Re: [asterisk-users] Call Quality Measuring

[Sevana Oy]

Hi Patrick,

You are welcome to try our tools out for active and passive voice quality
measurement tools. It's waveform analysis (like PESQ or POLQA) and VoIP
metrics analysis (like G.107 E-model and other metrics).

You can read more at http://www.sevana.biz
or older site http://www.sevana.fi


On Tue, Mar 31, 2015 at 1:16 PM, Patrick Beaumont <
p.beaum...@hatsoffsoftware.co.uk> wrote:

> Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor
> in the near future.
>
> So can I assume from the lack of discussion nobody is using the “sip show
> channelstats” stuff?
>
> Regards,
> Patrick.
>
> On 31/03/2015 08:23, "Olivier"  wrote:
>
> >Some SIP hardphones (Polycom) or softphones (Counterpath) embed a
> >module that metter MOS.
> >
> >
> >Regards
> >
> >2015-03-25 14:21 GMT+01:00 Patrick Beaumont
> >:
> >> Hi everyone.
> >>
> >> We regularly get customers complaining about call quality issues. Most
> >>of
> >> the time it turns out to be their own broadband. Very occasionally
> >>server
> >> load. Does anyone have any advice or links to advice on measuring call
> >> quality?
> >>
> >> I’ve been playing around with “sip show channelstats” but can’t other
> >>than
> >> measuring the packet loss I don’t really know what I’m supposed to be
> >> looking for in order to say “ah ha! that’s the problem!”. I also don’t
> >> know what it’s limits are. Will the stats in “sip show channelstats”
> >>show
> >> a customer using a torrent client and saturating their own broadband
> >> connection?
> >>
> >> Regards,
> >> Patrick.
> >>
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[asterisk-users] ReceiveFax() fails over Dial()

[Dominique Haeber]

Hi all,

since asterisk 11 (1.6 was okay) failed the ReceiveFax-Application
when it called about "Dial" and a Local-Channel.

Directly from external to FaxReceive is no problem.

Cut from cli:

[...]
[Apr  1 11:12:31] -- Executing [s@macro-redirection:85] 
Dial("SIP/access-trunk-0001", "Local/0XXX40@x-xxx-companyX/n") in 
new stack
[...]
[Apr  1 11:12:31] -- Executing [s@macro-faxproceed:10] 
Set("Local/0XX40@x-xxx-companyX-0001;2", "FAXOPT(gateway)=no") in 
new stack
[Apr  1 11:12:31] -- Executing [s@macro-faxproceed:11] 
Answer("Local/0XX40@x-xxx-companyX-0001;2", "") in new stack
[Apr  1 11:12:31] -- Local/0XX40@x-xxx-companyX-0001;1 answered 
SIP/access-trunk-0001
[Apr  1 11:12:31]> 0x7f0b100568f0 -- Probation passed - setting RTP 
source address to XXX.XXX.XXX.XXX:14052
[Apr  1 11:12:31] -- Executing [s@macro-faxproceed:12] 
Wait("Local/0XX40@x-xxx-companyX-0001;2", "6") in new stack
[Apr  1 11:12:37] -- Executing [s@macro-faxproceed:13] 
ReceiveFAX("Local/0XX40@x-xxx-companyX-0001;2", 
"/var/spool/asterisk/fax/0XX40/1427879551.5.tif,Fd") in new stack
[Apr  1 11:12:37] -- Channel 
'Local/0XX40@x-xxx-companyX-0001;2' receiving FAX 
'/var/spool/asterisk/fax/0XX40/1427879551.5.tif'
[Apr  1 11:12:37]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.225458 
], stack sent 10 frames (200 ms) of silence.
[Apr  1 11:12:37]   == Using UDPTL TOS bits 184
[Apr  1 11:12:37]   == Using UDPTL CoS mark 5
[Apr  1 11:12:37]> 0x7f0b100568f0 -- Probation passed - setting RTP 
source address to XXX.XXX.XXX.XXX:14052
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.448203 
], channel sent 23 frames (460 ms) of silence.
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.508594 
], channel sent 3 frames (60 ms) of energy.
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.528360 
], channel sent 1 frames (20 ms) of silence.
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.548998 
], channel sent 1 frames (20 ms) of energy.
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.588275 
], channel sent 2 frames (40 ms) of silence.
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.748959 
], channel sent 8 frames (160 ms) of energy.
[Apr  1 11:12:38]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 000.749224 
], channel sent 1 frames (20 ms) of silence.
[Apr  1 11:12:40]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 002.820373 
], stack sent 130 frames (2600 ms) of energy.
[Apr  1 11:12:40]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 002.880379 
], stack sent 3 frames (60 ms) of silence.
[Apr  1 11:12:40]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 003.068482 
], channel sent 116 frames (2320 ms) of energy.
[Apr  1 11:12:40]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 003.148325 
], channel sent 4 frames (80 ms) of silence.
[Apr  1 11:12:42]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 005.227866 
], channel sent 104 frames (2080 ms) of energy.
[Apr  1 11:12:42]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 005.248242 
], channel sent 1 frames (20 ms) of silence.
[Apr  1 11:13:16]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 038.908270 
], channel sent 1683 frames (33660 ms) of energy.
[Apr  1 11:13:16]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 038.928126 
], channel sent 1 frames (20 ms) of silence.
[Apr  1 11:13:16]> Channel 
'Local/0XX40@x-xxx-companyX-0001;2' fax session '1', [ 038.968168 
], channel sent 2 frames (40 ms) of energy.
[Apr  1 11:13:16] -- Executing [h@macro-redirection:1] 
Set("SIP/access-trunk-0001", "CDR(userfield)= Hangup=16 
RTCP=ssrc=1981386606;themssrc=2116388711;lp=0;rxjitter=0.00;rxcount=2254;txjitter=0.000314;txcount=1949;rlp=0;rtt=0.00")
 in new stack
[Apr  1 11:13:16] -- Executing [h@macro-redirection:2] 
Hangup("SIP/access-trunk-0001", "") in new stack
[Apr  1 11:13:16]   == Spawn extension (macro-redirection, h, 2) exited 
non-zero on 'SIP/access-trunk-0001'
[Apr  1 11:13:16]   == Spawn extension (macro-redirection, s, 85) exited 
non-zero on 'SIP/access-trunk-0001' in macro 'redirection'
[Apr  1 11:13:16]   == Spawn extension (x-xxx-subscribers, 0XX44, 1) exited 
non-zero on 'SIP

