Re: [asterisk-users] problems with natted phones

[Michael L. Young]

El jue, 8 de jul. de 2021 a la(s) 14:58, Marek Greško (mgres...@gmail.com)
escribió:


> The asterisk is connected to the internet with public static IP address.
>
> The pjsip config contains:
>
>
What does your transport config look like?

Take a look at this wiki page:
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT

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Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

[Michael L. Young]

- On Feb 5, 2021, at 11:18 AM, Michael L. Young  wrote: 

> - On Feb 4, 2021, at 4:26 PM, Social Boh  wrote:

>> The problem is with this CentOS 7 glibc version:

>> 2.17-317.el7

>> After the library update and system reboog,
>> gotoif Asterisk application, stop to working

>> Any hint to solve?

> Until it is resolved, you can do a 'yum history' and note the transaction ID 
> of
> the update. Then try running 'yum history undo [transaction id]'. That should
> roll you back to the previous glibc.

> Looks like Red Hat is already working on it:
> https://access.redhat.com/solutions/5778071

Here is the Bugzilla report for anyone on RHEL / CentOS 7: 
https://bugzilla.redhat.com/show_bug.cgi?id=1925204 

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Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

[Michael L. Young]

- On Feb 4, 2021, at 4:26 PM, Social Boh  wrote: 

> The problem is with this CentOS 7 glibc version:

> 2.17-317.el7

> After the library update and system reboog,
> gotoif Asterisk application, stop to working

> Any hint to solve?

Until it is resolved, you can do a 'yum history' and note the transaction ID of 
the update. Then try running 'yum history undo [transaction id]'. That should 
roll you back to the previous glibc. 

Looks like Red Hat is already working on it: 
https://access.redhat.com/solutions/5778071 

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Michael Young 
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Re: [asterisk-users] call an IP camera?

[Ralph L. Miller]

The Grandstream camera product line has SIP output so you can "call" the camera

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Antony Stone
Sent: Thursday, September 24, 2020 10:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call an IP camera?

On Thursday 24 September 2020 at 16:31:33, hw wrote:

> Hi,
> 
> is it possible to "call" an IP camera?

Only if it talks SIP (which some do, generally door entry cameras with a push 
button input and often a lock release output).

> I'm thinking about something like bridging with a music stream, but instead
> of streaming audio, bridge with the video stream from the camera.

So, maybe you should treat it like a music stream such as music on hold?

> It would be very cool if I could just call the camera and see what's going
> on.  Ffmpeg shows the following streams available from the camera:
> 
> Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 12
> fps, 12 tbr, 90k tbn, 24 tbc
> 
> Stream #0:0: Video: h264 (Main), yuv420p(progressive), 640x352, 12 fps,
> 12 tbr, 90k tbn, 24 tbc
> 
> Perhaps it's not even necessary to recode the stream?

Very likely, but what you're looking at there is the media format; you also 
need some sort of signalling protocol if you're going to call it from 
Asterisk.

I would start with something like 
https://www.voip-info.org/asterisk-config-musiconholdconf/
https://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
(or any more up to date documentation if you can find it).

I've never tried that with video, but given how the media negotiation between 
Asterisk and SIP devices is handled, I would expect it to work given 
compatible codecs.


Antony.

-- 
I bought a book on memory techniques, but I've forgotten where I put it.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

[Michael L. Young]

> From: "John Hughes" 
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
> 
> Sent: Thursday, May 14, 2020 2:10:45 AM
> Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and
> alaw; asterisk wants to translate g729 -> alaw. WHY?

> I am having a problem with one of my callers who is using either g729 or 
> alaw. I
> can do alaw but not g729 so asterisk should negotiate alaw right? In fact from
> the sip debug it looks like it does, but then I get the dreaded 
> "channel.c:5630
> set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the
> call hangs up. Why?

> Last minute thought: Is it possible that the caller is sending g729 in RTP 
> even
> though the SIP negotiation clearly chooses alaw? Maybe I need some RTP
> debugging.

> Asterisk 13.14.1 on Debian, using chan_sip.
Hi John, 

Maybe a newer version of Asterisk would help? The latest release for 13 is 
version 13.33. The version you are on was released 3 years ago. 

Here is an issue which looks like what you describe and was fixed in 13.16 
[ https://issues.asterisk.org/jira/browse/ASTERISK-26143 | 
https://issues.asterisk.org/jira/browse/ASTERISK-26143 ] 

Not sure if this is the answer to your problem but thought that I would throw 
that out there. 

Michael L. Young 

(elguero) 
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Re: [asterisk-users] permission woes on systemd

[Michael L. Young]

- Original Message -
> From: "sean darcy" 
> To: "Asterisk Users Mailing List, Non-Commercial Discussion" 
> 
> Sent: Tuesday, January 21, 2020 9:22:28 PM
> Subject: [asterisk-users] permission woes on systemd

[..]

> So why would starting asterisk as user asterisk work, but fail using
> systemd ?
> 

Have you checked SELinux?  After creating the configuration files, did you run 
'restorecon' on the appropriate asterisk directories?  If not, the files are 
not labeled correctly and SELinux might be denying access.

Just a thought.

Michael

(elguero)

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[asterisk-users] ODBC connection pooling

[L S]

We are still on Asterisk 1.8.18.

Updated to UnixODBC 2.3.7 and MySQL Connector/ODBC 8.0 to use the pooling
in UnixODBC. Asterisk runs smoothly with those, but it seems there is no
pooling; even though 'odbc show' displays 'Pooled: Yes' - Connections in
Use is always 1.

I just would like to know if odbc pooling is/was doable in Asterisk 1.8.18.
Res_odbc has settings like pooling, limit, share-connections, but not sure
if these apply to mysql odbc connections. (I know res_odbc went thru a lot
of changes since version 13 and pooling is pretty straightforward now, but
it will take us a while to upgrade.)

Anyway here is my configuration:

odbcinst.ini
[ODBC]
Pooling=Yes

[MySQL]
Driver=/usr/lib64/libmyodbc8w.so
FileUsage=1
CPTimeout=120


res_odbc.conf
[asterisk]
enabled => yes
dsn => asterisk-connector
username => user
password => passwd
pre-connect => yes
share-connections => no
limit => 5
pooling => yes

Thanks,
Matt
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Re: [asterisk-users] 100% CPU after upgrade.

[Michael L. Young]

- Original Message -
> From: "Mike Diehl" <mdiehlena...@gmail.com>
> Sent: Monday, April 3, 2017 5:45:58 PM
> Subject: Re: [asterisk-users] 100% CPU after upgrade.

> Those are all rational questions, so here we go:
> 
> We upgraded from 11.x, though the system was a backup server, so it was never
> actually used.
> 
> The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty
> of power for what I'm asking it to do.  The system is configured via RT using
> a local Mysql database.
> 

Which distro are you running?  How are you starting Asterisk (init script / 
systemd)?

--
Michael L. Young

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Re: [asterisk-users] asterisk-users Digest, Vol 150, Issue 17

[Henrique L.]

hi,

Do you edit your

voicemail.conf?
[default]
1091=(number to access your voicemail in your phone ex: 1234)





2017-01-25 16:00 GMT-02:00 :

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> Today's Topics:
>
>1.  Asterisk 13.13.1 (Motty Cruz)
>2. Re: Asterisk 13.13.1 (Olivier)
>
>
> --
>
> Message: 1
> Date: Tue, 24 Jan 2017 12:03:05 -0800
> From: "Motty Cruz" 
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Subject: [asterisk-users]  Asterisk 13.13.1
> Message-ID: <5887b2fb.86c5620a.8e94a.d...@mx.google.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
> starting to complaint about packets loss, conversations are choppy!
>
>
>
>
> I don't even know where to start looking! Choppy conversations happened
> within users. I am using sip.conf
>
>
>
> [1091]
>
> type=friend
>
> context=sip-phone
>
> call-limit=2
>
> trustrpid=no
>
> callerid="dev1" <1091>
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> username=1091
>
> secret=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@default
>
> nat=force_rport,comedia
>
> canreinvite=no
>
>
>
> extensions.conf
>
> exten => 1091,hint,SIP/${EXTEN}
>
> exten => 1091,1,Dial(SIP/${EXTEN},15,t)
>
> exten => 1091,2,Voicemail(${EXTEN}@default,u)
>
> exten => 1091,102,Voicemail(${EXTEN}@default,b)
>
> exten => 1091,103,Hangup
>
>
>
> [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
>
> Retransmission timeout reached on transmission
> 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request)
> --
> See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
>
>
> any ideas?
>
>
>
> Thanks!
>
> Motty
>
> -- next part --
> An HTML attachment was scrubbed...
> URL:  attachments/20170124/e9731841/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 25 Jan 2017 13:30:00 +0100
> From: Olivier 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Asterisk 13.13.1
> Message-ID:
>  gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
> How did you installed Asterisk 1.8 and 13 ? From source or from package ?
>
> I would be curious to see what would happen after downgrading back to 1.8.
>
> 2017-01-24 21:03 GMT+01:00 Motty Cruz :
>
> > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users
> are
> > starting to complaint about packets loss, conversations are choppy!
> >
> >
> >
> >
> > I don?t even know where to start looking! Choppy conversations happened
> > within users. I am using sip.conf
> >
> >
> >
> > [1091]
> >
> > type=friend
> >
> > context=sip-phone
> >
> > call-limit=2
> >
> > trustrpid=no
> >
> > callerid="dev1" <1091>
> >
> > disallow=all
> >
> > allow=ulaw
> >
> > allow=alaw
> >
> > username=1091
> >
> > secret=X
> >
> > dtmfmode=rfc2833
> >
> > host=dynamic
> >
> > mailbox=10091@default
> >
> > nat=force_rport,comedia
> >
> > canreinvite=no
> >
> >
> >
> > extensions.conf
> >
> > exten => 1091,hint,SIP/${EXTEN}
> >
> > exten => 1091,1,Dial(SIP/${EXTEN},15,t)
> >
> > exten => 1091,2,Voicemail(${EXTEN}@default,u)
> >
> > exten => 1091,102,Voicemail(${EXTEN}@default,b)
> >
> > exten => 1091,103,Hangup
> >
> >
> >
> > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
> >
> > Retransmission timeout reached on transmission
> 7c803889-63e1b3fe-c2b5ef77@
> > 192.168.0.191 for seqno 156 (Critical Request) -- See
> > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> >
> > Packet timed out after 32000ms with no response
> >
> >
> >
> > any ideas?
> >
> >
> >
> > Thanks!
> >
> > Motty
> >
> > --
> > _
> > -- 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:
> > 

Re: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6

[Michael L. Young]

- On May 4, 2016, at 8:49 AM, Mamadou NGOM n...@numericap.com wrote:

> Hello everybody,

> When I call my extension the agi script don't work well. when I look at the 
> cli,
> that is what I have:

> AGI Tx >> agi_request: **.php
> AGI Tx >> agi_channel: SIP/myprovider-0007
> AGI Tx >> agi_language: fr
> AGI Tx >> agi_type: SIP
> AGI Tx >> agi_uniqueid: ***
> AGI Tx >> agi_version: 13.8.0
> AGI Tx >> agi_callerid:*
> AGI Tx >> agi_calleridname: unknown
> AGI Tx >> agi_callingpres: 0
> AGI Tx >> agi_callingani2: 0
> AGI Tx >> agi_callington: 0
> AGI Tx >> agi_callingtns: 0
> AGI Tx >> agi_dnid: 
> AGI Tx >> agi_rdnis: unknown
> AGI Tx >> agi_context: default
> AGI Tx >> agi_extension: 
> AGI Tx >> agi_priority: 13
> AGI Tx >> agi_enhanced: 0.0
> AGI Tx >> agi_accountcode:
> AGI Tx >> agi_threadid: *
> AGI Tx >> agi_arg_1: 56
> AGI Tx >>
> AGI Rx << SET VARIABLE ** 2
> AGI Tx >> 510 Invalid or unknown command
> -- AGI Script ***.php completed, returning 0

> I looked on the Internet but I saw a clear answer

> it is sure that it is for the compatibility between php5.6 and agi. if 
> somebody
> can help me.

