.

For the times where there might be a service interruption outside of
that window (either when it needs to be coordinated with our colo
provider, or if the maintenance will take longer than one hour), we
will send notice of the impending service interruption to just the
asterisk-announce mailing list[1].

This will help us in planning service upgrades and maintenance, and
reduce the amount of unnecessary email for the community.

 [1]: http://lists.digium.com/mailman/listinfo/asterisk-announce

 -- Digium's Asterisk Development Team

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Lefteris Zafiris | 22 May 2013 02:11
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Automatic Speech Recognition and Text To Speech using iSpeech

Hi,

a set of AGI scripts that provide ASR and TTS for asterisk using the iSpeech API (http://www.ispeech.org/) are available on this page:


This is the first public release, updates will soon follow.
Feel free to test and report.

Regards,

Lefteris Zafiris

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asterisk users | 21 May 2013 19:11
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Failed to authenticate device "Ext 110"

I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn.  It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows:

[May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" <sip:110 <at> 192.168.6.2>;tag=1130259112

Asterisk 192.168.6.2
OpenVPN on router 10.8.0.1
Remote Yealink phone 10.8.0.6

The remote phone shows as being registered:
PBX*CLI> sip show peers
Name/username  Host  Dyn Forcerport ACL Port     Status      Description
110/110   10.8.0.6  D   A  5062   OK (111 ms) Yealink OpenVPN

Also, if there is voicemail in the mailbox for 110, the phone's message light is lit and it beeps periodically.

toshi*CLI> sip show peer 110


  * Name       : 110
  Description  : Yealink OpenVPN
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : remote-phones
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     :
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 1
  Pickupgroup  : 1
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      : 110
  VM Extension : asterisk
  LastMsgsSent : 1/0
  Call limit   : 4
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "Ext 110" <110>
  MaxCallBR    : 384 kbps
  Expire       : 608
  Insecure     : no
  Force rport  : No
  Symmetric RTP: No
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 10.8.0.6:5062
  Defaddr->IP  : 10.8.0.6:5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 110
  SIP Options  : (none)
  Codecs       : (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing :  No
  Status       : OK (237 ms)
  Useragent    : Yealink SIP-T28P 2.61.23.3 00:15:65:xx.xx.xx
  Reg. Contact : sip:110 <at> 10.8.0.6:5062
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

sip.conf:

[110]
context=remote-phones
type=peer
host=dynamic
qualify=1500
canreinvite=no
dtmfmode=rfc2833
progressinband=no
callgroup=1
pickupgroup=1           ; We can do call pickup for call group 1
call-limit=4
busy-level=1
qualify=yes
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
nat=no
qualify=8000
description=Yealink OpenVPN
defaultuser=110
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx
callerid="Ext 110" <110>
mailbox=110
defaultip=10.8.0.6
port=5060
disallow=all
allow=ulaw

Any suggestions on what might be happening here, and how it could be resolved?

THANKS ALL!


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Ahmed Munir | 21 May 2013 17:19
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Asterisk Log rotate not working

Hi,

Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily basis which was working perfect. Now in couple of months back, the logrotate feature is not working at all but simply appending the logs in 'messages' file. Listing down down the configuration for logrotate below;

/var/log/asterisk/messages {
missingok
rotate 5
daily
postrotate
/usr/sbin/asterisk -rx 'logger reload' > /dev/null 2> /dev/null
endscript
}

As asterisk is running by user: root so no need set asterisk permissions 'create 0640 asterisk asterisk' in above configuration.

Please advise so I can resolve this issue.



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Ahmed Munir Chohan

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Gmane