Re: [asterisk-users] Moving from Redfone's Fonebridge to Allo 2nd Gen PRI card

[Amit Patkar]

Hi

Allo also offer gateways - similar to Fonebridge. Have you 
tried/evaluated that? It will help us to know those results.

What is CPU & RAM utilization on this server?
What kind of work load you run? Does it involve transcoding (codecs 
used)? Are these calls passed on to SIP client or these are IVR calls?
How many cards are installed in this server? (Total PRI terminated on 
this server).

What is average call duration?

*Regards,*
Amit

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[asterisk-users] Moving from Redfone's Fonebridge to Allo 2nd Gen PRI card

[tirveni yadav]

We have been running around than 40 asterisk servers running on Debian
Squeeze for last three years, handling traffic of more than few
hundred thousand calls per day.

Our setup's PRI-banks were using Redfone's Fonebridge. We had PRIs
from multiple telephony providers. And Redfone's Fonebridge handled
all that easily.

But all good things come to end. Redfone's Fonebridge was not
available anymore. And we had to seek replacement.

We got to know abot Allo's 2nd Generation PRI card with echo cancellation.

Though we had not used any Allo product before, hence we were unsure
of using them. But what made us select Allo for migration was that,
Allo provides 5 years of warranty on their products.

Hence Allo card was selected.

We have started to use the card on the production and results have
been good. In eight hours in a day it is handling around 50,000 calls,
without any issues. There are no issues in voice quality.


Allo PRI Card:   2aCP4e (2nd Gen)
Processor:  Intel(R)Xeon(R) CPU   E5620  @ 2.40GHz
OS  Debian GNU/Linux 7.6 (wheezy)
RAM:   16 GB
10AM 6 PM 5 Per day
Libpri: 1.4.12-2
Asterisk: 1.8.13.1~dfsg1-3+deb7u3


-- 
Regards,

Tirveni Yadav

What is this Universe ? From what it arises ? Into what does it go?
In freedom it arises, In freedom it rests and into freedom it melts away.
Upanishads.

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[asterisk-users] Planned maintenance for community services today, Wednesday, August 6th, 2014

[Digium's Asterisk Development Team]

Tonight the Asterisk issue tracker will have intermittent availability
due to maintenance. This maintenance will begin at approximately 9:00
PM CST[1] and should last no longer than one hour.

The affected services are:

* issues.asterisk.org

Thank you for your support!

[1]: http://tinyurl.com/lhhfsua (see converted times)

-- 
Digium's Asterisk Development Team

Check us out at: http://digium.com & http://asterisk.org

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[asterisk-users] Asterisk IP7960 and MWI Issue

[Paul Greenberg]

All,


I am running the following setup:

Linux 3.10.0-123.4.2.el7.x86_64 #1 SMP Mon Jun 30 16:09:14 UTC 2014 x86_64 
x86_64 x86_64 GNU/Linux
Asterisk 12.4.0

Cisco IP Phone 7960G


I have an issue with MWI. For some reason after I delete my voicemail messages, 
the MWI of the phone is ON for another 20-30 minutes. It seems that there is a 
poll interval of some kind. I am not sure whether it is a setting on the phone 
or the asterisk.


Any  ideas?


Best Regards,
Paul Greenberg, Esq.

Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
E-mail: p...@greenberg.pro
Tel:  201-402-6777
Fax:  201-301-8876
Cell: 212-380-7343
Web: http://www.greenberg.pro/
Twitter: @nymetrolaw

CONFIDENTIALITY NOTICE: The information contained in this email, including any 
attachment(s), is confidential information that may be privileged and exempt 
from disclosure under applicable law, and is intended only for the exclusive 
use by the person(s) mentioned above as recipient(s). If you are not the 
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Re: [asterisk-users] Loud Ringers and paging systems...