Make sure there are no windows or dos line endings in that php script.  Try 
running it through dos2unix and see if that solves your issue.

Regards,
Michael

(elguero)

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Re: [asterisk-users] PRI timing settings

[Scott L. Lykens]

On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere 
j...@jeff.netmailto:j...@jeff.net wrote:

I wrote earlier today about a new PRI installation in the Caribbean, where all 
outbound calls are functioning fine *except* calls to Sprint phone numbers, 
which get rejected immediately as busy.

I don’t know what expectations for CLID your carrier might have, or for that 
matter the upstream carrier, however, we found through our CLEC here in the US 
that while the CLEC was happy to take e.164 formatted numbers from us as CLID, 
Global Crossing would reject them further upstream resulting in our calls to 
many toll frees being rejected.

Switching to 10 digit CLID on all outbound calls through that PRI solved the 
problem.

I don’t know if this is your problem but be sure your CLID is in the most 
simple format possible for your region to help rule it out.

sl
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[asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card

[Scott L. Lykens]

 reveals that the card generated 100,000 interrupts 
without being serviced and the kernel disabled it (and also reveals that the 
card is apparently on its own IRQ):

maintenance@sip:~$ cat /proc/interrupts
   CPU0   CPU1
  0: 46  0   IO-APIC-edge  timer
  1: 10  0   IO-APIC-edge  i8042
  7:  1  0   IO-APIC-edge
  8:  0  0   IO-APIC-edge  rtc0
  9:  0  0   IO-APIC-fasteoi   acpi
 12:  4  0   IO-APIC-edge  i8042
 14:  0  0   IO-APIC-edge  pata_amd
 15:  0  0   IO-APIC-edge  pata_amd
 16:304  0   IO-APIC-fasteoi   nouveau
 19:   1221  0   IO-APIC-fasteoi   eth1
 21:   8681  0   IO-APIC-fasteoi   sata_nv
 22:  0  0   IO-APIC-fasteoi   ehci_hcd:usb1
 23:  0  0   IO-APIC-fasteoi   ohci_hcd:usb2
 25: 10  1   IO-APIC-fasteoi   wct4xxp
NMI:  1  1   Non-maskable interrupts
LOC:  17884  19728   Local timer interrupts
SPU:  0  0   Spurious interrupts
PMI:  1  1   Performance monitoring interrupts
IWI:   1554815   IRQ work interrupts
RTR:  0  0   APIC ICR read retries
RES:   6566   8577   Rescheduling interrupts
CAL:220   4521   Function call interrupts
TLB:638504   TLB shootdowns
TRM:  0  0   Thermal event interrupts
THR:  0  0   Threshold APIC interrupts
MCE:  0  0   Machine check exceptions
MCP:  1  1   Machine check polls
ERR:  1
MIS:  0

Any ideas on how I can further diagnose and pursue this? Google does not reveal 
much related to this issue that is useful.

Thank you!

--
Scott L. Lykens
Keystone Medical Management Solutions, Inc.
+1 814 325-7500 x501 -- www.kmmsinc.comhttp://www.kmmsinc.com

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[asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card

[Scott L. Lykens]

 Just to be sure, what's the output of vmstat 10 10?

From within a minute or so of the system starting, keep in mind that the 
TE410P’s IRQ is disabled so the sys value is not representative of actual use 
had it been.

maintenance@sip:~$ vmstat 10 10
procs ---memory-- ---swap-- -io -system-- --cpu-
 r  b   swpd   free   buff  cache   si   sobibo   in   cs us sy id wa st
 0  0  0 7714300  42712 17629200   45837  369  369  1  3 91  4  0
 0  0  0 7714336  42720 17632400 0 4  194  396  0  0 99  0  0
 0  0  0 7714676  42720 17632400 0 5  197  397  0  0 100  0 
 0
 0  0  0 7714732  42736 17632400 0 8  216  443  0  0 99  0  0
 0  0  0 7714736  42744 17632400 0 2  195  395  0  0 99  0  0
 0  0  0 7714736  42744 17632400 0 0  200  420  0  0 99  0  0
 0  0  0 7714712  42752 17632400 0 4  205  414  0  0 99  0  0
 0  0  0 7714760  42804 17632400 023  216  430  0  0 98  2  0
 0  0  0 7714756  42812 17632400 0 4  201  409  0  0 99  0  0

Thank you.

--
Scott L. Lykens
Keystone Medical Management Solutions, Inc.
+1 814 325-7500 x501 -- www.kmmsinc.comhttp://www.kmmsinc.com

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Re: [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card

[Scott L. Lykens]

On Jun 1, 2014, at 11:01 AM, jg webaccounts...@jgoettgens.de wrote:

 Yes, I can see this. Another thing to check would be to start from a 
 different OS (eg from a USB stick) and see how the card behaves on the 
 otherwise same hardware.
 
 Since your ProLiant G2 server is almost 10 years old, and the TE410P works 
 with 3.3V only 
 (http://www.digium.com/en/products/telephony-cards/digital/quad-span), it 
 might be worth to check this.

The server is equipped with a 3.3v PCI-X slot. 
(https://h10057.www1.hp.com/ecomcat/hpcatalog/specs/provisioner/05/411095-421.htm).

It is an old server but it has worked just fine for the task of hosting 
Asterisk for some time and I prefer not to spend $2,000+ to replace both the 
server and the PCI card with more modern hardware. Admittedly, the TE410P is 
new to the equation in the last several months but only in the last few weeks 
has this really become a problem to the point of affecting use. In fact, I was 
on a call Thursday morning for about an hour that was entirely SIP but during 
that time the system started blocking and other users could no longer make 
calls - even though my call was unaffected.

The server is equipped with an AMD 8132 PCI-X bridge which apparently is known 
for being difficult in regards to interrupts. Google reveals that a few drivers 
have workarounds related to this chipset and to a range of revisions that mine 
happens to fall into.

I will build a live-cd based usb key later on today and test the hardware 
independent of its present OS.

Thank you.

Scott
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Re: [asterisk-users] Asterisk 11.10.0 Now Available

[Michael L. Young]

- Original Message -
 From: cov...@ccs.covici.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, May 29, 2014 6:42:05 PM
 Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available
 
   * ASTERISK-23754 - [patch] Use var/lib directory for log file
configured in asterisk.conf (Reported by Igor Goncharovsky)
 Is this mandatory -- what is wrong with /var/log/asterisk for those
 files?
 

The title on that issue is very misleading.  The patch that went in was just 
for chan_ooh323.  The change was to have chan_ooh323 use the log directory 
configured in asterisk.conf instead of using a hard coded value.

Michael

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Re: [asterisk-users] Login by AMI ok, by AJAM fails

[Michael L. Young]

- Original Message - 

 From: Michelle Dupuis mdup...@ocg.ca
 To: Asterisk Users List asterisk-users@lists.digium.com
 Sent: Friday, May 16, 2014 2:43:30 PM
 Subject: [asterisk-users] Login by AMI ok, by AJAM fails

 --
 root@apbx:/tmp# curl
 http://localhost:5039/asterisk/rawman?action=loginuser=testsecret=test
 [1] 15548
 [2] 15549
 root@pbx:/tmp# Response: Error
 Message: Authentication failed
 [1]- Done curl http://localhost:5039/asterisk/rawman?action=login
 [2]+ Done user=test

I believe it should be username instead of user for the query parameter.

Michael

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Re: [asterisk-users] Login by AMI ok, by AJAM fails

[Michael L. Young]

- Original Message -
 From: Michelle Dupuis mdup...@ocg.ca
 To: Asterisk Users List asterisk-users@lists.digium.com
 Sent: Friday, May 16, 2014 3:39:35 PM
 Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
 
 You're right - but I tried username too and it fails.  I can't
 understand why AMI authenticates and AJAM fails...
 

Have you taken a look at the Wiki yet?

https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP

In looking at that, I see some mistakes in what you are trying to do.  Please 
take a look at that and give it a try.

Michael

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Re: [asterisk-users] Login by AMI ok, by AJAM fails

[Michael L. Young]

- Original Message -
 From: Michelle Dupuis mdup...@ocg.ca
 To: Asterisk Users List asterisk-users@lists.digium.com
 Sent: Friday, May 16, 2014 4:29:05 PM
 Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
 
 From: asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com on behalf of Michael L.
 Young myo...@acsacc.com
 Sent: Friday, May 16, 2014 4:16 PM
 To: Asterisk Users List
 Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
 
 Have you taken a look at the Wiki yet?
 
 https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP
 
 In looking at that, I see some mistakes in what you are trying to do.
  Please take a look at that and give it a try.

Well, you need to login first.  Since you are using cURL, you need to turn the 
cookie engine on so that it will store and send cookies.

Also, you need to send the login request to 
http://localhost:5039/asterisk/manager?action=loginuser=testsecret=test; and 
not rawman.  Once you are logged in, then you can get raw manager output.

I hope that helps.

Michael

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Re: [asterisk-users] Login by AMI ok, by AJAM fails

[Michael L. Young]

- Original Message -
 From: Michael L. Young myo...@acsacc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Friday, May 16, 2014 4:55:30 PM
 Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
 
 - Original Message -
  From: Michelle Dupuis mdup...@ocg.ca
  To: Asterisk Users List asterisk-users@lists.digium.com
  Sent: Friday, May 16, 2014 4:29:05 PM
  Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
  
  From: asterisk-users-boun...@lists.digium.com
  asterisk-users-boun...@lists.digium.com on behalf of Michael L.
  Young myo...@acsacc.com
  Sent: Friday, May 16, 2014 4:16 PM
  To: Asterisk Users List
  Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
  
  Have you taken a look at the Wiki yet?
  
  https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP
  
  In looking at that, I see some mistakes in what you are trying to
  do.
   Please take a look at that and give it a try.
 
 Well, you need to login first.  Since you are using cURL, you need to
 turn the cookie engine on so that it will store and send cookies.
 
 Also, you need to send the login request to
 http://localhost:5039/asterisk/manager?action=loginuser=testsecret=test;
 and not rawman.  Once you are logged in, then you can get raw
 manager output.
 
 I hope that helps.
 
 Michael

Sorry that I messed up the thread while trying to un-top post your message.  
The above was in response to your prior message:

  I've done all of that (and I set the AJAM to listen to 5039).  What 
  mistakes do you see?