[asterisk-users] PJSIP Sends BYE with Wrong IP

[Trey Hilyard]

Hello -

I am trying to decide if I have stumbled across a bug in PJSIP or I am just
missing something. My Asterisk has two interfaces, an "internal" eth0 and
an "external" eth1. In pjsip.conf, I define the following transports:

[trusted]
type=transport
protocol=udp
bind=10.xx.yy.zz:5060

[untrusted]
type=transport
protocol=udp
bind=12.4.aa.bb:5060

My internal endpoints use transport=internal and external endpoints use
transport=external. I guess that's obvious.

My netstat shows both transports listening:
Proto Recv-Q Send-Q Local Address   Foreign Address State
PID/Program name
udp0  0 12.4.aa.bb:5060   0.0.0.0:*
  25494/asterisk
udp0  0 10.xx.yy.xx:5060  0.0.0.0:*
  25494/asterisk

Everything works fine, most of the time. INVITEs, 1XX, 2XX are sent to the
right interface using the right source IP. But, when Asterisk tries to send
a BYE to any internal endpoint, it sends using the external IP, but it is
sent of the correct internal interface eth0. Only the IP layer is
incorrect. The SIP layer has the correct IP in the Via header. From what I
can tell, only BYE is affected.

I didn't have this problem with chan_sip. Am I just missing some
configuration?

To test, I have set up the most simple extension I can think of, and it
duplicates the condition:
exten => _9090,1,Answer
 same => n,Wait(2)
 same => n,Hangup
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Re: [asterisk-users] Update peer IP address

[Tech Support]

If I correctly understand what the problem is, what I did was write a 
script that runs out of CRON every 15 minutes. It checks the outside IP address 
by querying http://checkip.dyndns.org and compares it to the IP address stored 
in the parameter “externip” in the [general] section of sip.conf. If the two 
values are the same, the script exits quietly. If they are different, the 
script updates “externip” with the new address, does a sip reload, and shoots 
me an email saying there was an update. It's a fairly simple and 
straightforward process and does the job. I use this script for all PBX’s that 
are behind a NAT. I hope this helps.

Regards;

John

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Heckl
Sent: Wednesday, April 01, 2015 2:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Update peer IP address

 

 

Scott, thank you four your reply.

 

I had already though about both options, but the problem is, that after an ip 
change AND a new registration the ip address of the peer is not updated 
automatically. INVITES are answered with 401.

 

Only after a sip reload the peer works again.

 

That can't be normal...

 

Daniel


Am 31.03.2015 um 22:45 schrieb Scott Griepentrog :

You have two options for dealing with an IP change during the registration 
period:

 

1) set the registration time to shorter period of time to minimize the downtime

 

2) detect that the IP address has changed via whatever method available, and 
then issue a "sip reload" CLI command to asterisk, which will cause it to 
resend registrations immediately.

 

On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl  wrote:

Maybe someone could elaborate on my first question again.





If the ip address changes while a REGISTER period, the ip address of the peer 
isn't been updated. How can asterisk update the ip address of the peer?


Am 31.03.2015 um 12:36 schrieb Daniel Heckl :

Hello Sebastian,

 

I had already seen this list of the hosts, but it is not active. All servers 
with which my Asterisk has been communicated are not listed.

 

A port scan, to eventually update the list, found hundreds of servers provided 
in the address range 217.0.0.0/13 with open port 5060, some were even not 
found. I think there must be another solution.

 

If I change insecure to insecure=port,invite - could that be a solution?

 

Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? 
Has there anyone experience with dynamic ip addresses of Asterisk?