[Kevin Larsen]

> 
> Will your approach handle ringing more than one of the three 
> extensions simultaneously?
> 
>   --Don

Not if they are in the same paging zone, but neither would using the night 
ringer function on the pa system, so I consider that acceptable. Not even 
sure what would be considered correct in the case of two at the same time. 
First come/first serve is the only thing that comes to mind as being 
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[asterisk-users] Anyone have any experience with inbound SIP trunks from Simwood?

[A J Stiles]

I'm trying -- unsuccessfully! -- to configure an inbound trunk with Simwood, 
and I was hoping someone on this list might have managed to do this.

I have configured some numbers to route to a SIP endpoint
  %e164@customer's server
and convinced the customer to open up UDP ports 5060 and 1 - 2.

Calling the number gets a SIP request from Simwood.  The customer's machine 
then sends a SIP 401 response.  Simwood send an ACK .  and then nothing.  
Nothing appears in the Asterisk CLI; to get the SIP trace I used the command

# ngrep -t -q -n -q -Wbyline -deth0 1283 port 5060

(note that 1283 = the STD code from which the call is originating, so it 
should show up in any related packets.)


##  sip.conf  ##
[simwood_in_slough]
type=friend
host=178.22.140.34
fromdomain=178.22.140.34
permit=178.22.140.34/255.255.255.255
qualify=no
context=from-simwood
dtmfmode=rfc2833
insecure=invite,port
disallow=all
allow=alaw
nat=yes
directmedia=no

##  extensions.conf  ##
[from-simwood]
extension => s,1,NoOp(Call via Simwood form '${CALLERID(num)}' to '${EXTEN}')
extension => s,n,Hangup()


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Loud Ringers and paging systems...

[Don Kelly]

The basic concept is that the original call will run a script that creates a
call file to call the paging system and play a specific audio file. It also
passes into the paging call its channel name. In the call to the paging
system, I use the SHARED function to write back to the original calls'
channel the channel name of the paging call. Then when the original call is
answered, it runs a subroutine that redirects the paging call to a priority
that hangs that call up. 

Will your approach handle ringing more than one of the three extensions
simultaneously?

 

  --Don

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Re: [asterisk-users] Checking for human answer

[Tiago Geada]

Hello


We use originate that places a call in a queue (channel parameter is a
Local/dialplan)

When the call is answered in queue, it is bridged with the operator, and
then starts the second channel leg: Dial out to wherever trough local
channel


we set a sip header with dialstatus, so if the operator hangs the call, we
see a CANCEL back in our pbx


On 20 July 2014 17:20, Valter Nogueira  wrote:

> In fact, Asterisk console shows a message warning that call is not
> finished because of the macro leg
>
>
>
>
> 2014-07-20 13:19 GMT-03:00 Valter Nogueira :
>
> No, I am testing with IP phones.
>>
>> When caller hangs-out the macro is not aware - but when calle hangs the
>> macro is.
>>
>>
>> 2014-07-20 12:31 GMT-03:00 Doug Lytle :
>>
>> Valter Nogueira wrote:
>>>
 The problem is in the opposite side - when someone call us and hangs
 before the operator press the number.

>>>
>>> Then my guess would be you're on analog lines?
>>>
>>> Without call supervision on the line, there will be no way of detecting
>>> when an analog call has been dropped, other then when the operator has
>>> decided there is nobody there and hangs up at which point the call should
>>> be dropped.
>>>
>>> Digital lines and VOIP lines shouldn't have this issue since they have
>>> call supervision.
>>>
>>>
>>> Doug
>>>
>>>
>>>
>>>
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>>
>>
>
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Re: [asterisk-users] Loud Ringers and paging systems...

[Kevin Larsen]

> if you use a papt2 or so spa2101 then you could have alert info set 
> to different lengths or styles of ringers
> 
> i use that in a dorm with phones and have the phones ring short 
> rings at night so it wont wake up the students 

I do not use either of those devices, but after posting this yesterday, I 
did end up coming up with a way to do this making use of Asterisk dialplan 
code and the paging side of the paging/night ringer system.

