Re: [asterisk-users] Call List Campaign to an IVR

[Pete Mundy]

On 2/02/2017, at 9:52 pm, A J Stiles  wrote:
> 
> but in simple solidarity with everyone who has ever 
> been pissed off by a machine-initiated spam marketing phone call at an 
> inappropriate moment, I am not going to tell you how to do it.
> 

Hat-tip to you, AJ :)

Pete



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[asterisk-users] Problem with rport (CGNAT) going from Linux kernel 3.16 to 4.9

[martin f krafft]

Hello,

I operate an Asterisk server (v11.13.1) on Debian stable, and it's
rock-solid. The other day, however, I accidentally upgraded the
kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped
working.

Below you can find my analysis while running the 4.9.0 kernel. 888
is a simply Echo() extension. I am calling it from a phone behind
carrier-grade NAT ("mtvic-main"). The problem is that the Asterisk
server sends RTP to the 100.64.0.0/10 address I have on the internal
side of NAT, even though the Asterisk server correctly (?)
transports the actual socket on the outside via rport (cf. the 401
Unauth response).

Once I boot back into 3.16.0, it all works again. I didn't capture
any logs yet, but since audio works, I am led to believe that the
100.64.0.0/10 address is not being used.

Right now it works, but eventually, the kernel upgrade will be
required. It's possible that a newer Asterisk will work with the v4
kernel, but in any case I'd be interested in finding out the root of
the problem at hand.

Any hints appreciated. Thank you!


>>> sip.conf <<<
[general]
nat=auto_force_rport,auto_comedia

[mtvic-main]
md5secret=xxx
context=mtvic-in-main
callerid="Martin in windy Wellington <60>"
dtmfmode=rfc2833
context=from-office
type=friend
directmedia=no
host=dynamic
nat=force_rport,comedia