 

Daniel

 

Am 30.03.2015 um 20:09 schrieb Sebastian Kemper :

 

On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:



Hello

I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom
Germany. We have sometimes problems with incoming and outgoing calls.
I hope I can explain it understandable.


Hello Daniel,

I'll find myself in the same situation a few weeks from now :-)





For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de
), the message is answered with OK and the
peer is registered.

Usually INVITES comes now from this ip address. All works fine. But
sometimes INVITES comes from an other IP address, for example
217.0.23.100. This request Asterisk responds with 401 Unauthorized.

In the next register procedure REGISTER are sent to the new ip address
and answered also with OK. But qualify OPTIONS are continue be sent to
the old ip address. Incoming and outgoing calls are canceled. Outgoing
calls are answered with Forbidden.

Even if the REGISTER procedure works with the new ip address, the
peers are connected with the old address.

Waiting doesn’t help, only a „sip reload“ update the ip address of the
peer. 

What is the solution for this problem? How can asterisk update the
peer?


I think the solution - for the inbound issue at least - could be to add
more hosts as a peer. Have a looks at this forum post:

http://www.ip-phone-forum.de/showthread.php?t=268787 

 &p=1999371&viewfull=1#post1999371

The user used a template and than he added peers, each with its own IP
address. The provided list was last updated in 2014, though, so I assume
the provider in the meantime has added to that list.

It looks pretty tedious, though, I mean there could be dozens of IPs
you'd have to add. But I guess this is the way to go with Asterisk 11
and chan_sip.

The future looks brighter :-) I read that with pjsip, which I understand
is the replacement for chan_sip, you can have one peer entry and match
an IP range instead of a single host. That should tidy up the dialplan.

What I'm a little afraid of is the SIP provider using IPs out of a range
that they also use for other services. Maybe out of the same range they
hand out IPs to their customers. I guess we got to be careful :-)

Kind regards,
Sebastian




The As

Re: [asterisk-users] Update peer IP address

[Daniel Heckl]

John,

thank you four your answer. I think you have misunderstood the problem. It’s 
about a ip address change of the sip trunk, not of my asterisk server.

Kind regards,
Daniel

> Am 01.04.2015 um 16:40 schrieb Tech Support  >:
> 
> If I correctly understand what the problem is, what I did was write a 
> script that runs out of CRON every 15 minutes. It checks the outside IP 
> address by querying http://checkip.dyndns.org  
> and compares it to the IP address stored in the parameter “externip” in the 
> [general] section of sip.conf. If the two values are the same, the script 
> exits quietly. If they are different, the script updates “externip” with the 
> new address, does a sip reload, and shoots me an email saying there was an 
> update. It's a fairly simple and straightforward process and does the job. I 
> use this script for all PBX’s that are behind a NAT. I hope this helps.
> Regards;
> John

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Re: [asterisk-users] Update peer IP address

[Andres]

On 4/1/15 10:48 AM, Daniel Heckl wrote:

John,

thank you four your answer. I think you have misunderstood the 
problem. It’s about a ip address change of the sip trunk, not of my 
asterisk server.
You would probably benefit by enabling the DNS Manager to allow for 
dynamic IP changes:


# cat dnsmgr.conf
[general]
enable=yes ; enable creation of managed DNS lookups
;   default is 'no'
refreshinterval=180   ; refresh managed DNS lookups every  seconds
;   default is 300 (5 minutes)




Kind regards,
Daniel

Am 01.04.2015 um 16:40 schrieb Tech Support >:


If I correctly understand what the problem is, what I did was 
write a script that runs out of CRON every 15 minutes. It checks the 
outside IP address by queryinghttp://checkip.dyndns.organd compares 
it to the IP address stored in the parameter “externip” in the 
[general] section of sip.conf. If the two values are the same, the 
script exits quietly. If they are different, the script updates 
“externip” with the new address, does a sip reload, and shoots me an 
email saying there was an update. It's a fairly simple and 
straightforward process and does the job. I use this script for all 
PBX’s that are behind a NAT. I hope this helps.

Regards;
John







--
Technical Support
http://www.cellroute.net

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Re: [asterisk-users] Update peer IP address

[Sebastian Kemper]

On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote:
> Hello Sebastian,
> 
> I had already seen this list of the hosts, but it is not active. All
> servers with which my Asterisk has been communicated are not listed.
> 
> A port scan, to eventually update the list, found hundreds of servers
> provided in the address range 217.0.0.0/13 with open port 5060, some
> were even not found. I think there must be another solution.
> 
> If I change insecure to insecure=port,invite - could that be a
> solution?

Hello Daniel,

I've asked myself that, too. But I don't have access to the connection,
yet, so I can't test it right away.

Kind regards,
Sebastian

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Re: [asterisk-users] Update peer IP address

[Sebastian Kemper]

On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
> On 4/1/15 10:48 AM, Daniel Heckl wrote:
> > John,
> >
> > thank you four your answer. I think you have misunderstood the
> > problem. It’s about a ip address change of the sip trunk, not of my
> > asterisk server.
> You would probably benefit by enabling the DNS Manager to allow for
> dynamic IP changes:
> 
> # cat dnsmgr.conf [general] enable=yes ; enable creation
> of managed DNS lookups ;   default is 'no' refreshinterval=180   ;
> refresh managed DNS lookups every  seconds ;   default is 300 (5
> minutes)

Hello Andres,

I read that same suggestion elsewhere in connection with Deutsche
Telekom, so it seems there's some benefit in it.

