date:20170213

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Anas Moiz

Ok, I also tried to hangup directly through dialplan, it doesn't work.

  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b0", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b0'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b1", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b1'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b2", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b2'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b3", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b3'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b4", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b4'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b5", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b5'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b6", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b6'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b7", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b7'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b8", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b8'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0b9", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0b9'
  == Using SIP RTP CoS mark 5
-- Executing [12023300643@default:1]
Hangup("SIP/66.226.76.70-d0ba", "41") in new stack
  == Spawn extension (default, 12023300643, 1) exited non-zero on
'SIP/66.226.76.70-d0ba'



On Tue, Feb 14, 2017 at 6:03 AM, Joshua Colp  wrote:

> On Mon, Feb 13, 2017, at 08:58 PM, Anas Moiz wrote:
> > Yes Joshua, Its SIP and but the problem is I have tried everything but it
> > doesn't seem to work.
> >
> > In the SIP Trace I can see that I am sending 503 Service Unavailable as a
> > response.
> >
> > You can check the SIP trace attached below:
> >
> > 162.243.107.173:5060 -> 66.226.76.70:5060
> > SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP
> > 66.226.76.70:5060;branch=
> > z9hG4bK643.e44ea565.0;received=66.226.76.70;rport=5060 Via: SIP/2.0/UDP
> > 74.117.36.136;received=74.117.36.136;rport=5060;branch=
> z9hG4bKHBe9cmy3QX2Se
> > From: ;tag=5H54caUKre8gc To: <
> > sip:12023300643@162.243.107.173:5060>;tag=as61c328a0 Call-ID:
> > 15-8824754a-f58560c9-335bcd48-45558f71 CSeq: 103180201 INVITE Server:
> > user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> > NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length:
> > 0
>
> You would need to determine what will stop the remote server from
> sending you the call again. Once you do that and can provide what it is
> then we can figure out how to get Asterisk to do that. As it is the
> problem isn't Asterisk, it is what is sending you the call.
>
> --
> 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
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Joshua Colp

On Mon, Feb 13, 2017, at 08:58 PM, Anas Moiz wrote:
> Yes Joshua, Its SIP and but the problem is I have tried everything but it
> doesn't seem to work.
> 
> In the SIP Trace I can see that I am sending 503 Service Unavailable as a
> response.
> 
> You can check the SIP trace attached below:
> 
> 162.243.107.173:5060 -> 66.226.76.70:5060
> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP
> 66.226.76.70:5060;branch=
> z9hG4bK643.e44ea565.0;received=66.226.76.70;rport=5060 Via: SIP/2.0/UDP
> 74.117.36.136;received=74.117.36.136;rport=5060;branch=z9hG4bKHBe9cmy3QX2Se
> From: ;tag=5H54caUKre8gc To: <
> sip:12023300643@162.243.107.173:5060>;tag=as61c328a0 Call-ID:
> 15-8824754a-f58560c9-335bcd48-45558f71 CSeq: 103180201 INVITE Server:
> user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
> NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length:
> 0

You would need to determine what will stop the remote server from
sending you the call again. Once you do that and can provide what it is
then we can figure out how to get Asterisk to do that. As it is the
problem isn't Asterisk, it is what is sending you the call.

-- 
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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Anas Moiz

Yes Joshua, Its SIP and but the problem is I have tried everything but it
doesn't seem to work.

In the SIP Trace I can see that I am sending 503 Service Unavailable as a
response.

You can check the SIP trace attached below:

162.243.107.173:5060 -> 66.226.76.70:5060
SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 66.226.76.70:5060;branch=
z9hG4bK643.e44ea565.0;received=66.226.76.70;rport=5060 Via: SIP/2.0/UDP
74.117.36.136;received=74.117.36.136;rport=5060;branch=z9hG4bKHBe9cmy3QX2Se
From: ;tag=5H54caUKre8gc To: <
sip:12023300643@162.243.107.173:5060>;tag=as61c328a0 Call-ID:
15-8824754a-f58560c9-335bcd48-45558f71 CSeq: 103180201 INVITE Server:
user_Anas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
-- 
_
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New to Asterisk? Start here:
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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Joshua Colp

On Mon, Feb 13, 2017, at 05:46 PM, Anas Moiz wrote:
> Hi Everyone,
> 
> I am dealing with a problem for now and its really annoying.
> 
> I want to hangup calls from AGI but it seems that my AGI is not rejecting
> the calls properly.
> 
> {
> $agi->verbose("number-not-in-service");
> $agi->exec("Congestion","1");
> $agi->hangup();
> exit;
> }
> 
> with the above logic, all of my calls should be rejected and should be
> disconnected instantaneously.
> 
> But this doesn't seem to be happening, in asterisk CLI I can see
> that AGI is executing multiple times.
> 
> Can anyone tell what I am doing wrong?