Michael

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Re: [asterisk-users] Security log format / content

[Michael L. Young]

- Original Message - 

 From: Michelle Dupuis mdup...@ocg.ca
 To: Asterisk Users List asterisk-users@lists.digium.com
 Sent: Thursday, March 27, 2014 12:55:21 AM
 Subject: [asterisk-users] Security log format / content

 I've noticed that the Asterisk (v11) security log captures attempts
 do dial without first authenticating, and places the number dialed
 into the accountid field.

 I'm trying to distinguish between failed attempts to register and
 attempts to dial without registering, but the security log treats
 them identically (using the accountid field for either the username
 or number dialed). I have noticed that the eventversion field is set
 to 2 for failed dial attempts, and 1 otherwise.

 Is this coincidence? Or can I rely on the eventversion=2 in the
 future to distinguish these two event types? (I've looked here:
 https://wiki.asterisk.org/wiki/display/AST/Security+Log+File+Format
 but it doesn't really help)

The eventversion field is just a way to distinguish different versions of the 
same event.  Between Asterisk 10 and 11, that particular event's logging output 
changed requiring a bump up in the version.  It should not be used to 
distinguish different events.

What do you mean by eventversion field is set to 2 for failed dial attempts, 
and 1 otherwise?  What is the event?  I have a feeling those are two different 
events.

You are correct about the events looking identical whether it is a failed 
registration or a failed dial attempt.  From the standpoint of Asterisk, an 
attempt was made to either register or place a call but the credentials failed. 
 Therefore, an InvalidPassword event is logged.

When an authorized device successfully places a call, you will only have a 
ChallengeSent entry in your log.

If an attempt to place a call is made and it does not respond back with the 
right credentials to the challenge sent to Asterisk, then you will have a 
ChallengeSent entry with a subsequent InvalidPassword.  You should be able 
to connect the two events based on the fields in those events.

If a successful attempt to register is made, you will have a ChallengeSent 
with a subsequent SuccessfulAuth.  If it is not successful, then you will 
have a ChallengeSent with a subsequent InvalidPassword.  Again, there 
should be enough information present with the other fields to help connect the 
events together.

The security events in Asterisk are designed to present the events.  It does 
not determine anything else for you.  You have to create a consumer of those 
events that can attempt to connect the dots for you.  Hopefully we are 
providing enough information for the consumer to do whatever you would like the 
consumer to do with the information.

I hope that helps.

Michael

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Re: [asterisk-users] Asterisk ignoring nat settings

[Michael L. Young]

- Original Message - 

 From: Andres and...@telesip.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Wednesday, January 15, 2014 7:51:28 PM
 Subject: Re: [asterisk-users] Asterisk ignoring nat settings

 
 Why don't you try with nat=yes. It should be equivalent to what you
 have but who knows. It might just work.

I am curious why you would say that nat=yes might work over 
nat=force_rport,comedia?  As you stated, they are the same.  nat=yes is 
deprecated and should be discouraged from being used.

Michael

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Re: [asterisk-users] Asterisk ignoring nat settings

[Michael L. Young]

- Original Message -
 From: Andres and...@telesip.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, January 16, 2014 4:17:53 PM
 Subject: Re: [asterisk-users] Asterisk ignoring nat settings
 
  I am curious why you would say that nat=yes might work over
  nat=force_rport,comedia?  As you stated, they are the same.
   nat=yes is deprecated and should be discouraged from being
  used.
 I had no idea it was deprecated.  I have never seen such a warning in
 Asterisk 1.8.X

The OP didn't specify which version of Asterisk he was using.  In Asterisk 1.8, 
nat was not a combinable list of options.  In Asterisk 11 it was.  So, I 
figured that since he was asking about nat=force_rport,comedia that he was on 
Asterisk 11 and in that version, nat=yes is deprecated.  I apologize about 
not clarifying the version that I was talking about.

Michael

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Re: [asterisk-users] 11.5.0 - SIP registration not retrying after failures

[Michael L. Young]

 From: Tony Mountifield t...@softins.co.uk
 To: asterisk-users@lists.digium.com
 Sent: Friday, November 8, 2013 10:39:25 AM
 Subject: [asterisk-users] 11.5.0 - SIP registration not retrying after
 failures
 
 I had a SIP problem on an 11.5.0 system that I look after. It
 registers
 with a SIP trunk provider, and at one point the provider had an issue
 that
 caused registration to fail.
 
 The problem was that Asterisk did not keep retrying, and it was not
 until
 it was restarted that registration was re-established.

A fix for this was actually just committed and will be in 11.7.  There is a 
release candidate available if you want to try it out.  You want to look for 
the register_retry_403 option that was added to the sip.conf file.

Michael


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Re: [asterisk-users] Asterisk 11.5 not honoring RTP port change in RE-INVITE

[Michael L. Young]

- Original Message - 

 From: Noah Engelberth nengelbe...@team-meta.net

 I have an Asterisk 11.5 system, using SIP Realtime and operating as a
 ITSP. One of my customer’s endpoints is a NetVanta 7100 PBX system
 that has a SIP trunk connection to my Asterisk box. The NV 7100 has
 a public IP on it that doesn’t have any NAT between it and my
 Asterisk system. When the customer transfers a call from one handset
 to a voicemail box, the NV 7100 sends a RE-INVITE to Asterisk with
 SDP information for a different RTP port number. Asterisk is ACKing
 the RE-INVITE, but never changes media over to the new port number.

 AdTran is saying it’s Asterisk’s problem, since the Wireshark trace
 shows Asterisk is ACKing the re-invite but not changing ports. I do
 see that the Session ID number is different in the two invites (the
 REINVITE has a higher ID number than the original 200 OK that sets
 up the call – my test call was inbound to the NV7100). However, the
 REINVITE’s version number is lower (1) than the 200 OK’s SDP version
 number (which was the same as the SDP Session ID number). I see in
 the sip.conf.sample file that “By default, Asterisk will honor the
 session version number in SDP packets and will only modify the SDP
 session if the version number changes”. Given that I don’t have
 ignoresdpversion=yes either globally or for this peer, does this
 mean that Asterisk will only honor new SDP packets if the version is
 higher, or will it honor any change? Or should I be looking
 somewhere else?

You have pretty much found what the issue is.  The AdTran is not properly 
incrementing the SDP version.

Look at the comments on these issues for clarification on why Asterisk is 
actually following the RFC3264:

https://issues.asterisk.org/jira/browse/ASTERISK-20633
https://issues.asterisk.org/jira/browse/ASTERISK-20642
https://issues.asterisk.org/jira/browse/ASTERISK-21411

RFC3264
If the offered SDP is different from the previous SDP, some constraints are
placed on its construction, discussed below.

Nearly all aspects of the session can be modified. New streams can
be added, existing streams can be deleted, and parameters of existing
streams can change. When issuing an offer that modifies the session,
the o= line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP.

Therefore, in order to work with devices that do not handle the SDP version 
properly, sip.conf has the ignoresdpversion option.

Michael
(elguero)

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Re: [asterisk-users] External sip phones register with the servers IP...

[Michael L. Young]

- Original Message -
 From: Carlos Chavez cur...@telecomabmex.com
 To: asterisk-users@lists.digium.com
 Sent: Thursday, August 1, 2013 8:41:19 PM
 Subject: [asterisk-users] External sip phones register with the servers IP...
 
 We have just updated our office server to Asterisk 11.4.0 from 1.8.15
 and
 internally everything is working fine.  The problem we are having is
 that we
 cannot use any external phone connected through the Internet.  This
 used to
 work fine with 1.8 but since the upgrade whenever you register any
 phone from
 an outside network the phone tries to register using the servers
 internal IP.
 
 I endo up having something like this:
 
 Sending to 187.163.93.235:58545 (no NAT)
 -- Registered SIP '2003' at 192.168.2.50:58545
 Reliably Transmitting (no NAT) to 192.168.2.50:58545:
 OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0
 Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0
 Max-Forwards: 70
 From: asterisk sip:asterisk@192.168.2.50;tag=as4ed13172
 To: sip:2003@192.168.2.50:58545;ob
 Contact: sip:asterisk@192.168.2.50:5060
 Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060
 CSeq: 102 OPTIONS
 User-Agent: Asterisk PBX 11.4.0
 Date: Fri, 02 Aug 2013 00:27:48 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
 INFO, PUBLISH
 Supported: replaces, timer
 Content-Length: 0
 
 I really cannot understand what is wrong, I have checked my sip.conf
 configuration and it is the same as in past versions.  externaddr and
 localnet
 are set to the proper values.  Any ideas?

Did you look at the CHANGES file?  There are new settings for NAT.  If you are 
using the same settings as in 1.8, there is a posiblity that you will have 
problems depending on what settings you have (which you did not include in this 
message).

Also, I would recommend 11.5 since there was a one-way audio issue fixed 
related to using the two new NAT settings.

-- Michael 
(elguero)

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Re: [asterisk-users] Playing a sound file during a call

[Michael L. Young]

- Original Message -
 From: Richard Mudgett rmudg...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, May 2, 2013 8:24:49 PM
 Subject: Re: [asterisk-users] Playing a sound file during a call
 
  On Thu, May 2, 2013 at 3:37 PM, Carlos Alvarez
  car...@televolve.com
  wrote:
 
  In case anyone else sees this discussion in the future, the
  Set(__DYNAMIC_FEATURES) line can't be over a certain length or it
  stops parsing anything after that.
   
 
 You can also put dynamic feature group names into the
 DYNAMIC_FEATURES list.

Also, I know this doesn't help you now but, in Asterisk 12 the limit has been 
eliminated.

Take a look at https://issues.asterisk.org/jira/browse/ASTERISK-20680

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Re: [asterisk-users] rtcachefriends and rtautoclear on change password

[Michael L. Young]

- Original Message -
 From: Leandro Dardini ldard...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, March 26, 2013 5:28:22 AM
 Subject: [asterisk-users] rtcachefriends and rtautoclear on change password
 
 Hello friends,
 I am using from a long time rtcachefirends=yes and rtautoclear=yes in
 my sip.conf for asterisk 11.2.1.
 
 I have found the data of the peers are never reloaded from the
 database, so if you change the password for a peer, it will continue
 to work with the old password. Do you think it is the expected
 behaviour?
 
 From the documentation for rtautoclear=yes
 
 If set to yes, when the registration expires, the friend will
 vanish from the configuration until requested again. If set
 to an integer, friends expire within this number of seconds
 instead of the registration interval.
 
 The phone will renew the registration before it expires, so maybe it
 never expires.
 
 I have tried to set the rtautoclear to 60, but the result is the
 same,
 the new password is never enforced.
 
 Any suggestion apart from removing the rtcachefriends?

With rtcachefriends turned on, the realtime peer is cached in memory.  
Therefore, in order to clear the cache for that peer, you should check into 
issuing the command sip prune realtime peer peername if you want to clear 
out only the one peer.  If you want to reload the peer back in memory after 
clearing it out, you can issue sip show peer peername load to load it back 
from the db.