Daniel, did you try it out already?

Kind regards,
Sebastian

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[asterisk-users] Asterisk 11.17.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 11.17.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.17.0 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-17899 - Handle crypto lifetime in SDES-SRTP negotiation
  (Reported by Dwayne Hubbard)

Bugs fixed in this release:
---
 * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
  res_odbc (Reported by ibercom)
 * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
  with replaces (Reported by Eelco Brolman)
 * ASTERISK-24479 - Enable REF_DEBUG for module references
  (Reported by Corey Farrell)
 * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
  fully disconnect underlying socket, leading to events being
  dropped with no additional information (Reported by Matt Jordan)
 * ASTERISK-24772 - ODBC error in realtime sippeers when device
  unregisters under MariaDB (Reported by Richard Miller)
 * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
  (Reported by Corey Farrell)
 * ASTERISK-24799 - [patch] make fails with undefined reference to
  SSLv3_client_method (Reported by Alexander Traud)
 * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
  for playing back messages stored in IMAP - play_message: No
  origtime (Reported by Graham Barnett)
 * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
  OSX with 64 bit integers (Reported by Corey Farrell)
 * ASTERISK-24796 - Codecs and bucket schema's prevent module
  unload (Reported by Corey Farrell)
 * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
  (Reported by Ashley Sanders)
 * ASTERISK-24797 - bridge_softmix: G.729 codec license held
  (Reported by Kevin Harwell)
 * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
  thread ID being passed to pthread_kill (Reported by JoshE)
 * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
  fail (Reported by Terry Wilson)
 * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
  SRTP for audio, but they responded without it' is ambiguous and
  wrong in some cases (Reported by Rusty Newton)
 * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
  error response and BYE are sent to the caller (Reported by
  Makoto Dei)
 * ASTERISK-18105 - most of asterisk modules are unbuildable in
  cygwin environment (Reported by feyfre)
 * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
 * ASTERISK-24838 - chan_sip: Locking inversion occurs when
  building a peer causes a peer poke during request handling
  (Reported by Richard Mudgett)
 * ASTERISK-24825 - Caller ID not recognized using
  Centrex/Distinctive dialing (Reported by Richard Mudgett)
 * ASTERISK-24739 - [patch] - Out of files -- call fails --
  numerous files with inodes from under /usr/share/zoneinfo,
  mostly posixrules (Reported by Ed Hynan)
 * ASTERISK-23390 - NewExten Event with application AGI shows up
  before and after AGI runs (Reported by Benjamin Keith Ford)
 * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
  voicemail stored in LDAP (Reported by Graham Barnett)
 * ASTERISK-24808 - res_config_odbc: Improper escaping of
  backslashes occurs with MySQL (Reported by Javier Acosta)
 * ASTERISK-20850 - [patch]Nested functions aren't portable.
  Adapting RAII_VAR to use clang/llvm blocks to get the
  same/similar functionality. (Reported by Diederik de Groot)
 * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
  by Frank DiGennaro)
 * ASTERISK-21038 - Bad command completion of "core set debug
  channel" (Reported by Richard Kenner)
 * ASTERISK-18708 - func_curl hangs channel under load (Reported by
  Dave Cabot)
 * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
  Atis Lezdins)
 * ASTERISK-24876 - Investigate reference leaks from
  tests/channels/local/local_optimize_away (Reported by Corey
  Farrell)
 * ASTERISK-24817 - init_logger_chain: unreachable code block
  (Reported by Corey Farrell)
 * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
  snuffy)
 * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
  under OpenBSD (Reported by snuffy)

Improvements made in this release:
---
 * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
  Couldn't find mailbox %s in context (Reported by Graham Barnett)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0

Thank y

[asterisk-users] Asterisk 13.3.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.0 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-24703 - ARI: Add the ability to "transfer" (redirect) a
  channel (Reported by Matt Jordan)
 * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
  (Reported by Dwayne Hubbard)