Is this SIP? If so what may be happening is that the system sending you
calls may not consider a 503 Service Unavailable (which Congestion will
send) to be a final response which terminates the call and thus send you
a call again, and again, and again, in hopes that you'll accept it.
Since that is behavior of the system sending you calls you would need to
determine if there is any SIP response which will provide the behavior
you need and then find the appropriate cause code to cause it to be
sent.

-- 
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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Certified Asterisk 13.13-cert1 Now Available

2017-02-13 Thread Asterisk Development Team

The Asterisk Development Team has announced the release of Certified Asterisk 
13.13-cert1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 13.13-cert1 resolves several issues reported 
by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Improvements made in this release:
---
 * ASTERISK-25063 - [patch]add X.509 subject alternative name
  support to Asterisk TLS support (Reported by Maciej Szmigiero)
 * ASTERISK-26558 - app_queue: add variable to know if the call is
  not answered after a queue (Reported by scgm11)
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
  (Reported by scgm11)
 * ASTERISK-26538 - codec_opus: Add sample to
  configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
 * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
  'ari set debug' CLI commands (Reported by Matt Jordan)
 * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
  blacklisting host subnets that are not involved in RTP (Reported
  by Michael Walton)
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
  translation codec. (Reported by Kevin Harwell)
 * ASTERISK-26289 - Announcer channels in ConfBridges cause
  inefficiencies (Reported by Mark Michelson)
 * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
  dialplan know what fax transport was used (Reported by Alexei
  Gradinari)
 * ASTERISK-26220 - Add support for noreturn function attributes.
  (Reported by Corey Farrell)
 * ASTERISK-22131 - Update the make dependencies script to pull,
  build, and install the correct pjproject (Reported by Matt
  Jordan)
 * ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
  (Reported by JoshE)
 * ASTERISK-26159 - res_hep: enabled by default and information
  sent to default address (Reported by Ross Beer)
 * ASTERISK-26088 - Investigate heavy memory utilization by
  res_pjsip_pubsub (Reported by Richard Mudgett)
 * ASTERISK-26059 - [patch]core: New channel variable FORWARDERNAME
  (Reported by Alexei Gradinari)
 * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
  "call_id" to contacts (Reported by Alexei Gradinari)
 * ASTERISK-26055 - [patch]res_pjsip: chatty verbose messages
  (Reported by Alexei Gradinari)
 * ASTERISK-26010 - [patch]func_odbc: single database connection
  should be optional (Reported by Alexei Gradinari)
 * ASTERISK-25994 - [patch]res_pjsip: module load priority
  (Reported by Alexei Gradinari)
 * ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
  by Alexei Gradinari)
 * ASTERISK-25835 - Authentication using 'Username' field from
  Digest (Reported by Ross Beer)
 * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
  performace (Reported by Alexei Gradinari)
 * ASTERISK-25865 - Message-Account Missing From PJSIP MWI
  (Reported by Ross Beer)
 * ASTERISK-25444 - [patch]Music On Hold Warning misleading
  (Reported by Conrad de Wet)

Bugs fixed in this release:
---
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
  hung up via ARI (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
  leading to stuck channels. (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
  (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
  manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
  (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
  request  to endpoint (Reported by Ross Beer)
 * ASTERISK-26621 - app_queue: Queue application does not ring
  members with Local interface (Reported by Jonas Kellens)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
  does not work. (Reported by Richard Mudgett)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
  function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26672 - Crash when setting remote address on RTP
  instance (Reported by Richard Mudgett)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
  bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  is invalid (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
  with frames leading to spammy log messages (Reported by Jonathan
  Rose)
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
  setting up new calls (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing

[asterisk-users] Asterisk 14.3.0 Now Available

2017-02-13 Thread Asterisk Development Team

The Asterisk Development Team has announced the release of Asterisk 14.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
  (Reported by Richard Mudgett)

Bugs fixed in this release:
---
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
  by nappsoft)
 * ASTERISK-26767 - ARI channelvars cause memory leak (Reported by
  SÃ©bastien Duthil)
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
  hung up via ARI (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
  leading to stuck channels. (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
  (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
  manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26704 - res_odbc.conf contains deprecated
  configuration: 'pooling', 'shared_connections', 'limit', and
  'idlecheck' options were replaced by 'max_connections'.
  (Reported by Anthony Messina)
 * ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref
  count trap tripped. (Reported by Richard Mudgett)
 * ASTERISK-21094 - MixMonitorMute mutes through stream if already
  slinear (e.g. Originate) (Reported by David Woolley)
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
  (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
  request  to endpoint (Reported by Ross Beer)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
  wrong byte order on Intel platform when using slin codec
  (Reported by Frankie Chin)
 * ASTERISK-26754 - build_tools: make_build_h does not handle \ in
  user name  (Reported by Kirill Katsnelson)
 * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite:
  fwrite() returned error: Broken pipe" (Reported by Kirill
  Katsnelson)
 * ASTERISK-26755 - app_queue: Random queues disappear on "core
  reload queue all" (Reported by Kirill Katsnelson)
 * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups"
  after match in .conf has no effect (Reported by Michael Maier)
 * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support
  for SRV (Reported by Joshua Colp)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
  does not work. (Reported by Richard Mudgett)
 * ASTERISK-26740 - voicemail API test: uses varlibdir instead of
  datadir for a sound file (Reported by Tzafrir Cohen)
 * ASTERISK-26739 - voicemail API test: confuses expected and
  actual values (Reported by Tzafrir Cohen)
 * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every
  sorcery memory cache populate (Reported by Ustinov Artem)
 * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
  (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
  (Reported by Aaron An)
 * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
  (Reported by Alexander Traud)
 * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
  (Reported by Alexander Traud)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
  function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26684 - res_pjsip: Various issues with compact SIP
  headers (Reported by Joshua Elson)
 * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
  Headers Enabled (Reported by JoshE)
 * ASTERISK-26672 - Crash when setting remote address on RTP
  instance (Reported by Richard Mudgett)
 * ASTERISK-26621 - app_queue: Queue application does not ring
  members with Local interface (Reported by Jonas Kellens)
 * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with
  MWI wasn't registered (Reported by Michael Kuron)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
  bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  is invalid (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
  with frames leading to spammy log messages (Reported by Jonathan
  Rose)
 * ASTERISK-26653 - pjproject_bundled doesn't verify already
  downloaded tarballs (Reported by George Joseph)
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
  setting up new calls (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
  line (Reported by JÃ¸rgen H)
 *

[asterisk-users] Asterisk 13.14.0 Now Available

2017-02-13 Thread Asterisk Development Team

The Asterisk Development Team has announced the release of Asterisk 13.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
  (Reported by Richard Mudgett)