Michael

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Re: [asterisk-users] Asterisk 1.8 and dual stack support

[Michael L. Young]

- Original Message -
 From: Jaap Winius jwin...@umrk.nl
 To: asterisk-users@lists.digium.com
 Sent: Thursday, March 21, 2013 12:47:57 PM
 Subject: [asterisk-users] Asterisk 1.8 and dual stack support
 
 Hi folks,
 
 Following an upgrade to Debian wheezy, I'm now running Asterisk
 1.8.13.1.
 As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version
 can
 support IPv6. However, it seems that I can't get it to support both
 IPv4
 and IPv6 at the same time. For example, if in sip.conf I set the
 bindaddr
 variable to '::' it will only listen on IPv6 and none of my IPv4-only
 friends and peers will be able to connect to it. On the other hand,
 if I
 set it to '0.0.0.0' then it will not listen on IPv6.

How are you determining that it is not listening on IPv4?

bindaddr=:: should allow you to support dual stack.

Michael


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Re: [asterisk-users] Asterisk 1.8 and dual stack support

[Michael L. Young]

- Original Message -
 From: Jaap Winius jwin...@umrk.nl
 To: asterisk-users@lists.digium.com
 Sent: Thursday, March 21, 2013 5:27:37 PM
 Subject: Re: [asterisk-users] Asterisk 1.8 and dual stack support
 
 That's what I thought would happen. When I set bindaddr=:: and use
 'netstat -lpn |grep 5060' it shows:
 
   udp6 0   0 :::5060   :::* 9898/asterisk
 
 Services like this usually also support IPv4 and as much is suggested
 by
 this comment in the sip.conf that comes with my Asterisk package:
 
   ; (Note that using bindaddr=:: will show only a single
   ; IPv6 socket in netstat. IPv4 is supported at the same
   ; time using IPv4-mapped IPv6 addresses.)
 
 However, the moment I reload my sip.conf with bindaddr=::, my entire
 list
 of IPv4-only peers loses contact with Asterisk with warnings about
 the
 network being unreachable. So, it would appear that the version of
 Asterisk that I'm using is operating with a single stack socket.

Let me try to understand this.  With bindaddr set as bindaddr=::, upon 
starting Asterisk, you are fine and all your IPv4 peers connect properly.  
Therefore, dual stack is working at this point.  Upon issuing a sip reload, 
your peers lose their ability to communicate with Asterisk?  Is that correct?  
What does netstat -lpn |grep 5060 show after the reload?

These network unreachable warnings are from Asterisk or your peers?

What version of Asterisk are you using?

Asterisk 1.8.0 had IPv6 support in it.  Therefore, every minor version released 
since would still have IPv6 support in it.

Michael

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Re: [asterisk-users] Asterisk 1.8 Streaming MOH timing interface

[Michael L. Young]

- Original Message - 

 From: Bob Pierce westman...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Cc: g...@westmancom.com
 Sent: Monday, February 4, 2013 6:14:26 PM
 Subject: [asterisk-users] Asterisk 1.8 Streaming MOH timing interface

 We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just
 today our streaming music on hold stopped working. I remember when
 we had first installed 1.8 we had an issue where the streaming music
 on hold would not work because Music On Hold was using the DAHDI
 timing module. We needed the DAHDI timing module loaded so that
 paging would work. However, at that time we upgraded to 1.8.5.0 and
 the system loaded properly with both the dahdi and pthread timing
 module with Music On Hold using the pthread timing module. In that
 state, everything worked properly - Streaming Music On Hold worked
 as well as Paging. That has all continued to work properly for the
 last 40 weeks.

 I'm wondering of for some reason the Music on Hold service is now
 using the DAHDI timing module because when I do module show like
 timing I see:
 CLI module show like timing
 Module Description Use Count
 res_timing_dahdi.so DAHDI Timing Interface 33
 res_timing_pthread.so pthread Timing Interface 0
 2 modules loaded

 I believe that the pthread used to show a use count of at least 1
 with the Music On Hold service using that timing source. I suspec
 that if I restart the Asterisk service everything will come back up
 the way that it did last time. However, I'm wondering if there would
 be a way to switch the Music On Hold module back to using pthread
 timing without restarting the Asterisk service.

Bob, I would recommend upgrading to the latest version.  There have been a lot 
of security and bug fixes since 1.8.5.  There was a bug fixed, over a year ago 
(1.8.9), which sounds exactly like what you are experiencing.  The latest 
version is 1.8.20.1.

Regards,
Michael

(elguero)

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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

- Original Message - 

 From: Logan Bibby lo...@keobi.com

 Does anyone have a good contact for their sales? I've attempted
 calling their Enterprise sales a few times and was just spun around
 in circles. Having a sales rep I can just call would be awesome.

Logan,

We have an account manager that we deal with directly for changes or new 
orders.  Supposedly, every customer has their own account manager.

Michael

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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

 From: Carlos Alvarez car...@televolve.com

 It may be too late for this, but in working with another RBOC who
 didn't want to deal with Asterisk, I just asked what they do
 support, and modified the headers sent by Asterisk to claim that it
 was one of the devices on that list. Done.

Like everyone else, I was laughing as well when I read this.

One engineer stated that they like to have an SBC to manipulate the headers to 
normalize things.  I stated that Asterisk was capable of manipulating headers 
if need be.  You just proved that it works :)

Michael

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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

- Original Message -
 From: Matthew J. Roth mr...@imminc.com

 Your email documents the same experience we had years ago.  It was
 strange reading it and I was shocked that nothing has changed in that
 much time.  Asterisk will work with Verizon's IP trunking product,
 but
 they're trying to make you jump through some old hoops first.

Those were my thoughts.  They are making this a lot more complicated than it 
really needs to be.  I think the main thing they are worried about is having to 
support something they don't know anything about.  Well, they won't have to 
support it; we will.  Just as long as they are SIP compliant, there should be 
no issues.

 We were using Verizon IP trunks over an MPLS network in 2008.  At the
 time, they did not require IPSEC for signaling.  However, they did
 want us to install an SBC and actually provided us with an AudioCodes
 nCite 1000 at their cost.  It just acted as a proxy, so it didn't
 affect interoperability with Verizon's IP trunks and I wouldn't
 buy one only to satisfy them.

One of the engineers stated that they have received the direction to only use 
standard equipment.  So, they are afraid that our setup will not pass ICB 
since it doesn't fit into their standard way of doing things.

 We were quite happy with the service, so I'd encourage you to go
 ahead
 with the field trial without putting an SBC in place.  Remember that
 you will be paying them, so they should be working to fit your design
 and if they reject you for some arcane reason then you are better off
 with another provider anyway.
 
 Don't hesitate to let them know that you know you're jumping through
 the same hoops that have been in place since 2008 and you'd
 appreciate
 it if they would streamline the process to save time and money.  Tell
 them that Asterisk should already be on their certified list of
 approved devices because they've been running field trials and
 production setups on it for years.

It is good to hear that you were happy with the service.  I have my 
reservations with all the hoops they are making us jump through and that gives 
me a bit of confidence that it will be worth it.

I did tell them that Asterisk is being used all over the place as well as in 
big call centers.  I know that I have seen others in the Asterisk community on 
Verizon.  Verizon seems to be hung up on this certification stuff and it is 
hard to explain to them that this is not a piece of hardware you buy and 
plugin.  We can build our own servers and put Asterisk on it, and they seem to 
cringe when they hear that.

Thanks for your input Matthew.  It is appreciated.

Michael

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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

- Original Message - 
 From: Carlos Alvarez car...@televolve.com

 Sounds like the same huge effort it takes to work with
 Qwest/Centurylink, and in the long run we found it simply isn't
 worth it. The few benefits of working with an RBOC are countered by
 the many drawbacks of working with an RBOC.

 Also we recently acquired a half million minutes/mo from a company
 who was tired of dealing with Qwest SIP. They said the same thing I
 said above.

 I suppose the point of what I'm saying is you should really think
 about what you think you will gain from a relationship with them,
 and whether all this is worth it (all this means now and how their
 attitude will affect you forever).

Trust me, this was not my choice... They are not fun to deal with when it came 
to our PRI lines.  After dealing with dropped calls and errors on the T1s, they 
wouldn't admit that they had a problem until finally they looked at the 
hardware at the CO while we were down hard (which cost us about 4 - 6 hours 
downtime) and said, Oh, we do have a problem.  To make a long story short, it 
was fixed and has been good since but I was really trying to move us away from 
Verizon.  Unfortanately, it boils down to cost and Verizon being as big as they 
are were able to make a deal (getting us out of contracts that had been signed, 
credits, etc.) that the ultimate decision maker here at the company went for.  
That decision maker also has the mindset that we have to stick with the phone 
company for some reason.  I was strongly against it and wanted to go with a 
different company.  So, I have to deal with it now.

Thanks for your input.  It pretty much echoes my sentiments.

Michael

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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

- Original Message -
 From: Matthew J. Roth mr...@imminc.com

 At least Verizon maintains a consistent customer experience.  ; )
 
 Overall, we've found the service to be reliable and stable, but when
 there are problems or changes needed you're dealing with Verizon and
 the
 w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y.

Haha... that is funny... it is sooo true.

Well, you are right.  Once it is working, it is usually pretty stable.  Just a 
pain in the butt when things are not working.  Hopefully we can get through the 
Field Trial and that is all I have to worry about for a while.

Thanks Matthew for all the encouragement as I go down this temporary (I hope) 
unpleasant path.

Michael

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[asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

All,

We are in the process of trying to setup our network to use Verizon's SIP 
trunking product.  They say that since Asterisk is not on their certified 
list of approved devices, we need to go through a field trial to get it 
approved before allowing us to use their service.

Where we are at is getting the design approved.  We are trying to watch our 
budget at the same time.  We have used other providers without any issues with 
our current setup but it seems that Verizon has their own standards when it 
comes to this and they don't seem very keen on linux and open source.  Yet, 
they are willing to work with us and want to see the field trial succeed 
instead of being rejected from another group within Verizon who will have to 
approve the final design.

Has anyone in the community had experience with Verizon and their SIP product?  
Were you able to get through the field trial successfully?

What was the design that you used to get Asterisk certified with Verizon's 
network?

Where I am at is that they want us to use an SBC.  One engineer asked about 
Cisco Call Manager.  I told them that basically if I can accomplish the same 
thing with a Linux box (routing box and sip proxy box) without having to spend 
money on SBCs or expensive Cisco gear, that is the route we would like to go.  
We are looking at the possibility of handling 140 concurrent calls... that is 
what they are designing on their end as well.

So, I am asking the community for any input.  I have read on here and seen on 
IRC that some in the community are successfully using Asterisk with Verizon 
SIP.  Verizon was going to check and see if they have any notes about that and 
those particular setups.  Can anyone help share any information or tidbits on 
how they were able to sucessfully work with Verizon?

Thanks,

-- 
Michael L. Young 

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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Michael L. Young]

- Original Message -
 From: Steven Howes steve-li...@geekinter.net

 I *think* Verizon require IPSEC for the signalling, so it may be
 worth reading up on configuring IPSEC in Linux (or acquiring
 additional hardware) whilst you're looking at the Asterisk part.
 This could have just been for a specific product / contract or
 something, I don't recall the details exactly.

I should have probably stated that this is going to be going through an MPLS 
network being setup with Verizon.  They may not be requiring that since it is 
within their network, not going over the internet.  They have not said anything 
about the the need to secure the traffic coming from them or to them since the 
VoIP traffic will be on Verizon's network.

Thanks for the heads up, though.  I will keep that in mind.

Michael

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Re: [asterisk-users] * Waiting for asterisk to shutdown .............