Bugs fixed in this release:
---
 * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
  string copy (Reported by Yura Kocyuba)
 * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
  sorcery.conf false ERROR messages may occur (Reported by Joshua
  Colp)
 * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
  (Reported by Matt Jordan)
 * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
  res_odbc (Reported by ibercom)
 * ASTERISK-24479 - Enable REF_DEBUG for module references
  (Reported by Corey Farrell)
 * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
  fully disconnect underlying socket, leading to events being
  dropped with no additional information (Reported by Matt Jordan)
 * ASTERISK-24772 - ODBC error in realtime sippeers when device
  unregisters under MariaDB (Reported by Richard Miller)
 * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
  is destroyed by ARI during shutdown (Reported by Richard
  Mudgett)
 * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
  by Zane Conkle)
 * ASTERISK-24015 - app_transfer fails with PJSIP channels
  (Reported by Private Name)
 * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
  transfer scenario. (Reported by Mark Michelson)
 * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
  Niklas Larsson)
 * ASTERISK-24716 - Improve pjsip log messages for presence
  subscription failure (Reported by Rusty Newton)
 * ASTERISK-24612 - res_pjsip: No information if a required sorcery
  wizard is not loaded (Reported by Joshua Colp)
 * ASTERISK-24768 - res_timing_pthread: file descriptor leak
  (Reported by Matthias Urlichs)
 * ASTERISK-24685 - "pjsip show version" CLI command (Reported by
  Joshua Colp)
 * ASTERISK-24632 - install_prereq script installs pjproject
  without IPv6 support (Reported by Rusty Newton)
 * ASTERISK-24085 - Documentation - We should remove or further
  document the 'contact' section in pjsip.conf (Reported by Rusty
  Newton)
 * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
  JoshE)
 * ASTERISK-24700 - CRASH: NULL channel is being passed to
  ast_bridge_transfer_attended() (Reported by Zane Conkle)
 * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
  (Reported by Corey Farrell)
 * ASTERISK-24799 - [patch] make fails with undefined reference to
  SSLv3_client_method (Reported by Alexander Traud)
 * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
  Events (Reported by klaus3000)
 * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
  call (Reported by Marcel Manz)
 * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
  (Reported by Panos Gkikakis)
 * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
  for playing back messages stored in IMAP - play_message: No
  origtime (Reported by Graham Barnett)
 * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
  OSX with 64 bit integers (Reported by Corey Farrell)
 * ASTERISK-24796 - Codecs and bucket schema's prevent module
  unload (Reported by Corey Farrell)
 * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
  (Reported by Ashley Sanders)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  is invalid (Reported by Rusty Newton)
 * ASTERISK-24785 - 'Expires' header missing from 200 OK on
  REGISTER (Reported by Ross Beer)
 * ASTERISK-24677 - ARI GET variable on channel provides unhelpful
  response on non-existent variable (Reported by Joshua Colp)
 * ASTERISK-24797 - bridge_softmix: G.729 codec license held
  (Reported by Kevin Harwell)
 * ASTERISK-24812 - ARI: Creating channels through /channels
  resource always uses SLIN, which results in unneeded transcoding
  (Reported by Matt Jordan)
 * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
  thread ID being passed to pthread_kill (Reported by JoshE)
 * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
  fail (Reported by Terry Wilson)
 * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
  SRTP for audio, but they respond

[asterisk-users] PJSIP Endpoint AOR question

[Dan Cropp]

I am running asterisk 13.1.0

In pjsip.conf, the endpoint section has an aors and an auth field.

I can name the auth field anything I want.  The key is to set the auth=field 
accordingly.
However, when I try this with the aors field, it never works.  It seems I have 
to name the aors=field to match the name of the endpoint section.

Is this correct?

Would there ever be a need for multiple aors to a single endpoint?  Since the 
field is named aors, I thought this would be possible.  How would I do this if 
I have to name the aor the name of the endpoint?

This fails...

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[aor3]
type = aor
max_contacts = 1
remove_existing = yes

[auth3]
type = auth
username = 1003
password = Password

[1003]
type = endpoint
context = Test
transport = transport1
auth = auth3
aors = aor3
dtmf_mode = inband
device_state_busy_at = 1
disallow = all
allow = ulaw


This succeeds...

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[1003]
type = aor
max_contacts = 1
remove_existing = yes

[auth3]
type = auth
username = 1003
password = Password

[1003]
type = endpoint
context = Test
transport = transport1
auth = auth3
aors = 1003
dtmf_mode = inband
device_state_busy_at = 1
disallow = all
allow = ulaw

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Re: [asterisk-users] PJSIP Endpoint AOR question

[Trey Hilyard]

I don't know why you have issues using different names. I have multiple
AORs assigned to a single endpoint and it works fine. I have to admit that
my AORs do contain the endpoint name, though. For example, for endpoint
"myswitch" I have two AORs, "myswitch_1" and "myswitch_2", and I assign
them to the endpoint with aors=myswitch_1,myswitch_2.

When you say that the first example fails, what specifically fails? Do the
PJSIP modules load but then you get an error when trying to dial to an
endpoint, or maybe receive a call from one?