Bugs fixed in this release:
---
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
  by nappsoft)
 * ASTERISK-26704 - res_odbc.conf contains deprecated
  configuration: 'pooling', 'shared_connections', 'limit', and
  'idlecheck' options were replaced by 'max_connections'.
  (Reported by Anthony Messina)
 * ASTERISK-21094 - MixMonitorMute mutes through stream if already
  slinear (e.g. Originate) (Reported by David Woolley)
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
  hung up via ARI (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
  leading to stuck channels. (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
  (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
  manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
  (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
  request  to endpoint (Reported by Ross Beer)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
  wrong byte order on Intel platform when using slin codec
  (Reported by Frankie Chin)
 * ASTERISK-26754 - build_tools: make_build_h does not handle \ in
  user name  (Reported by Kirill Katsnelson)
 * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite:
  fwrite() returned error: Broken pipe" (Reported by Kirill
  Katsnelson)
 * ASTERISK-26755 - app_queue: Random queues disappear on "core
  reload queue all" (Reported by Kirill Katsnelson)
 * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups"
  after match in .conf has no effect (Reported by Michael Maier)
 * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support
  for SRV (Reported by Joshua Colp)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
  does not work. (Reported by Richard Mudgett)
 * ASTERISK-26740 - voicemail API test: uses varlibdir instead of
  datadir for a sound file (Reported by Tzafrir Cohen)
 * ASTERISK-26739 - voicemail API test: confuses expected and
  actual values (Reported by Tzafrir Cohen)
 * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every
  sorcery memory cache populate (Reported by Ustinov Artem)
 * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
  (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
  (Reported by Aaron An)
 * ASTERISK-26672 - Crash when setting remote address on RTP
  instance (Reported by Richard Mudgett)
 * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
  (Reported by Alexander Traud)
 * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
  (Reported by Alexander Traud)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
  function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26684 - res_pjsip: Various issues with compact SIP
  headers (Reported by Joshua Elson)
 * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
  Headers Enabled (Reported by JoshE)
 * ASTERISK-26621 - app_queue: Queue application does not ring
  members with Local interface (Reported by Jonas Kellens)
 * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with
  MWI wasn't registered (Reported by Michael Kuron)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
  bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  is invalid (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
  with frames leading to spammy log messages (Reported by Jonathan
  Rose)
 * ASTERISK-26653 - pjproject_bundled doesn't verify already
  downloaded tarballs (Reported by George Joseph)
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
  setting up new calls (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
  line (Reported by JÃ¸rgen H)
 * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
  aors (Reported by George Joseph)
 * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not
  Exist when transaction branch parameter

[asterisk-users] CALLS NOT HANGING UP THROUGH AGI

2017-02-13 Thread Anas Moiz

Hi Everyone,

I am dealing with a problem for now and its really annoying.

I want to hangup calls from AGI but it seems that my AGI is not rejecting
the calls properly.

{
$agi->verbose("number-not-in-service");
$agi->exec("Congestion","1");
$agi->hangup();
exit;
}

with the above logic, all of my calls should be rejected and should be
disconnected instantaneously.

But this doesn't seem to be happening, in asterisk CLI I can see
that AGI is executing multiple times.

Can anyone tell what I am doing wrong?

Thanks.
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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

2017-02-13 Thread Pete Mundy


+1! This sounds an awful lot like an ALG doing it best to 'help'...


> On 14/02/2017, at 6:38 am, Israel Gottlieb  wrote:
> 
> Disable all sip alg/helpers in the router



smime.p7s
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Re: [asterisk-users] ATA Adapter YGW30 - manual

2017-02-13 Thread Nathan Anderson

Behold: The Wayback Machine.  Link to manual: 
http://web.archive.org/web/20070224144946/http://www.yntx.com/files/YGW30en.rar

Manual says user/pass is root/test.

-- Nathan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neu pat
Sent: Monday, February 13, 2017 9:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ATA Adapter YGW30 - manual

I have one: YGW30 1FXS,1FXO SIP ATA unit
it was made by company Yuxin I think they are no longer in business.

I forgot the default user name / password for log-in.  Does anybody
know what was the default login or have a manual?

-- 
Regards,
Joseph

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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

2017-02-13 Thread Israel Gottlieb

 Disable all sip alg/helpers in the router


  Original Message  
From: andregronwal...@gmail.com
Sent: February 13, 2017 6:40 PM
To: asterisk-users@lists.digium.com
Reply-to: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Some further information:
asterisk version: 13.13.1, pjsip (pjproject) 2.5.5


regards,
andre

Am 13.02.2017 um 17:32 schrieb Andre Gronwald:
> Hi all,
> I have a strange issue, with a some kind complicate architecture...
> A router of our internet provider is in front of another bintec rs353j 
> router, at which my freepbx installation is located. However, NAT etc. 
> seems to work fine.
> BUT: Something is not working...:
> When registering my sip-trunk towards my provider (3 different 
> providers, all behave comparable), everything works at first. Calls 
> are possible. But after some time, when the next REGISTER happens, the 
> answer of my provider is sent towards the wrong port. My freePBX 
> listens on 55060, where the first registration request are answered as 
> they should. in the second registration request wrong ports are used. 
> besides this, Header "Expires" is set to "0" and no "Allows" are 
> listed...
> [...]