[Michael L. Young]

- Original Message -
 From: Joseph syscon...@gmail.com
 To: asterisk-users@lists.digium.com
 Sent: Saturday, November 24, 2012 12:54:12 AM
 Subject: [asterisk-users] * Waiting for asterisk to shutdown .
 
 I'm running asterisk on a small box,
 Intel-R-_Atom-TM-_CPU_330_@_1.60GHz
 and when I try to restart the asterisk it fails.
 
 /etc/init.d/asterisk restart
   * Caching service dependencies ...   [ ok ]
   * Killing wrapper script ... [ ok ]
   * Stopping asterisk PBX gracefully ...
 
 * Waiting for asterisk to shutdown
 .
   * Failed.
 
 When I run /etc/init.d/asterisk status
 I get: * status: started
 
 At this point I have to kill the process ID
 zap it (/etc/init.d/asterisk zap)
 and restart it.
 
 Why asteriks can not shut down properly?
 How can I monitor this process and restart it?

What version of Asterisk are you running?

Michael
(elguero)

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Re: [asterisk-users] SIP Debugging Information..

[Michael L. Young]

- Original Message -
 From: Howard Leadmon how...@leadmon.net
 To: asterisk-users@lists.digium.com
 Sent: Saturday, November 24, 2012 3:19:10 PM
 Subject: [asterisk-users] SIP Debugging Information..
 
 
  I did a little googling, but didn't seem to find anything specific
  to
 answer the question.   I am trying to debug some calls on an Asterisk
 system
 (AsteriskNow) that are dropping, and when the general logs didn't
 nail
 anything I turned on SIP Debugging on the trunk to the provider.
 Basically the complaint is that when some call in, regardless of if
 the call
 is answered, or if Vmail answers it, it drops the calls in a matter
 of
 seconds.   The strange thing is, that the system processes many
 hundreds of
 calls daily, but only a couple specific incoming callers are seeing
 the
 drops.  I would have thought a NAT issue, but why does this only
 affect a
 specific group of incoming callers, the rest go about their business
 just
 fine.  I think thinking bandwidth.com is mucking something up, but
 again I
 have no specific proof one way or another, so why the debugging.
 
  When one of the problem callers is dropped, in the SIP debugging I
  see:
 
   chan_sip.c: Scheduling destruction of SIP dialog
 '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE)
 
  
 Is this the remote end (ie bandwidth.com) dropping the call, or is
 the local
 Asterisk server dropping the call?

[snip]
 ---
 [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c:
 --- SIP read from UDP:216.82.224.202:5060 ---
 BYE sip:4104159270@10.98.4.36:5060 SIP/2.0
 Record-Route: sip:216.82.224.202;lr;ftag=gK0b66d829
 Record-Route: sip:67.231.4.93;lr=on;ftag=gK0b66d829
 Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKe902.53bde7e.0
 Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bKe902.32697e93.0
 Via: SIP/2.0/UDP 192.168.27.72:5060;branch=z9hG4bK0bBac8c2c3cb90659df
 From: sip:7173381800@192.168.27.72;isup-oli=0;tag=gK0b66d829
 To: sip:+14104159270@67.231.4.93;tag=as0850c6db
 Call-ID: 285991942_79966325@192.168.27.72
 CSeq: 297 BYE
[snip]

If I am reading this right, it looks like a BYE is coming in from the far end, 
Bandwidth.com.

Michael
(elguero)

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Re: [asterisk-users] Intruder

[Michael L. Young]

- Original Message - 

 From: Felix Vazquez felix.vazq...@theboshgroup.com
 To: asterisk-users@lists.digium.com
 Sent: Friday, November 16, 2012 11:20:46 AM
 Subject: [asterisk-users] Intruder

 I am in the asterisk CLI and can see an unidentified caller trying
 the make calls out of the asterisk system. How do I stop them? How
 do I identify them and how can I see how the go in?

 This is an example of what I would see:

 NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call from '' to
 extension '90111235551212' rejected because extension not found.

I would recommend you read README-SERIOUSLY.bestpractices.txt, top level of 
source code.

Another thing you can do is turn on security logging if you are using Asterisk 
10/11.  Take a look at logger.conf.  It may provide you with some extra 
information on who is trying to make the call.

Take a look at this page:
https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations

I would recommend using fail2ban as well.

Michael
(elguero)


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Re: [asterisk-users] Error: astobj2.c: refcount -1 on object

[Michael L. Young]

- Original Message -
 From: Ishfaq Malik i...@pack-net.co.uk
 To: asterisk-users@lists.digium.com
 Sent: Wednesday, November 14, 2012 4:05:21 AM
 Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object
 
 Hi
 
 I'm using 1.8.7.0. This morning I got an alert telling me
 
 Asterisk on my-host exited on signal 11.  Might want to take a
 peek.
 
 When I had a look at the logs I can see a lot of errors like
 
 ERROR[14924] astobj2.c: refcount -1 on object 0x2aaad4069068
 ERROR[14924] astobj2.c: refcount -2 on object 0x2aaad4069068
 
 All the way up to
 
 ERROR[14924] astobj2.c: refcount -2004 on object 0x2aaad4069068
 
 Everything up to this point was completely normal.
 
 Does anyone know what this error means and what causes it?

I would recommend updating to the latest version.  We are up to 1.8.18 and 
1.8.19 is around the corner.  There have been a lot of bug fixes and you might 
find that whatever caused this issue is already fixed.

Michael
(elguero)

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Re: [asterisk-users] Error: astobj2.c: refcount -1 on object

[Michael L. Young]

- Original Message -
 From: Ishfaq Malik i...@pack-net.co.uk
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, November 14, 2012 9:25:37 AM
 Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
 
 Thanks for the advice but that's not really a quick and easy option
 for
 us. We would not be able to upgrade to another version without doing
 full regression testing on the candidate upgrade version and we've
 been
 using this version for at least half a year and this is the first
 time
 we've had this crash and this error.
 
 Also, just upgrading doesn't enlighten me to what is going on to
 cause
 this error.

Sorry, I thought I was offering a quick and easy option.  Since, this is a 
minor release upgrade, there shouldn't (won't say there won't) be any changes 
to cause you problems.  But, I understand you have to follow your procedures 
before upgrading.  Software is not perfect.

Since this is not happening all the time, it may take a while for you to try 
and figure out what is wrong; that is if you can reproduce it.  Therefore, in 
my opinion, I think you would be better off using your time to consider 
upgrading especially with a lot of bugs and security updates being in the 
latest version.  Use that time to run through your regression testing.

Anyways, if you want to go the path to try and figure out what caused this, I 
beleive you will need to look at the following information:

https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging

Hope that helps,

Michael
(elguero)

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Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

[Michael L. Young]

- Original Message -
 From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, November 13, 2012 4:35:54 AM
 Subject: Re: [asterisk-users] Asterisk crashing when trying to start a Jabber 
 session with ejabberd
 
 On 11/13/2012 12:11 AM, Phil Reynolds wrote:
 [snip]
  It turns out to be a known issue:
 
  https://issues.asterisk.org/jira/browse/ASTERISK-19532
 
  ... and can be fixed by applying the patch at:
 
  https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch
 
  I will file the details with Debian too...
 
 Is it an omission that this fix has not been applied to the 11 tree?
  From the looks of ASTERISK-19532 it seems that the fix has only been
 applied to 1.8 and 10.
 

If you click on the link for ASTERISK-19532, there is a tab in the Activity 
section labeled Subversion.  It shows that the patch was applied to 1.8, 10, 
11 and trunk.

Michael
(elguero)

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Re: [asterisk-users] 11.0.1: more sip registry woes

[Michael L. Young]

- Original Message -
 From: sean darcy seandar...@gmail.com
 To: asterisk-users@lists.digium.com
 Sent: Wednesday, November 7, 2012 9:20:58 AM
 Subject: Re: [asterisk-users] 11.0.1: more sip registry woes
 
 On 11/06/2012 09:45 PM, Michael L. Young wrote:
  - Original Message -
  From: sean darcy seandar...@gmail.com
  To: asterisk-users@lists.digium.com
  Sent: Tuesday, November 6, 2012 7:51:04 PM
  Subject: [asterisk-users] 11.0.1: more sip registry woes
 
  Upgrade to 11. This worked on 10.X.X
 
  sip.conf:
 
  register=myusername:password@nyc.teliax.net
 
  telnet  nyc.teliax.net 5060
  Trying 8.14.120.23...
  Connected to nyc.teliax.net.
  Escape character is '^]'.
 
  sip show registry
  Hostdnsmgr Username
Refresh
  StateReg.Time
  nyc.teliax.net:5060 N  my user name
120
  Unregistered
  1 SIP registrations.
 
 
  Nothing on the cli to show any problems.
 
  teliax says no problems on their end.
 
  In 10 if I wasn't registered I got lots on registration failed
  messages.
 
  Added this to sip.conf:
 
  registertimeout=20 ; retry registration calls every 20
  seconds (default)
  registerattempts=0   ; if 0 try forever
 
  which is supposed to be the default anyhow.
 
  I am registered without any problems to nyc.teliax.net.
 
  How is your peer definition set in sip.conf?  Try turning verbosity
  up on the console and also set sip debug on for your peer in
  order to see the communication between your server and Teliax.
   Hopefully, that will provide some clues as to why you are not
  registering.
 
  Michael
  (elguero)
 
 
 Are you running 11.0.1?
 
 sean

I am running trunk which is essentially 11.0.1 since there have not been any 
changes made in that area of the code since it was branched.

Michael
(elguero)

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Re: [asterisk-users] 11.0.1: more sip registry woes

[Michael L. Young]

- Original Message -
 From: sean darcy seandar...@gmail.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, November 6, 2012 7:51:04 PM
 Subject: [asterisk-users] 11.0.1: more sip registry woes
 
 Upgrade to 11. This worked on 10.X.X
 
 sip.conf:
 
 register=myusername:password@nyc.teliax.net
 
 telnet  nyc.teliax.net 5060
 Trying 8.14.120.23...
 Connected to nyc.teliax.net.
 Escape character is '^]'.
 
 sip show registry
 Hostdnsmgr Username   Refresh
 StateReg.Time
 nyc.teliax.net:5060 N  my user name
  120
 Unregistered
 1 SIP registrations.
 
 
 Nothing on the cli to show any problems.
 
 teliax says no problems on their end.
 
 In 10 if I wasn't registered I got lots on registration failed
 messages.
 
 Added this to sip.conf:
 
 registertimeout=20 ; retry registration calls every 20
 seconds (default)
 registerattempts=0   ; if 0 try forever
 
 which is supposed to be the default anyhow.

I am registered without any problems to nyc.teliax.net.

How is your peer definition set in sip.conf?  Try turning verbosity up on the 
console and also set sip debug on for your peer in order to see the 
communication between your server and Teliax.  Hopefully, that will provide 
some clues as to why you are not registering.

Michael
(elguero)

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Re: [asterisk-users] Case-sensitivity of Dialplan variables.

[Michael L. Young]

- Original Message -
 From: Ira i...@extrasensory.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, October 3, 2012 3:21:50 AM
 Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
 
 At 07:59 PM 10/2/2012, you wrote:
 
 While true that most users are probably not programmers, most people
 administering Asterisk would be system / network admins,
 correct?  System admins and networking admins are used to working in
 environments such as Linux where variables and file names are case
 sensitive.
 