On Wed, Apr 1, 2015 at 2:53 PM Dan Cropp  wrote:

> I am running asterisk 13.1.0
>
>
>
> In pjsip.conf, the endpoint section has an aors and an auth field.
>
>
>
> I can name the auth field anything I want.  The key is to set the
> auth=field accordingly.
>
> However, when I try this with the aors field, it never works.  It seems I
> have to name the aors=field to match the name of the endpoint section.
>
>
>
> Is this correct?
>
>
>
> Would there ever be a need for multiple aors to a single endpoint?  Since
> the field is named aors, I thought this would be possible.  How would I do
> this if I have to name the aor the name of the endpoint?
>
>
>
> This fails...
>
>
>
> [transport1]
>
> type = transport
>
> bind = 0.0.0.0
>
> protocol = udp
>
>
>
> [aor3]
>
> type = aor
>
> max_contacts = 1
>
> remove_existing = yes
>
>
>
> [auth3]
>
> type = auth
>
> username = 1003
>
> password = Password
>
>
>
> [1003]
>
> type = endpoint
>
> context = Test
>
> transport = transport1
>
> auth = auth3
>
> aors = aor3
>
> dtmf_mode = inband
>
> device_state_busy_at = 1
>
> disallow = all
>
> allow = ulaw
>
>
>
>
>
> This succeeds...
>
>
>
> [transport1]
>
> type = transport
>
> bind = 0.0.0.0
>
> protocol = udp
>
>
>
> [1003]
>
> type = aor
>
> max_contacts = 1
>
> remove_existing = yes
>
>
>
> [auth3]
>
> type = auth
>
> username = 1003
>
> password = Password
>
>
>
> [1003]
>
> type = endpoint
>
> context = Test
>
> transport = transport1
>
> auth = auth3
>
> aors = 1003
>
> dtmf_mode = inband
>
> device_state_busy_at = 1
>
> disallow = all
>
> allow = ulaw
>
>
> --
> _
> -- 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
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Re: [asterisk-users] PJSIP Endpoint AOR question

[Trey Hilyard]

I just realized that you are asking about dynamic AORs, not static Contacts
in an AOR. That may be the difference. I have never actually tried giving a
dynamic AOR a different name. And you wouldn't want more than one dynamic
AOR, you'd just use an AOR that allowed more than 1 contact.

On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard  wrote:

> I don't know why you have issues using different names. I have multiple
> AORs assigned to a single endpoint and it works fine. I have to admit that
> my AORs do contain the endpoint name, though. For example, for endpoint
> "myswitch" I have two AORs, "myswitch_1" and "myswitch_2", and I assign
> them to the endpoint with aors=myswitch_1,myswitch_2.
>
> When you say that the first example fails, what specifically fails? Do the
> PJSIP modules load but then you get an error when trying to dial to an
> endpoint, or maybe receive a call from one?
>
> On Wed, Apr 1, 2015 at 2:53 PM Dan Cropp  wrote:
>
>> I am running asterisk 13.1.0
>>
>>
>>
>> In pjsip.conf, the endpoint section has an aors and an auth field.
>>
>>
>>
>> I can name the auth field anything I want.  The key is to set the
>> auth=field accordingly.
>>
>> However, when I try this with the aors field, it never works.  It seems I
>> have to name the aors=field to match the name of the endpoint section.
>>
>>
>>
>> Is this correct?
>>
>>
>>
>> Would there ever be a need for multiple aors to a single endpoint?  Since
>> the field is named aors, I thought this would be possible.  How would I do
>> this if I have to name the aor the name of the endpoint?
>>
>>
>>
>> This fails...
>>
>>
>>
>> [transport1]
>>
>> type = transport
>>
>> bind = 0.0.0.0
>>
>> protocol = udp
>>
>>
>>
>> [aor3]
>>
>> type = aor
>>
>> max_contacts = 1
>>
>> remove_existing = yes
>>
>>
>>
>> [auth3]
>>
>> type = auth
>>
>> username = 1003
>>
>> password = Password
>>
>>
>>
>> [1003]
>>
>> type = endpoint
>>
>> context = Test
>>
>> transport = transport1
>>
>> auth = auth3
>>
>> aors = aor3
>>
>> dtmf_mode = inband
>>
>> device_state_busy_at = 1
>>
>> disallow = all
>>
>> allow = ulaw
>>
>>
>>
>>
>>
>> This succeeds...
>>
>>
>>
>> [transport1]
>>
>> type = transport
>>
>> bind = 0.0.0.0
>>
>> protocol = udp
>>
>>
>>
>> [1003]
>>
>> type = aor
>>
>> max_contacts = 1
>>
>> remove_existing = yes
>>
>>
>>
>> [auth3]
>>
>> type = auth
>>
>> username = 1003
>>
>> password = Password
>>
>>
>>
>> [1003]
>>
>> type = endpoint
>>
>> context = Test
>>
>> transport = transport1
>>
>> auth = auth3
>>
>> aors = 1003
>>
>> dtmf_mode = inband
>>
>> device_state_busy_at = 1
>>
>> disallow = all
>>
>> allow = ulaw
>>
>>
>> --
>> _
>> -- 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
>
>
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Re: [asterisk-users] PJSIP Endpoint AOR question

[Joshua Colp]

Dan Cropp wrote:

I am running asterisk 13.1.0

In pjsip.conf, the endpoint section has an aors and an auth field.

I can name the auth field anything I want. The key is to set the
auth=field accordingly.

However, when I try this with the aors field, it never works. It seems I
have to name the aors=field to match the name of the endpoint section.

Is this correct?

Would there ever be a need for multiple aors to a single endpoint? Since
the field is named aors, I thought this would be possible. How would I
do this if I have to name the aor the name of the endpoint?