# sip show peer output below
>>> /sip.conf <<<



>>> debug output <<<
[Feb  2 08:35:24] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:24] INVITE sip:8...@madduck.net;user=phone SIP/2.0
[Feb  2 08:35:24] Via: SIP/2.0/UDP 
100.64.45.19:5865;branch=z9hG4bK2c95e270486c659f91f1baa7712ebc80;rport
[Feb  2 08:35:24] From: "Penny & Martin / Wellington" 
;tag=4132889942
[Feb  2 08:35:24] To: 
[Feb  2 08:35:24] Call-ID: 4239363066@192_168_15_112
[Feb  2 08:35:24] CSeq: 2 INVITE
[Feb  2 08:35:24] Contact: 
[Feb  2 08:35:24] Max-Forwards: 70
[Feb  2 08:35:24] User-Agent: S685IP/02227000
[Feb  2 08:35:24] Supported: replaces
[Feb  2 08:35:24] Allow-Events: message-summary, refer
[Feb  2 08:35:24] Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, 
SUBSCRIBE, NOTIFY
[Feb  2 08:35:24] Content-Type: application/sdp
[Feb  2 08:35:24] Content-Length: 375
[Feb  2 08:35:24] 
[Feb  2 08:35:24] v=0
[Feb  2 08:35:24] o=mtvic-main 8602 68 IN IP4 100.64.45.19
[Feb  2 08:35:24] s=Mapping
[Feb  2 08:35:24] c=IN IP4 100.64.45.19
[Feb  2 08:35:24] t=0 0
[Feb  2 08:35:24] m=audio 8602 RTP/AVP 9 8 0 96 97 2 18 101
[Feb  2 08:35:24] a=rtpmap:9 G722/8000
[Feb  2 08:35:24] a=rtpmap:8 PCMA/8000
[Feb  2 08:35:24] a=rtpmap:0 PCMU/8000
[Feb  2 08:35:24] a=rtpmap:96 G726-32/8000
[Feb  2 08:35:24] a=rtpmap:97 AAL2-G726-32/8000
[Feb  2 08:35:24] a=rtpmap:2 G726-32/8000
[Feb  2 08:35:24] a=rtpmap:18 G729/8000
[Feb  2 08:35:24] a=fmtp:18 annexb=no
[Feb  2 08:35:24] a=rtpmap:101 telephone-event/8000
[Feb  2 08:35:24] a=fmtp:101 0-16
[Feb  2 08:35:24] <->
[Feb  2 08:35:24] --- (14 headers 16 lines) ---
[Feb  2 08:35:24] Sending to 219.88.239.74:43525 (NAT)
[Feb  2 08:35:24] Sending to 219.88.239.74:43525 (NAT)
[Feb  2 08:35:24] Using INVITE request as basis request - 
4239363066@192_168_15_112
[Feb  2 08:35:24] Found peer 'mtvic-main' for 'mtvic-main' from 
219.88.239.74:43525
[Feb  2 08:35:24] 
[Feb  2 08:35:24] <--- Reliably Transmitting (NAT) to 219.88.239.74:43525 --->
[Feb  2 08:35:24] SIP/2.0 401 Unauthorized
[Feb  2 08:35:24] Via: SIP/2.0/UDP 
100.64.45.19:5865;branch=z9hG4bK2c95e270486c659f91f1baa7712ebc80;received=219.88.239.74;rport=43525
[Feb  2 08:35:24] From: "Penny & Martin / Wellington" 
;tag=4132889942
[Feb  2 08:35:24] To: ;tag=as39e92fd2
[Feb  2 08:35:24] Call-ID: 4239363066@192_168_15_112
[Feb  2 08:35:24] CSeq: 2 INVITE
[Feb  2 08:35:24] Server: Asterisk PBX
[Feb  2 08:35:24] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY, INFO, PUBLISH, MESSAGE
[Feb  2 08:35:24] Supported: replaces, timer
[Feb  2 08:35:24] WWW-Authenticate: Digest algorithm=MD5, realm="madduck.net", 
nonce="2a4c925b"
[Feb  2 08:35:24] Content-Length: 0
[Feb  2 08:35:24] 
[Feb  2 08:35:24] 
[Feb  2 08:35:24] <>
[Feb  2 08:35:24] Scheduling destruction of SIP dialog 
'4239363066@192_168_15_112' in 32000 ms (Method: INVITE)
[Feb  2 08:35:25] 
[Feb  2 08:35:25] <--- SIP read from UDP:219.88.239.74:43525 --->
[Feb  2 08:35:25] ACK sip:8...@madduck.net;user=phone SIP/2.0
[Feb  2 08:35:25] Via: SIP/2.0/UDP 
100.64.45.19:5865;branch=z9hG4bK2c95e270486c659f91f1baa7712ebc80;rport
[Feb  2 08:35:25] From: "Penny & Martin / Wellington" 
;tag=4132889942
[Feb  2 08:35:25] To: ;tag=as39e92fd2
[Feb  2 08:35:25] Call-ID: 4239363066@192_168_15_112
[Feb  2 08:35:25] CSeq: 2 ACK
[Feb  2 08:35:25] Contact: 
[Feb  2 08:35:25] Max-Forwards: 70
[Feb  2 08:35:25] User-Agent: 

[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

[Motty Cruz]

Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error: 

 

-- SIP/voipeer-084b redirecting info has changed, passing it to
SIP/1007-084a

-- SIP/voipeer-084b is busy

  == Everyone is busy/congested at this time (1:1/0/0)

-- Timeout on SIP/1007-084a

-- Executing [t@phones:1] Playback("SIP/1007-084a", "goodbye") in
new stack

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

--  Playing 'goodbye.slin' (language 'en')

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

-- Executing [t@phones:2] Hangup("SIP/1007-084a", "") in new stack

 

Sip.conf 

[1007]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1007>
disallow=all
allow=ulaw
allow=alaw
username=1007
secret=X
dtmfmode=rfc2833
host=dynamic
mailbox=1007@default
nat=force_rport,comedia

 

Is it a codec issue? Or missed configuration? Asterisk does not know how to
translate busy signal. 

Your help is appreciated!

Thanks,

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Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[George Joseph]

On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy  wrote:

> Yes, from_user was set, removing those entries solved the problem.
>
> Can someone please explain to me the correct use for fromuser field?
>

from_user forces the user portion of the From header to a specific value on
calls that go TO the device represented by the endpoint.  Most often it's
used with a service provider when the service provider requires that all
calls it accepts have some sort of account identifier in the From header
instead of the original caller's info.  I can't think of a scenario where
you'd need to use from_user with a phone.