 If someone is moving from a GUI interface to CLI, then they
 would/should know that case sensitivity is important and therefore
 the change shouldn't pose a problem.
 
 I'm not a system / network admin, at least not for Linux. I have one
 Linux machine, it runs Asterisk and Samba. I can usually make
 Asterisk do what I want. Samba works but I have little to no idea
 why.  I run yum update occasionally and I run V11 trunk or whatever
 the proper name would be for the development version.

I can think of some situations where case sensitivity could be a problem.  I 
hope I am not out in left field with my thinking.   Asterisk can be found in 
companies that have several offices.  Asterisk could be used in a cluster.  
Asterisk may be administered by many different folks at a company and probably 
more than one Asterisk box.  If those individuals are expecting the variables 
to be case sensitive, it becomes a problem trying to debug problems in the dial 
plan.  They may not know that an individual in one office is doing things one 
way because they are not expecting variables to be case sensitive while another 
individual is expecting things to be case sensitive.  It really can create a 
lot of trouble and confusion in bigger deployments versus a single individual 
administering his own box.
 
 If there was a compiler and declared variables then case makes
 perfect sense. Without that, I'd never get a C program to work.
 
 I know people want case sensitivity, it's the right way to do it,
 but how does it help Asterisk?

This helps Asterisk by following a more or less established standard that 
everyone expects.  I believe that this case-insensitivity in the dial plan 
actually came as a surprise to some who had never stumbled across it before.  
Again, those with experience in unix/linux environments have been trained that 
variables are case sensitive and they do not have to be programming in C to 
know that.

 Does anyone have configurations that would be broken by case
 insensitivity?

Some people might have broken dial plans and that is why this was brought up on 
the list in order to gain attention and feedback.  But, it will only break for 
the next release.  It won't affect current releases.  Instead, Mark is planning 
on documenting the current behavior on the Asterisk wiki.  From what I am 
observing so far, it looks like it may only affect a small number of people.  
My feeling is that the majority may have already been using variables as if 
they were case sensitive already.  That was how variables were documented on 
the Asterisk wiki... as being case sensitive.

 If not, then what is the upside of enforcing case sensitivity?

The upside is that we have consistency.  This helps to keep bug reporrts to a 
minimum and in my opinion helps the end user not to create problems for 
themselves.  The example mentioned in the issue being worked on, is say, an 
application is expecting the variable ${MIXMONITOR_FILENAME}.  A user thinks, 
Hey, the dial plan is case insensitive and uses ${mixmonitor_filename} or 
${MixMonitor_FileName} to set the file name.  They find out that the variable 
is being ignored.  They later check the variable ${MIXMONITOR_FILENAME} (notice 
all uppercase) in the dial plan and it shows him that it is set.  They then 
think there is a bug in Asterisk... well, the problem is that they didn't set 
the variable according to what app_mixmonitor is expecting.  The application IS 
case sensitive when it comes to variables.  So, this is the confusion that can 
be caused by having one part of Asterisk be case sensitive and another part of 
Asterisk NOT be case sensitive.

I hope this explanation helps those reading this to understand better what is 
trying to be resolved here.  At least, this is the way I am understanding the 
reason for the proposal presented to the list.

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Re: [asterisk-users] Case-sensitivity of Dialplan variables.

[Michael L. Young]

- Original Message -
 From: Matthew Jordan mjor...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, October 3, 2012 12:17:56 PM
 Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
 
 From Mark's original e-mail:
 
 Some of you might be eager to propose a configuration option to
 decide which it should be. I'm sick of having hundreds of options
 in Asterisk to slightly tweak the behavior one way or another. This
 needs to go one way or the other, not be configurable.
 
 I can't overstate how much I agree with this.  A configuration option
 to
 'tweak' the behavior in pbx.c is much more likely to introduce
 problems than
 solve them.  If a clear consensus cannot be reached, I'd err on the
 side
 of doing nothing than put in yet another configuration option.

I agree that a configuration option is not the solution.  I am not seeing what 
the big deal is.  Software changes between major releases.  Someone is not 
going to, or at least they shouldn't if their livelihood depends on it, upgrade 
their machines without doing the proper preparation for upgrading.  That means 
reading the UPGRADE.txt file and outlining what needs to be done to upgrade 
their system if there are features they need in the new version or simply want 
to be on the latest version.  Then they should test those changes as well 
before putting it into production.

We are probably a year away from seeing a release for the version of Asterisk 
where this change would occur.  We are two years away from an LTS version of 
Asterisk.  So, I think there would be plenty of time for evaluation and testing 
to be performed by those affected.  Especially, as in the case of what Raj 
mentioned at the beginning of his prior email, not too many people may even be 
affected by this change just like he won't be.

Michael L. Young
(elguero)

PS:  If you can't tell, I am really for this change and doing so without any 
configuration options :)

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Re: [asterisk-users] Case-sensitivity of Dialplan variables.

[Michael L. Young]

- Original Message -
 From: Vladimir Mikhelson v...@mikhelson.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 2, 2012 9:02:18 PM
 Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
 
 
 On 10/1/2012 4:15 PM, Mark Michelson wrote:
 
  What I plan to do, no matter which way the vote goes, is to
  document
  on the wiki how things currently behave in Asterisk, to include the
  example I gave above (or something similar anyway). Depending how
  the vote goes, I will make the necessary code changes in Asterisk
  trunk. I will document the behavior change both in UPGRADE.txt and
  on the wiki.
 
  When considering which way you lean, consider that we really don't
  have much of a precedent to go on. For instance, dialplan
  applications
  are case-insensitive (answer and Answer and ANSWER are all
  the
  same). Dialplan functions, on the other hand, are case sensitive
  (HASH would be evaluated properly but hash would not). My
  personal
  opinion is that all variable evaluations should be case-sensitive.
  I don't feel all that strongly about it though and could easily be
  swayed the other way if people respond overwhelmingly in
  opposition.
 

 
 First you need to consider compatibility with currently supported
 packages which include auto-generated dial plans like AsteriskNow,
 PIAF,
 etc.  If you plan to break their functionality you need to at least
 coordinate your move with the maintainers.

This change would go into trunk, the development line of code.  So, I am not 
sure that a coordinated effort would need to be made here.  As with every major 
release, maintainers of external packages usually check the UPGRADE.txt.  There 
almost always will be something to change between major versions in order to 
keep in step with the new release.
 
 Then you may want to consider backwards compatibility with packages
 still widely used but not actively supported any more like Trixbox.
 Maybe not the best example as their WEB site says, This is the
 current
 stable release based on Asterisk 1.6.

Again, I am not sure there is a need for backwards compatibility.  This change 
would go into Asterisk 12, which is not LTS.  I would think that most packages 
would be focused more on working with the LTS version of Asterisk.  I could be 
wrong though.  Given that Asterisk 11 is the current LTS that will be probably 
be released soonish out of beta, the next LTS version is a couple of years away 
giving plenty of time for people to make the necessary changes.
 
 If you really want to make it not settable (and this is big, not a
 minor
 change, if I were you I would definitely make it settable) then I
 would
 go with case-insensitive as it allows for various custom notations,
 e.g.
 Hungarian notation in naming of custom variables without a later
 painstaking investigation whether nCallID is equal to nCallId or
 not.  Consider the fact, most of the dial plan debugging happens in
 the
 logs or in the Console Screen.  Someone may want to spell nCALLID
 just
 to be able to see the difference between Latin l and I where the
 first one is L lower case and the second one is i upper case.

I didn't quite follow this logic.  Your example, in my mind, would actually be 
easier to debug with this change.

If you know that variables are case sensitive, you know that you have to check 
for a typo in your variable name if you are not getting what you were 
expecting. Here, in my email client, the l and i are very distinct as well 
as the console I work in.

Just my thoughts on the above concerns presented.

Michael L. Young
(elguero)

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Re: [asterisk-users] Case-sensitivity of Dialplan variables.

[Michael L. Young]

- Original Message -
 From: Ira i...@extrasensory.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, October 2, 2012 8:11:32 PM
 Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
 
 
 Given that many of the users were not programmers and didn't likely
 grow up in a case sensitive world I'd also vote for case
 insensitivity. I fall into that category, I grew up with dBase,
 Clipper and VB and case issues get me all the time when I program in
 C.
 
 Allowing case insensitivity does not stop someone from using case
 consistently and While I guess there could be a reason why you'd want
 to use the word hash in the forms hash, Hash and HASH and have them
 be 3 different items, I'm guessing that the people trying to get
 their feet wet moving from Asterisk-Now to Asterisk would be confused
 to say the least if someone did that in example code.

While true that most users are probably not programmers, most people 
administering Asterisk would be system / network admins, correct?  System 
admins and networking admins are used to working in environments such as Linux 
where variables and file names are case sensitive.

If someone is moving from a GUI interface to CLI, then they would/should know 
that case sensitivity is important and therefore the change shouldn't pose a 
problem.

Just some thoughts in regards to the concerns brought up.

Michael L. Young
(elguero)

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Re: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)

[Michael L. Young]

- Original Message -
 From: Thorsten Göllner t...@ovm-group.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, June 18, 2012 11:52:15 AM
 Subject: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging
 (mysql, odbc)
 

 /etc/odbcinst.ini
 
 [MySQL]
 Description = MySQL ODBC MyODBC Driver
 Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so
 FileUsage = 1

Try adding:

Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so

Adjust the path according to where this file can be found on your system.


 So here are the config file for asterisk.
 
 /etc/asterisk/res_odbc.conf
 -
 [mysql]
 enabled = yes
 dsn = MySQL-asterisk
 username = asterisk
 password = qpalym
 pre-connect = yes

For MySQL, I think you also need:

backslash_is_escape = yes

Give those two things a try and see if that helps.

Regards,
Michael

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Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file

[Michael L. Young]

- Original Message - 

 From: Jayesh Labade jayesh.lab...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Friday, May 25, 2012 2:09:58 AM
 Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44
 bytes file

 Hello Michael,

 Thanks a lot for your immediate help. After applying patch MixMonitor
 started works normally,

 I can understand that this can be Happen in asterisk 10.4 but as a
 stable and Long support version 1.8.12.0 this should not happen. I
 got same error in both version.

 Anyways this patch solved my problem.

Jayesh,

Glad to hear that the information helped figure out what was going on and also 
provided a fix.

In the 1.8 line, this has been fixed as well and will be in future releases.

In an ideal world, there would be no bugs in software.  LTS doesn't mean bug 
free.  It means that it will be supported over a longer period of time which 
should result in more real world use and more bug fixing resulting in a more 
stable product with time.

Regards,
Michael

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Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file

[Michael L. Young]

- Original Message - 

 From: Jayesh Labade jayesh.lab...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thursday, May 24, 2012 4:10:29 PM
 Subject: [asterisk-users] Asterisk MixMonitor starts recording 44
 bytes file

 Hello All,

 I have installaed asterisk 10.4 in my machine. Now suddenly
 MixMonitor application starts generating 44 Bytes of Recording file.
 Is this new tye of Bug? Help me..