Whether you have to name it the same or not is up to the SIP client 
registering. Some allow you to specify the AOR you are registering 
against. Some assume that the username you are authenticating as is the 
same as your AOR.


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] PJSIP Endpoint AOR question

[Dan Cropp]

Thanks Joshua.

That must be it.  I'm using PhonerLite and a Cisco SPA504G phone.

Have a great day!


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, April 01, 2015 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP Endpoint AOR question

Dan Cropp wrote:
> I am running asterisk 13.1.0
>
> In pjsip.conf, the endpoint section has an aors and an auth field.
>
> I can name the auth field anything I want. The key is to set the 
> auth=field accordingly.
>
> However, when I try this with the aors field, it never works. It seems 
> I have to name the aors=field to match the name of the endpoint section.
>
> Is this correct?
>
> Would there ever be a need for multiple aors to a single endpoint? 
> Since the field is named aors, I thought this would be possible. How 
> would I do this if I have to name the aor the name of the endpoint?

Whether you have to name it the same or not is up to the SIP client 
registering. Some allow you to specify the AOR you are registering against. 
Some assume that the username you are authenticating as is the same as your AOR.

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] PJSIP Endpoint AOR question

[Dan Cropp]

Thanks Trey.

Have a great day!

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Trey Hilyard
Sent: Wednesday, April 01, 2015 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP Endpoint AOR question

I just realized that you are asking about dynamic AORs, not static Contacts in 
an AOR. That may be the difference. I have never actually tried giving a 
dynamic AOR a different name. And you wouldn't want more than one dynamic AOR, 
you'd just use an AOR that allowed more than 1 contact.

On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard 
mailto:kct...@gmail.com>> wrote:
I don't know why you have issues using different names. I have multiple AORs 
assigned to a single endpoint and it works fine. I have to admit that my AORs 
do contain the endpoint name, though. For example, for endpoint "myswitch" I 
have two AORs, "myswitch_1" and "myswitch_2", and I assign them to the endpoint 
with aors=myswitch_1,myswitch_2.

When you say that the first example fails, what specifically fails? Do the 
PJSIP modules load but then you get an error when trying to dial to an 
endpoint, or maybe receive a call from one?

On Wed, Apr 1, 2015 at 2:53 PM Dan Cropp 
mailto:d...@amtelco.com>> wrote:
I am running asterisk 13.1.0

In pjsip.conf, the endpoint section has an aors and an auth field.

I can name the auth field anything I want.  The key is to set the auth=field 
accordingly.
However, when I try this with the aors field, it never works.  It seems I have 
to name the aors=field to match the name of the endpoint section.

Is this correct?

Would there ever be a need for multiple aors to a single endpoint?  Since the 
field is named aors, I thought this would be possible.  How would I do this if 
I have to name the aor the name of the endpoint?

This fails...

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[aor3]
type = aor
max_contacts = 1
remove_existing = yes

[auth3]
type = auth
username = 1003
password = Password

[1003]
type = endpoint
context = Test
transport = transport1
auth = auth3
aors = aor3
dtmf_mode = inband
device_state_busy_at = 1
disallow = all
allow = ulaw


This succeeds...

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[1003]
type = aor
max_contacts = 1
remove_existing = yes

[auth3]
type = auth
username = 1003
password = Password

[1003]
type = endpoint
context = Test
transport = transport1
auth = auth3
aors = 1003
dtmf_mode = inband
device_state_busy_at = 1
disallow = all
allow = ulaw

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[asterisk-users] Asterisk 13.3.0 compiled with clang on FreeBSD crashes

[Guido Falsi]

Hi,

I'm maintaining the FreeBSD ports for asterisk(With madpi...@freebsd.org
as identity). Here's a link to the
asterisk13 port for your reference:

http://www.freshports.org/net/asterisk13/

I performed some tests with RC1 and am doing some final tests with the
final release before committing the update.

Up to now the ports forced using gcc, version 4.8 lately, to compile it.
And for this update I'll keep things unmodified.

I tested compiling it with clang, on FreeBSD releases using it by
default (10.0 onward).

I discovered that on FreeBSD 10.1 amd64, asterisk 13.3.0-RC1 compiled
with clang 3.4.1 crashes at startup, while it seems to work quite fine
when compiled with gcc 4.8.

Some data follows, trimmed down a little, but if any more in depth
analysis is required I'll try to provide it.

Here's a verbose log of the crash (only the final part):

  == Registered custom function 'AMI_CLIENT'
  == Parsing '/usr/local/etc/asterisk/manager.conf': Found
  == Parsing '/usr/local/etc/asterisk/users.conf': Found
  == Parsing '/usr/local/etc/asterisk/enum.conf': Found
  == Registered application 'CallCompletionRequest'
  == Registered application 'CallCompletionCancel'
  == Parsing '/usr/local/etc/asterisk/ccss.conf': Found
  == Parsing '/usr/local/etc/asterisk/ccss.conf': Found
Segmentation fault (core dumped)
root@asterisk:~ #

I used gdb, to get some backtrace data:

(gdb) bt
#0  0x000803261970 in strcasecmp_l () from /lib/libc.so.7
#1  0x00532926 in media_info_cmp ()
#2  0x00459bdc in internal_ao2_traverse ()
#3  0x00459f49 in __ao2_find ()
#4  0x00533bdf in process_file ()
#5  0x005330cc in media_index_update ()
#6  0x005890a2 in update_index_cb ()
#7  0x00459bdc in internal_ao2_traverse ()
#8  0x00459925 in __ao2_callback ()
#9  0x00588ea9 in ast_sounds_reindex ()
#10 0x005890ca in ast_sounds_index_init ()
#11 0x00452836 in main ()

So it looks like it crashes in main/media_index.c line 140.