The basic concept is that the original call will run a script that creates 
a call file to call the paging system and play a specific audio file. It 
also passes into the paging call its channel name. In the call to the 
paging system, I use the SHARED function to write back to the original 
calls' channel the channel name of the paging call. Then when the original 
call is answered, it runs a subroutine that redirects the paging call to a 
priority that hangs that call up.

If anyone is interested, I have my proof of concept code that I could post 
up to the group.-- 
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Re: [asterisk-users] From and To headers contain same account in INVITEs

[Olli Heiskanen]

Hi,

There we go, that was it. Thank you Joshua!

cheers,
Olli


2014-08-06 15:26 GMT+03:00 Joshua Colp :

> Olli Heiskanen wrote:
>
>> Hello,
>>
>
> Kia ora,
>
>  I noticed a strange thing while testing my Asterisk-Kamailio Realtime
>> setup. In an INVITE the From and To headers contain the same number when
>> calling through a Realtime integration setup. This happens when the
>> INVITE leaves Asterisk.
>>
>> Can you guys tell me what might be causing this? I have 6...@testers.com
>>  as a websocket client and 7...@testers.com
>>  (caller) using a Zoiper client (db output
>>
>> below). The call itself works, audio and all, only those headers are
>> puzzling to me. I noticed this when I tried to add a label saying '700
>> calling' on my web page. The same thing happens when I call from 660 to
>> 700.
>>
>
> Your configuration has "fromuser" set which explicitly sets the user
> portion of the From header to what you specify. This is commonly used for
> ITSPs as they use that to determine who you are trying to authenticate as.
> If you require this to be set then caller id information has to be
> transported in a different manner (RPID or PAI).
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] From and To headers contain same account in INVITEs

[Joshua Colp]

Olli Heiskanen wrote:

Hello,


Kia ora,


I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the
INVITE leaves Asterisk.

Can you guys tell me what might be causing this? I have 6...@testers.com
 as a websocket client and 7...@testers.com
 (caller) using a Zoiper client (db output
below). The call itself works, audio and all, only those headers are
puzzling to me. I noticed this when I tried to add a label saying '700
calling' on my web page. The same thing happens when I call from 660 to
700.


Your configuration has "fromuser" set which explicitly sets the user 
portion of the From header to what you specify. This is commonly used 
for ITSPs as they use that to determine who you are trying to 
authenticate as. If you require this to be set then caller id 
information has to be transported in a different manner (RPID or PAI).


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Loud Ringers and paging systems...

[Israel Gottlieb]

if you use a papt2 or so spa2101 then you could have alert info set to
different lengths or styles of ringers

i use that in a dorm with phones and have the phones ring short rings at
night so it wont wake up the students


On Tue, Aug 5, 2014 at 10:24 PM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:

> Working on a paging system for one of my sites and running into something
> I can't believe is this hard. In one of the zones, they want to have three
> different extensions ring over the pa system, using it as a loud ringer.
> Now the paging system does have a loud ringer built in and I can easily
> have it do a simultaneous ring, but all of the extensions will sound the
> same over the loud ringer. Of course, we want them to have different rings
> over the pa system so that all three people don't have to check their phone
> every time it rings.
>
> So far, the only semi solution I am coming up with (short of buying three
> different loud ringers and wiring them into the paging system) is to have
> my dialplan generate a call file that will make a second call to the paging
> system and play out an audio file based on who we are doing the loud ringer
> for. This has the disadvantage that it isn't a true loud ringer as it will
> only play for however long I tell it to and it won't cut off if they answer
> the phone before the audio file finishes playing.
>
> Anyone have any suggestions about a better way to handle this? Really
> hoping there is an Asterisk dialplan solution as I don't want to triple my
> paging hardware just to add one tiny piece of functionality.
>
> Kevin Larsen
> --
> _
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[asterisk-users] From and To headers contain same account in INVITEs

[Olli Heiskanen]

Hello,

I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.