 Best Regards,
 Jayesh Labade


Jayesh,

Is this machine x86?  There was a bug that was recently fixed and should show 
up in 10.5.

https://issues.asterisk.org/jira/browse/ASTERISK-19727

Regards,
Michael

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Re: [asterisk-users] All circuits are busy now on outgoing trunk call

[Michael L. Young]

 [0K
 --- SIP read from UDP:192.168.9.251:5060 ---
 SIP/2.0 404 Not Found
 
 Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687
 
 To:
 sip:0722490994@192.168.9.251;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06
 
 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7
 
 CSeq: 102 INVITE
 
 Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250
 
 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO)
 
 Content-Length: 0

I think the 404 Not Found being returned from the server is a clue as to what 
the problem is.

Michael L. Young
(elguero)

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Re: [asterisk-users] What version to upgrade to...?

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 12/11/2011 10:59 PM, Mike Diehl wrote:

 Should I go to 1.8.x?  Or all the way up to 10.x?  This is a
 production system and I can't afford to be testing code.


The 1.8 series is the current LTS release.

Barry


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Re: [asterisk-users] TDM400 FXO stopped working

[Michael L. Young]

- Original Message -
 From: Remco Barendse aster...@barendse.to
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Friday, September 23, 2011 5:27:27 AM
 Subject: [asterisk-users] TDM400 FXO stopped working
 Hi list
 
 I have 2 servers with a TDM400 card, port 1 populated by an FXO (red)
 module), port 4 populated with an FXS module. I am using dahdi
 linux and tools 2.5.0.1. The servers are running CentOS 4 and the
 other
 box CentOS 6.
 
 Both modules have been working fine but recently stopped working, when
 i
 start dahdi with just FXS enabled everything is fine.
 
 This is the error i get :
 Loading DAHDI hardware modules:
 wctdm: [ OK ]
 
 Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: Invalid
 argument
 (22)
 Selected signaling not supported
 Possible causes:
 FXO signaling is being used on a FXO interface (use a FXS
 signaling variant)
 RBS signaling is being used on a E1 CCS span
 Signaling is being assigned to channel 16 of an E1 CAS span
 [FAILED]
 
 
 This is in my system.conf :
 fxoks=1
 echocanceller=mg2,1
 # channel 2, WCTDM/4/1, no module.
 # channel 3, WCTDM/4/2, no module.
 fxsks=4
 echocanceller=mg2,4
 
 # Global data
 
 loadzone = nl
 defaultzone = nl

I think the clue is actually right there in the error message.

You say that port 1 is an FXO module?  Then your signaling is set wrong.  The 
signaling should be fxsks.

For port 4, it should be fxoks.

Remember, that in the configuration files, the signaling option used is 
opposite of what the module is.

Regards,
Michael Young
(elguero)

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Re: [asterisk-users] Polycoms rebooting themselves

[Michael L. Young]

- Original Message -
 From: Mike Diehl mdi...@diehlnet.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, August 30, 2011 5:13:22 PM
 Subject: Re: [asterisk-users] Polycoms rebooting themselves
 
 Well, we've taken the time to check out the wiring.  It's only 3
 years old and
 looks like the people who did it knew what they were doing.  Nice
 work.
 
 Rebooting the cable modem, router, and switch didn't fix the problem.
 
 Also, we had an instance today where ALL of the phones went down
 within
 minutes of each other.  The Internet connection was still active.
 
 Looks like more often than not, all of the phones die at the same
 time.
 
 Any other ideas?
 
 Mike.

How latest(ish) is the firmware?  I see in the release notes for 3.3.2 under 
corrections:

61147: SoundPoint IP 331, 335, 450, 550, 560, 650, 670: SoundStation IP 5000:
Phone reboots when a GET request is sent to the phone to
/TA/getParam?paramName=reg.1.ringType.

68063: Phone reboots when DHCP failover occurs.

(I know you have 335s but someone else mentioned they had issues with 550s)
70988: SoundPoint IP 550: Phone when powered by external AC power reboots
during playing of certain audio on full volume.

Just some things that came to mind when I saw your email.  I had just recently 
reviewed the release notes and they were fresh in my mind.

Hope you find a fix soon.

Regards,

Michael
(elguero)

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Re: [asterisk-users] Custom Dialplan

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 08/05/2011 04:32 AM, Richard Zulu wrote:

 I would like to import my dialplan into freepbx+asterisk since I am 
 switching to that...how can I create my own custom dialplan in
 freepbx?

I'm not sure why you'd want to... freepbx is anathema to custom
dialplans.  That said, I believe you end up naming your
extensions.conf file to extensions_additional.conf and freepbx will
pick it up when it starts.

It's been a long, long time since I've dealt with freepbx -- in fact I
went the other way:  from freepbx+asterisk to pure asterisk.  When I was
using freepbx that was the solution you seek.

Barry

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[asterisk-users] Voicemail not acting as documented.

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

In
http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=markup,
line 180 states:

 Voicemail now runs the externnotify script when pollmailboxes is
 activated and notices a change.

My voicemail.conf configuration for my LDAP vm storage is thus:

externnotify = /opt/asterisk/bin/mwi.pl
pollmailboxes = yes
pollfreq = 30

The script is called whenever I leave a voice mail as well as when I
listen to the voicemail via the voicemail() and voicemailmain()
applications.  When I listen to a voicemail using an email client the
script is not called.  My impression from that line in the CHANGES
document is that it should.

Is there some other parameter required to get this to fire or am I
reading more into that sentence from the CHANGES document than is
actually there?

Thanks.

Barry






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Re: [asterisk-users] Voicemail not acting as documented.

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 07/28/2011 02:42 PM, Barry L. Kline wrote:

 Is there some other parameter required to get this to fire or am I 
 reading more into that sentence from the CHANGES document than is 
 actually there?

Sorry for replying to my own post, but I've done some more
investigating. I glanced through the source for app_voicemail and am
beginning to wonder if there need be a physical SIP device configured to
use that mailbox for the mailbox to be polled.  Is that the case?

This Asterisk installation is acting as a VM server for a legacy phone
system and none of the VMboxes are actually connected to a SIP phone on
this box.  Can this be the source of my problem?

Barry
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Re: [asterisk-users] Voicemail not acting as documented.

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I found what I believe to be a bug and have submitted it:

https://issues.asterisk.org/jira/browse/ASTERISK-18207

Please correct me if I'm wrong.

Barry
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Re: [asterisk-users] FXO ports locking up

[Shawn L]

Doesn't seem to help.  I did it early yesterday morning and have
another 'stuck' call this morning

Does anyone have any other ideas on what I can do to correct this?

thanks

Shawn





CLI core show channels
Channel  Location State   Application(Data)
DAHDI/8-1(None)   Up  AppDial((Outgoing Line))
SIP/cordless8-04 725@out-phone8:1 Up  Dial(DAHDI/8/725)
2 active channels
1 active call


CLI core show channel DAHDI/8-1
 -- General --
   Name: DAHDI/8-1
   Type: DAHDI
   UniqueID: 1310421996.2359
  Caller ID: 725
 Caller ID Name: (N/A)
DNID Digits: (N/A)
   Language: en
  State: Up (6)
  Rings: 0
  NativeFormats: 0x4 (ulaw)
WriteFormat: 0x4 (ulaw)
 ReadFormat: 0x4 (ulaw)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 23
  Frames in: 2489590
 Frames out: 72966
 Time to Hangup: 0
   Elapsed Time: 13h49m51s
  Direct Bridge: SIP/cordless8-049c
Indirect Bridge: SIP/cordless8-049c
 --   PBX   --
Context: in-phone8
  Extension:
   Priority: 1
 Call Group: 0
   Pickup Group: 0
Application: AppDial
   Data: (Outgoing Line)
Blocking in: ast_waitfor_nandfds
  Variables:
BRIDGEPVTCALLID=2e52745c-7bdfef53@192.168.0.134
BRIDGEPEER=SIP/cordless8-049c
DIALEDPEERNUMBER=8/725
TRANSFERCAPABILITY=SPEECH

On Fri, Jul 8, 2011 at 7:25 PM, Alec Davis siva...@paradise.net.nz wrote:
 Is there a way to detect that there is no longer really an
 active call happening and force a hangup or reset the
 channel?  It'd be great if this could happen automatically.
 Or as a temporary fix , is there a way to setup and extension
 that the SIP phone could dial which would clear any active
 calls associated with it?  Right now if this happens, I need
 to login to the Asterisk CLI and issue a hangup command.  If
 I don't, the channel appears to be in-use forever.

 This may be the answer

 sip.conf:

 ;--- RTP timers
 
 ; These timers are currently used for both audio and video streams. The RTP
 timeouts
 ; are only applied to the audio channel.
 ; The settings are settable in the global section as well as per device
 ;
 rtptimeout=60                   ; Terminate call if 60 seconds of no RTP or
 RTCP activity
                                ; on the audio channel
                                ; when we're not on hold. This is to be able
 to hangup
                                ; a call in the case of a phone disappearing
 from the net,
                                ; like a powerloss or grandma tripping over
 a cable.

 Alec Davis


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Re: [asterisk-users] Anybody doing PRI over IP?

[Shawn L]

 Right, this is how I expected it to operate. My prior question though was 
 regarding the 'T1 over Ethernet' scheme someone mentioned which ran full 
 throughput all the time.


That is true.  If you're doing a clear-channel or pseudo-wire T1 over
ethernet you will always be using 1.54 Mbps weather the T1 has any
data on it or not.

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[asterisk-users] FXO ports locking up

[Shawn L]

I have a situation where I have an Asterisk box which receives 8
analog lines from a
Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a
call coming in
on port 1 of the digium FXO board is delivered to SIP phone 1, an
outgoing call on SIP
phone 2 goes out FXO line 2, etc.

This works fine normally, but every once in a while (no set time, or
pattern that I can
see -- It may be caused by the wifi sip phone going out of range of an
access point and
not coming back into range fast enough) the FXO port does not hangup
after the call is
terminated and just sits in an in-use state.  Since it's a 1-to-1
mapping, the SIP phone
associated with the in-use line now produces a fast busy when you
attempt to make a
call because it cannot get an outbound line.

Is there a way to detect that there is no longer really an active call
happening and force a
hangup or reset the channel?  It'd be great if this could happen
automatically.  Or as a
temporary fix , is there a way to setup and extension that the SIP
phone could dial which
would clear any active calls associated with it?  Right now if this
happens, I need to login
to the Asterisk CLI and issue a hangup command.  If I don't, the
channel appears to be
in-use forever.

Thanks

Shawn


The setup is fairly straight-forward

Extensions
[in-phone2]
exten = s,1,Answer()
exten = s,n,Noop(CALLERID(name))
exten = s,n,Noop(CALLERID(num))
exten = s,n,Dial(SIP/cordless2,25,tTo)
exten = s,n,Hangup

[out-phone2]
exten = _[*#0-9]!,1,Dial(${LINE2}/${EXTEN})
exten = _[*#0-9]!,2,Congestion()
exten = _[*#0-9]!,102,Congestion()

[cordless2]
type=friend
qualify=yes
rtptimeout=1
secret=
call-limit=1
nat=no
host=dynamic
canreinvite=no
context=out-phone2
callerid=cordless2 102

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Re: [asterisk-users] Call to *2*999... : IP-phone configration

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 06/21/2011 08:37 AM, Jonas Kellens wrote:

 At the moment, I don't really know what I'm looking for. So if anyone
 knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I
 can find out myself what settings to look for in other IP-phones.