>
> thanks
> Zakir
>
>
> On Wednesday, February 1, 2017 8:00 PM, "asterisk-users-request@lists.
> digium.com"  wrote:
>
>
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>   1. asterisk  callerid issue PJSIP Realtime (Zakir Mahomedy)
>   2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)
>
>
> --
>
> Message: 1
> Date: Wed, 1 Feb 2017 13:50:57 + (UTC)
> From: Zakir Mahomedy 
> To: "asterisk-users@lists.digium.com"
> 
> Subject: [asterisk-users] asterisk  callerid issue PJSIP Realtime
> Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
> I recently rolled out a new server with asterisk 14. ?On the Called user
> phone, the caller ID is the same as the Called User.
> eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the
> ext 405 phone displaying 405.
>
>
> We are using realtime PJSIP, I set the callerid field in the database but
> no luck.?
> - Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP
> CLID"") in new stack
> - Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = ?"ross"
> <406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f",
> "PJSIP/405") in new stack
> In the above dialplan, the callerid is been taken from the database PJSIP
> endpoints.?
> Here is the sip debugger files
> INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 192.168.1.82:5060
> ;branch=z9hG4bK714210067;rportFrom: "zak" 
> ;tag=2071662084To:
> Call-ID: 50172054-506...@bjc.bgi.B.ICCSeq: 21
> INVITEContact: "zak" Authorization: Digest
> username="406", realm="asterisk", nonce="1485956409/
> e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response="
> ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017",
> opaque="50d490d233efd03e", qop=auth, nc=0003
>
> INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP
> 197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
> ;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To:  405@192.168.1.209;ob>Contact: Call-ID:
> b4a83465-9105-4c70-9da1-11f410c37657
>
> <--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767
> --->SIP/2.0 180 RingingVia: SIP/2.0/UDP 197.245.99.113:5060;rport=
> 5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
> f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: ;tag=
> 77ea8869-273a-4f65-8128-e334b445f970To: ;
> tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 INVITEContact:  405@192.168.1.209:36767;ob>Allow: PRACK, INVITE, ACK, B
>
>
> ?ParameterName ? ? ? ? ? ? ? ? ? ? ?: ParameterValue?===
> ==?callerid ? ? ? ? ? ? ? ? ? ? ?
> ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? : allowed?callerid_tag
> ? ? ? ? ? ? ? ? ? ?:
> Zakir
>
> -- next part --
> An HTML attachment was scrubbed...
> URL:  attachments/20170201/ede9ff18/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 1 Feb 2017 08:52:59 -0700
> From: George Joseph 
> To: Zakir Mahomedy ,  Asterisk Users Mailing List
> - Non-Commercial Discussion 
> Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
> Message-ID:
> 

Re: [asterisk-users] PJSIP hangupcause how to

[Joshua Colp]

On Thu, Feb 2, 2017, at 08:11 AM, Saint Michael wrote:
> if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?
> in old SIP channel, we had ${HASH(SIP_CAUSE,)}
> but in PJSIP it has to be the outbound channel, which is gone when the
> control returns to the calling channel.

This functionality was replaced quite some time ago with HANGUPCAUSE[1].

[1] https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause

-- 
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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[asterisk-users] PJSIP hangupcause how to

[Saint Michael]

if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?
in old SIP channel, we had ${HASH(SIP_CAUSE,)}
but in PJSIP it has to be the outbound channel, which is gone when the
control returns to the calling channel.
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Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[Zakir Mahomedy]

Yes, from_user was set, removing those entries solved the problem.
Can someone please explain to me the correct use for fromuser field?
thanksZakir 

On Wednesday, February 1, 2017 8:00 PM, 
"asterisk-users-requ...@lists.digium.com" 
 wrote:
 

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than "Re: Contents of asterisk-users digest..."