> Any suggestions how to fix this? Or at least any idea what causes this?
>
> regards,
> andre


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Re: [asterisk-users] Disallow CALLS without registry

2017-02-13 Thread Антон Сацкий

sorry
NoOP(${DB_EXISTS(SIP/Registry/${CHANNEL(peername)})});

2017-02-13 19:31 GMT+02:00 Антон Сацкий :

> THINK i found a solution
>
> NoOP(${DB_EXISTS(/SIP/Registry/${CHANNEL(peername)})});
>
> THANKS TO ALL
>
> 2017-02-12 12:34 GMT+02:00 Frank Vanoni :
>
>> On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote:
>>
>> > > sip.conf configuration
>> > > In the [general] section, define:
>> > > [general]
>> > > ...
>> > > allowguest=no
>> > > alwaysauthreject=yes
>> > > ...
>>
>>
>> >
>>
>> With the above configuration on my Asterisk, I obtain the following
>> result:
>>
>> - if the phone is registered to Asterisk, I can place any call according
>> to the dial plan.
>>
>> - if the phone is NOT registered and I try to place a call, the phone
>> obtains a "403 forbidden" at any calling attempt.
>>
>>
>> Now, English is not my native language, but as far as I can understand,
>> "forbidden" means "not allowed" or "disallowed".
>>
>>
>>
>>
>> --
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best regards
> Antony
> tel.   +380669197533 <+380%2066%20919%207533>
> tel2. +380636564340 <+380%2063%20656%204340>
> Paypal http://paypal.me/Satskiy
> 
> satski...@gmail.com 
>



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

satski...@gmail.com 
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Re: [asterisk-users] Disallow CALLS without registry

2017-02-13 Thread Антон Сацкий

THINK i found a solution

NoOP(${DB_EXISTS(/SIP/Registry/${CHANNEL(peername)})});

THANKS TO ALL

2017-02-12 12:34 GMT+02:00 Frank Vanoni :

> On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote:
>
> > > sip.conf configuration
> > > In the [general] section, define:
> > > [general]
> > > ...
> > > allowguest=no
> > > alwaysauthreject=yes
> > > ...
>
>
> >
>
> With the above configuration on my Asterisk, I obtain the following
> result:
>
> - if the phone is registered to Asterisk, I can place any call according
> to the dial plan.
>
> - if the phone is NOT registered and I try to place a call, the phone
> obtains a "403 forbidden" at any calling attempt.
>
>
> Now, English is not my native language, but as far as I can understand,
> "forbidden" means "not allowed" or "disallowed".
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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

satski...@gmail.com 
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[asterisk-users] ATA Adapter YGW30 - manual

2017-02-13 Thread neu pat

I have one: YGW30 1FXS,1FXO SIP ATA unit
it was made by company Yuxin I think they are no longer in business.

I forgot the default user name / password for log-in.  Does anybody
know what was the default login or have a manual?

-- 
Regards,
Joseph

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Re: [asterisk-users] First SIP-registering succeeds, second doesn't

2017-02-13 Thread Andre Gronwald


Some further information:
asterisk version: 13.13.1, pjsip (pjproject) 2.5.5


regards,
andre

Am 13.02.2017 um 17:32 schrieb Andre Gronwald:

Hi all,
I have a strange issue, with a some kind complicate architecture...
A router of our internet provider is in front of another bintec rs353j 
router, at which my freepbx installation is located. However, NAT etc. 
seems to work fine.

BUT: Something is not working...:
When registering my sip-trunk towards my provider (3 different 
providers, all behave comparable), everything works at first. Calls 
are possible. But after some time, when the next REGISTER happens, the 
answer of my provider is sent towards the wrong port. My freePBX 
listens on 55060, where the first registration request are answered as 
they should. in the second registration request wrong ports are used. 
besides this, Header "Expires" is set to "0" and no "Allows" are 
listed...

[...]



Any suggestions how to fix this? Or at least any idea what causes this?

regards,
andre



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[asterisk-users] First SIP-registering succeeds, second doesn't

2017-02-13 Thread Andre Gronwald


Hi all,
I have a strange issue, with a some kind complicate architecture...
A router of our internet provider is in front of another bintec rs353j 
router, at which my freepbx installation is located. However, NAT etc. 
seems to work fine.