Can you guys tell me what might be causing this? I have 6...@testers.com as
a websocket client and 7...@testers.com (caller) using a Zoiper client (db
output below). The call itself works, audio and all, only those headers are
puzzling to me. I noticed this when I tried to add a label saying '700
calling' on my web page. The same thing happens when I call from 660 to
700.

My Asterisk is 11.11.0 running on CentOS 6.5.

An INVITE is sent from my client to Kamailio and then to Asterisk:
(both Kamailio and Asterisk are at 1.1.1.1)

INVITE sip:6...@testers.com;transport=UDP SIP/2.0
Record-Route: 
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0
Via: SIP/2.0/UDP 2.2.2.2:37730
;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z-
Max-Forwards: 16
Contact: 
To: 
From: ;tag=fd070807
Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2NkODg2OTk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 239

v=0
o=Z 0 0 IN IP4 2.2.2.2
s=Z
c=IN IP4 2.2.2.2
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

... and Asterisk responds with Trying:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0;received=1.1.1.1;rport=5060
Via: SIP/2.0/UDP 2.2.2.2:37730
;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z-
Record-Route: 
From: ;tag=fd070807
To: 
Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2NkODg2OTk.
CSeq: 2 INVITE
Server: I Am the Devil
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: 
Content-Length: 0

And when Asterisk sends out the INVITE, From and To headers both have the
same number:

INVITE sip:660@1.1.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5070;branch=z9hG4bK75de61d0;rport
Max-Forwards: 70
From: ;tag=as7b7c32a5
To: 
Contact: 
Call-ID: 7240b8a011890ec677f185f454858...@testers.com
CSeq: 102 INVITE
User-Agent: I Am the Devil
Date: Wed, 06 Aug 2014 09:54:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 801

v=0
o=root 969416519 969416519 IN IP4 1.1.1.1
s=Asterisk PBX 11.11.0
c=IN IP4 1.1.1.1
t=0 0
m=audio 18740 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:50d777041673316422560b90281fcd2e
a=ice-pwd:0093fdde724f8a411742661c31c90f21
a=candidate:H5bdd423d 1 UDP 2130706431 1.1.1.1 18740 typ host
a=candidate:S5bdd423d 1 UDP 1694498815 1.1.1.1 18740 typ srflx
a=candidate:H5bdd423d 2 UDP 2130706430 1.1.1.1 18741 typ host
a=candidate:S5bdd423d 2 UDP 1694498814 1.1.1.1 18742 typ srflx
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05
a=sendrecv


Here's the dialplan, nothing special:

exten => _XXX,1,NoOp(general : Dialed ${EXTEN})
 same => n,Dial(SIP/${EXTEN},3600,rt)
 same => n,Hangup


And here's how the clients are set in my db:

id: 4
  name: 660
ipaddr: 1.1.1.1
  port: 5060
regseconds: 1407320692
   defaultuser: 660
   fullcontact: sip:660@1.1.1.1:5060
 regserver:
 useragent:
lastms: 0
  host: dynamic
  type: friend
   context: default
  deny: 0.0.0.0/0.0.0.0
permit: 1.1.1.1
secret: NULL
 md5secret: NULL
  avpf: yes
 force_avp: yes
icesupport: yes
   directmedia: no
encryption: yes
   nat: force_rport,comedia
 callgroup: NULL
   pickupgroup: NULL
  language: NULL
  disallow: NULL
 allow: NULL
setvar: NULL
  callerid: NULL
  amaflags: NULL
  v

Re: [asterisk-users] different callerid for channels

[royj]

trying 
https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers+Specification

06.08.2014, 10:13, "r...@yandex.ru" :
> Hi, all.
>
> Is there any chance to set individual CALLERID(num) for channels SIP/peer1, 
> SIP/peer2 in a call Dial(SIP/peer1&SIP/peer2). There is an option to use 
> Dial(SIP/peer1&SIP/peer2,,M(set_callerid)), but the macro will be launched 
> after the channel answered. Not really want to use local channel because of 
> not quite usable cdr.
> Thanks.
>
> --
> _
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_
-- 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