Today's Topics:

  1. asterisk  callerid issue PJSIP Realtime (Zakir Mahomedy)
  2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)


--

Message: 1
Date: Wed, 1 Feb 2017 13:50:57 + (UTC)
From: Zakir Mahomedy 
To: "asterisk-users@lists.digium.com"
    
Subject: [asterisk-users] asterisk  callerid issue PJSIP Realtime
Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"

I recently rolled out a new server with asterisk 14. ?On the Called user phone, 
the caller ID is the same as the Called User.
eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the ext 
405 phone displaying 405.


We are using realtime PJSIP, I set the callerid field in the database but no 
luck.?
- Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP CLID"") 
in new stack
- Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = ?"ross" 
<406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f", 
"PJSIP/405") in new stack
In the above dialplan, the callerid is been taken from the database PJSIP 
endpoints.?
Here is the sip debugger files
INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 
192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak" 
;tag=2071662084To: Call-ID: 
50172054-506...@bjc.bgi.B.ICCSeq: 21 INVITEContact: "zak" 
Authorization: Digest username="406", 
realm="asterisk", nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa", 
uri="sip:405@192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d", 
algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e", qop=auth, 
nc=0003

INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP 
197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
 ;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To: 
Contact: Call-ID: 
b4a83465-9105-4c70-9da1-11f410c37657

<--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->SIP/2.0 
180 RingingVia: SIP/2.0/UDP 
197.245.99.113:5060;rport=5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
 f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: 
;tag=77ea8869-273a-4f65-8128-e334b445f970To: 
;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 
INVITEContact: Allow: PRACK, INVITE, ACK, B


?ParameterName ? ? ? ? ? ? ? ? ? ? ?: 
ParameterValue?=?callerid
 ? ? ? ? ? ? ? ? ? ? ? ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? : 
allowed?callerid_tag ? ? ? ? ? ? ? ? ? ?:
Zakir
 
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--

Message: 2
Date: Wed, 1 Feb 2017 08:52:59 -0700
From: George Joseph 
To: Zakir Mahomedy ,  Asterisk Users Mailing List
    - Non-Commercial Discussion 
Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
Message-ID:
    

[asterisk-users] asterisk13+app_queue scalability

[marek cervenka]

hi,

i have similar problem to 
https://issues.asterisk.org/jira/browse/ASTERISK-25806


do you know about some workarounds/patches for better scalability?

thanks

marek



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Re: [asterisk-users] Call List Campaign to an IVR

[A J Stiles]

On Thursday 02 Feb 2017, Amelye Chatila wrote:
> Hi,
> I need to make calls to a list of numbers one at a time and once the user
> pick the phone connects to an IVR where I can get few data, after  a call
> finishes the 2nd number get called and so forth.
> 
> I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does
> not seem to fill this need. I'm now looking GoAutodial & AsterCC.

It is possible to do everything you require with Dialplan and Bash scripting.  
It is also  (1)  highly illegal and  (2)  morally beneath contempt.  So, not 
out of any desire to avoid a charge of "conspiracy to misuse an electronic 
communications network" but in simple solidarity with everyone who has ever 
been pissed off by a machine-initiated spam marketing phone call at an 
inappropriate moment, I am not going to tell you how to do it.


-- 
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] Call List Campaign to an IVR

[Amelye Chatila]

To be more clear I need unattended campaign
On Feb 2, 2017 11:26 AM, "Amelye Chatila"  wrote:

> Hi,
>
>
> I need to make calls to a list of numbers one at a time and once the user
> pick the phone connects to an IVR where I can get few data, after  a call
> finishes the 2nd number get called and so forth.
>
> I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix
> does not seem to fill this need. I'm now looking GoAutodial & AsterCC.
>
> Anyone with an idea to solve this issue I 'll be thankful.
>
>
> Regards,
>
> Amelye
>
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[asterisk-users] Call List Campaign to an IVR

[Amelye Chatila]

Hi,


I need to make calls to a list of numbers one at a time and once the user
pick the phone connects to an IVR where I can get few data, after  a call
finishes the 2nd number get called and so forth.

I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does
not seem to fill this need. I'm now looking GoAutodial & AsterCC.

Anyone with an idea to solve this issue I 'll be thankful.


Regards,

Amelye
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