BUT: Something is not working...:
When registering my sip-trunk towards my provider (3 different 
providers, all behave comparable), everything works at first. Calls are 
possible. But after some time, when the next REGISTER happens, the 
answer of my provider is sent towards the wrong port. My freePBX listens 
on 55060, where the first registration request are answered as they 
should. in the second registration request wrong ports are used. besides 
this, Header "Expires" is set to "0" and no "Allows" are listed...


Good case:
2017/02/10 20:40:59.236563 192.193.194.99:55060 -> 217.10.79.9:5060
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj7d031cfc-7602-4ac1-a264-bcf8d267500b

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2367 REGISTER
Contact:
Expires: 300
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE

Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Content-Length: 0

2017/02/10 20:40:59.300873 217.10.79.9:5060 -> 192.193.194.99:55060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj7d031cfc-7602-4ac1-a264-bcf8d267500b

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de;tag=86e53dd608d1c001e0b8060625977563.11b6
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2367 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", 
nonce="WJ4Yd1ieF0tFmpj4o617dwQB2X5d9sKR"

Content-Length: 0

2017/02/10 20:40:59.301187 192.193.194.99:55060 -> 217.10.79.9:5060
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj5fb9a513-f49b-4823-8c63-6f0a38314d89

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2368 REGISTER
Contact:
Expires: 300
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE

Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Authorization: Digest username="custumoerIDe0", realm="sipgate.de", 
nonce="WJ4Yd1ieF0tFmpj4o617dwQB2X5d9sKR", uri="sip:sipgate.de:5060", 
response="1067666a187dd3413fcecede9820e87c"

Content-Length: 0

2017/02/10 20:40:59.367186 217.10.79.9:5060 -> 192.193.194.99:55060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj5fb9a513-f49b-4823-8c63-6f0a38314d89

From: sip:custumoeri...@sipgate.de;tag=b14c0d37-ef26-4dc5-b112-caf0c12a51f1
To: sip:custumoeri...@sipgate.de;tag=86e53dd608d1c001e0b8060625977563.262c
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2368 REGISTER
Contact: ;expires=300
Content-Length: 0

Bad case:
2017/02/10 20:43:08.324919 192.193.194.99:55060 -> 217.10.79.9:5060
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj9fb980fb-f4a3-465b-80bd-f25e668b723c

From: sip:custumoeri...@sipgate.de;tag=a5226407-51ae-41c9-845e-4791428aa44f
To: sip:custumoeri...@sipgate.de
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2369 REGISTER
Contact:
Expires: 0
Max-Forwards: 70
User-Agent: FPBX-13.0.190.12(13.13.1)
Content-Length: 0

2017/02/10 20:43:08.387098 217.10.79.9:5060 -> 192.193.194.99:61276
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
99.88.77.66:55060;rport;branch=z9hG4bKPj9fb980fb-f4a3-465b-80bd-f25e668b723c

From: sip:custumoeri...@sipgate.de;tag=a5226407-51ae-41c9-845e-4791428aa44f
To: sip:custumoeri...@sipgate.de;tag=86e53dd608d1c001e0b8060625977563.1e67
Call-ID: 6cf97519-e593-4a7e-bfe4-29141604665d
CSeq: 2369 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", 
nonce="WJ4Y+FieF8wCvpDGuyqz40rHntgzb6xy"

Content-Length: 0

Any suggestions how to fix this? Or at least any idea what causes this?

regards,
andre


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Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Re: [asterisk-users] CALLS NOT HANGING UP THROUGH AGI

[asterisk-users] Certified Asterisk 13.13-cert1 Now Available

[asterisk-users] Asterisk 14.3.0 Now Available

[asterisk-users] Asterisk 13.14.0 Now Available

[asterisk-users] CALLS NOT HANGING UP THROUGH AGI

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Re: [asterisk-users] ATA Adapter YGW30 - manual

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

Re: [asterisk-users] Disallow CALLS without registry

Re: [asterisk-users] Disallow CALLS without registry

[asterisk-users] ATA Adapter YGW30 - manual

Re: [asterisk-users] First SIP-registering succeeds, second doesn't

[asterisk-users] First SIP-registering succeeds, second doesn't

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