Anyone has some insight, suggestions, or ways to better diagnose this?

Thanks in advance.

-- 
Guido Falsi 

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[asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
with the same service provides. We have 8 phone numbers in total.

Incoming calls from the public are all correctly directed to appropriate
office handsets. However, the display on the reception phone (the only one
i care about) is always showing the same "SIP/Account1_0843214321" rather
than the account representing the number dialed.

For-instance, if Sam on her mobile calls "*08*", Asterisk will show
a log entry like the following:

-- Executing [s@incoming:1] Set("SIP/*Account1_08*", "
thedid=""NodePhone""") in new stack
But "Account1_*08*" (as the name suggests) has a phone number of "
*08*" and not "*08*".

So Sam's call will come through and be routed to the correct handset as the
business needs, but it seems that all incoming calls are being labeled as
though coming in on a different account. The effective problem is that the
calledID is now wrong.

I'm after some general advice on how to handle the problem.

Ta,


-Andrew
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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[John Kiniston]

Can you show us the CDR record for that call?

And maybe what your s priority of your incoming context is?

It should be easy to get what number was dialed, Try:

${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)}

Normally I display the callers number on my phones, Not the number they
dialed?

On Wed, Apr 1, 2015 at 4:50 PM, Andrew Galdes 
wrote:

> Hello all,
>
> I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
> with the same service provides. We have 8 phone numbers in total.
>
> Incoming calls from the public are all correctly directed to appropriate
> office handsets. However, the display on the reception phone (the only one
> i care about) is always showing the same "SIP/Account1_0843214321" rather
> than the account representing the number dialed.
>
> For-instance, if Sam on her mobile calls "*08*", Asterisk will
> show a log entry like the following:
>
> -- Executing [s@incoming:1] Set("SIP/*Account1_08*", "
> thedid=""NodePhone""") in new stack
> But "Account1_*08*" (as the name suggests) has a phone number of "
> *08*" and not "*08*".
>
> So Sam's call will come through and be routed to the correct handset as
> the business needs, but it seems that all incoming calls are being labeled
> as though coming in on a different account. The effective problem is that
> the calledID is now wrong.
>
> I'm after some general advice on how to handle the problem.
>
> Ta,
>
>
> -Andrew
>
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> _
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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andres]

On 4/1/15 7:50 PM, Andrew Galdes wrote:

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip 
accounts with the same service provides. We have 8 phone numbers in 
total.


Incoming calls from the public are all correctly directed to 
appropriate office handsets. However, the display on the reception 
phone (the only one i care about) is always showing the same 
"SIP/Account1_0843214321" rather than the account representing the 
number dialed.


For-instance, if Sam on her mobile calls "*08*", Asterisk will 
show a log entry like the following:


-- Executing [s@incoming:1] Set("SIP/*Account1_08*", 
"thedid=""NodePhone">"") in new stack


But "Account1_*08*" (as the name suggests) has a phone number 
of "*08*" and not "*08*".


It looks like all incoming calls are all being matched against the same 
entry in sip.conf.   A 'set set debug on' should clearly indicate this.  
Look for the line that says :  Found peer ''08'
So Sam's call will come through and be routed to the correct handset 
as the business needs, but it seems that all incoming calls are being 
labeled as though coming in on a different account. The effective 
problem is that the calledID is now wrong.


I'm after some general advice on how to handle the problem.

Ta,


-Andrew





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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Dmitriy Serov]

This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes

Carefully read the description, it is better to test in "after hours".

02.04.2015 2:50, Andrew Galdes пишет:

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip 
accounts with the same service provides. We have 8 phone numbers in 
total.


Incoming calls from the public are all correctly directed to 
appropriate office handsets. However, the display on the reception 
phone (the only one i care about) is always showing the same 
"SIP/Account1_0843214321" rather than the account representing the 
number dialed.


For-instance, if Sam on her mobile calls "*08*", Asterisk will 
show a log entry like the following:


-- Executing [s@incoming:1] Set("SIP/*Account1_08*", 
"thedid=""NodePhone">"") in new stack


But "Account1_*08*" (as the name suggests) has a phone number 
of "*08*" and not "*08*".


So Sam's call will come through and be routed to the correct handset 
as the business needs, but it seems that all incoming calls are being 
labeled as though coming in on a different account. The effective 
problem is that the calledID is now wrong.


I'm after some general advice on how to handle the problem.

Ta,


-Andrew




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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