On a Polycom phone you'd be looking for the 'digitmap' to make the
adjustments.

Barry

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[asterisk-users] Mitel PBX caller id format?

[Shawn L]

I'm setting up an asterisk server to extend several extensions from a mitel pbx.

I'd like to display the caller id that I receive from t he mitel pbx
on the sip phone. The mitel
PBX person has setup the PBX to send be callerid, but I don't see it.

I've set chan_dahdi up with
usecallerid=yes
cidstart=ring
cidsignalling=bell
callerid = asreceived
cid_rxgain = 0.0

Are there specific settings for receiving the callerid from a mitel
pbx, or does something need
to be changed on the PBX to send the callerid in the appropriate format?

Thanks

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Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities

[Michael L. Young]

- Original Message -
 From: Chris Maciejewski ch...@wima.co.uk
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, May 19, 2011 9:39:57 AM
 Subject: [asterisk-users] ConfBridge - Failed to find a bridge technology to  
 satisfy capabilities
 
 Hi,
 
 I am trying to use ConfBridge application, but it throws Failed to
 find a bridge technology to satisfy capabilities 0x4 (ulaw) error.
 Please see console output below.
 
 -- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005,
 1001) in new stack
 [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404
 join_conference_bridge: Trying to find conference bridge '1001'
 [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed
 to find a bridge technology to satisfy capabilities 0x4 (ulaw)
 [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368
 destroy_conference_bridge: Destroying conference bridge '1001'
 [May 19 13:36:05] ERROR[7452]: app_confbridge.c:435
 join_conference_bridge: Conference bridge '1001' could not be
 created.
 

I wonder if this recent commit to the 1.8 branch would help fix this issue at 
least with 1.8.

  Author: twilson
  Date: Thu May 19 18:28:13 2011
  New Revision: 319920

  URL: http://svnview.digium.com/svn/asterisk?view=revrev=319920
  Log:
  Revert part of a change to the bridging API code

  The capabilities used in the bridging API are very different than the
  ones used for formats. When the conversion was made expanding the bit
  width of codecs, the bridging code was accidentally accosted in ways
  that it didn't deserve.

  Modified:
  branches/1.8/include/asterisk/bridging.h
  branches/1.8/include/asterisk/bridging_technology.h
  branches/1.8/main/bridging.c

As far as why svn trunk is stating that it cannot create the bridge, the debug 
message is not very informative as to why it couldn't create the conference 
bridge from what I could see briefly looking at the debug logs you posted in 
another message.

Michael
(elguero)

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Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Shawn L]

Yes, I'm talking about mid-call.

I do have rtptimeout and qualify set, both to 30 seconds, which should be
plenty of time.
I set them both because if a phone moves out of range, and never comes back,
asterisk was keeping the channel open way to long.

On Wed, May 4, 2011 at 7:50 PM, Matt Riddell li...@venturevoip.com wrote:

 On 5/05/11 11:40 AM, Sherwood McGowan wrote:

 ChanIsAvail + dialplan routing to call parking lot


 Problem is, I think he's talking about mid call - so ChanIsAvail will have
 returned success - oh unless you can run it in the h exten?


 --
 Cheers,

 Matt Riddell
 ___

 http://www.venturevoip.com/news.php (Daily Asterisk News)
 http://www.venturevoip.com/exchange.php (Full ITSP Solution)
 http://www.venturevoip.com/cc.php (Call Centre Solutions)

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[asterisk-users] Cordless VoIP Phones and Access Point hand-off?

[Shawn L]

 I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310).   Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).

I've been able to get virtually seamless roaming between access points to
work in the past with data
but have never tried it with voice before.  Is there a way to get asterisk
to keep the call active for a
certain period of time even after the phone becomes un-reachable then
re-attach it when the phone
comes back, or hang it up if it never comes back?

 For instance, keep the call active for 30 seconds after the phone becomes
un-reachable.  if it comes back
in 30 seconds, re-attach the active call.  If not, hang it up.

Barring that, if the cordless phone becomes un-reachable is there a way to
automatically put the active call
on hold, or park it?  That's not the preferred solution, but it would work
great until I figure something else
out.

Thanks in advance
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[asterisk-users] Park a call when sip phone becomes unreachable?

[Shawn L]

I have a situation where we have an asterisk box that is extending several
Mitel PBX extensions to
some cordless SIP phones (Cisco WIP310).   Everything works great, except
when the cordless
phone walks out of range of one access point and into range of another
(cisco 1100 series APs).

I have another post to the list asking about how to speed up the handoff,
and keep the call active while
that's happing.  My question here is can a call be parked or placed on hold
if the SIP phone becomes
unreachable?  That way if a cordless phone user walks out of range and
'drops' the call, they have a
way to get it back

Thanks in advance
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Michael L. Young]

- Original Message -
 From: Olle E. Johansson o...@edvina.net
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, April 27, 2011 3:34:03 PM
 Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

 Friends,

 We have a discussion on asterisk-dev about the maintenance of the 1.4
 branch. According to the release plans, support for 1.4 was
 scheduled to close in April 2011 - basically now. After that, only
 security patches would be committed. This is already a delay from
 the original plan published by Russell Bryant.

 Unfortunately, I think this is way too early. My feeling and
 experience is that 1.8 is not ready for production in the
 environments I work in - large scale installations. Customers are
 not planning migration and all new installs are still 1.4. Tests
 we've been doing with 1.8 has failed within just a short time and so
 badly that customers has not paid me to spend any further time with
 1.8.


Whats the game plan to get 1.8 ready for production?  To me, for which I say 
this with all respect, some are focusing still on 1.4 instead of getting 1.8 to 
the level that some of the members of the community are wanting to see.  1.4 
has been very stable for a while.  To the point that I only pay attention to 
security releases to be honest.  It has been this way for quite a while now.  I 
personally have been focused more on using 1.8 when I can, mainly on 
non-critical servers, yet I will admit that I have enough confidence in it now 
to use on main servers.  Why?  Because I want to get my production servers off 
of 1.4 and 1.6.2 due to new features.  But, even if I didn't need or want the 
new features, the current state of 1.4 is excellent.  If I don't ever make the 
move beyond 1.4, how can I contribute to a better product?  By experimenting 
and not giving up at the first sign of trouble with the latest version, I feel 
that I can help to make 1.8 better which ultimately benefits me and the 
community.  I would like to hear a game plan before we just say, yes, lets keep 
focusing on 1.4 and then we will decide a deadline to stop support.  I am 
afraid that software is programmed by imperfect humans and there will always be 
a bug or two that crops up from time to time.  Do we want to keep waiting until 
we feel it is perfect?

One thing I have noticed, is that the bug fixes and patches being contributed 
for 1.8 and trunk are not being taken care of as quickly as it used to back in 
the early 1.4 days.  My feelings are that it is because there have been too 
many releases to work on.  Going back to focusing on just 1.8 and trunk, would 
go a long way to speeding up bug fixes to 1.8.  Again, just my opinion.

 Last time we went through this process with a LTS release (which we
 did not know then) it took over one year before we had a stable
 product to migrate away from 1.2 and jump on the 1.4 track.
 Hopefully, with the help of community, we can move up to 1.8 late
 this year or early next year. For me 1.8 is the focus, it's the LTS
 release.

 Not having a supported 1.4 version from the Digium-hosted
 repositories will mean that we will have to move to separate
 repositories or branch off from the main track. I already maintain a
 ton of subversion branches with various patches to 1.4 It takes a
 lot of time to manage this version that is a fork from the main 1.4
 branch. I will soon have to start working with subversion branches
 for 1.8 to create a compatible version for my customers to test,
 since most of the patches is not part of 1.8. After a few years of
 doing this, I know the work involved with managing code myself.

 The Digium team wants to go ahead and not support 1.4 any more, I
 want to keep 1.4 open for normal bug fixes. What do you think?

Was this really Digium's decision?  You keep mentioning Digium and implying 
them as the evil one in all of this (perhaps I am just misunderstanding your 
tone in your emails and if I am, I sincerely apologize for this) when I seem to 
recall plenty of discussion around these time lines and it was the community 
who set the deadlines, not Digium.  Digium is just trying to abide by the time 
lines outlined for them by the community.  They have already been nice enough 
to extend the deadline in order to finish working on outstanding bug reports 
and patches.  They have bills to pay too and have really tried to extend an 
olive branch to everyone in the community.  There has been a lot of activity on 
the 1.4 branch lately.  If I am wrong, I will gladly retract my comments.


 Kevin proposed that the community maintains the 1.4 branch without
 support from the Digium team. I don't think that's a good solution,
 but it may be the only solution.  I haven't got the resources to
 manage the 1.4 code myself, so I won't step forward as a maintainer
 if I can't get proper funding. Anyone else out there that has the
 time and resources to manage the code?

Again, 

[asterisk-users] Problems With DAHDI on Ubuntu

[Shawn L]

I have 2 separate Asterisk servers that are both exibiting this problem.  1
has a 4 port
FXO digium card, the other an 8 port.

For some reason when the machine reboots, the dahdi drivers are not properly
loaded.  Then asterisk
ends up starting without dahdi support.  I've tried everything that I can
think of, even to the point
of running dahdi_cfg in the asterisk startup script before asterisk itself
is started, but it doesn't help.

To fix it, all i have to do is login and run
dahdi_cfg
/etc/init.d/asterisk restart

but that's a pain to have to do after every reboot.  I've never had this
problem with asterisk systems in
the past, but now it's happening on the last 2 servers we've setup.  Does
anyone have any ideas?  I
can't understand why explicitly calling  dahdi_cfg or the dahdi_startup
script before starting asterisk
isn't working.

Thanks
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Re: [asterisk-users] Upgrade and recompilation

[Barry L. Kline]

On 02/01/2011 12:34 PM, Harel Cohen wrote:

 As one with theoretical knowledge in programing, but never on Linux, I
 can understand terms and code structure but I don’t know:
 
 1. What shell commands (e.g. ./configure, make, make install etc.)
 should I run to recompile Asterisk (same version)?
 
 2. What shell commands should I run if I want to apply a change to
 source code?
 
 3. Is there a general guide on how to upgrade Asterisk?

Read the README file included with the source.

Barry

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Re: [asterisk-users] PRI D-channel bouncing

[Michael L. Young]

-- 
Michael L. Young 
Administrative Claim Service, Inc. | IT Manager 
600 Main Street, Suite 5, Winchester, MA 01890 
www.acsacc.com 
Phone 781-721-1998 

- Original Message -
 From: Andrew Stewart astew...@notre1.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, August 10, 2010 9:33:45 AM
 Subject: [asterisk-users] PRI D-channel bouncing
 I need some help getting a system running for one of my company's
 plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and
 FreePBX 2.8.0.2.
 
 My D-Channel keeps bouncing. The telecom tech told me he thought that
 I might be using the wrong sync source, and I think I might have been,
 but I changed DAHDI system.conf to span=1,1,0,ESF,B8ZS (from
 span=1,0,0,ESF,B8ZS) and I am still having the same problem.
 (Although, the FreePBX DAHDI page only allows me to select 0 in the
 Sync/Clock Source field. 0 is the only option in the drop down.)


 *
 [r...@gch-asterisknow01 ~]# cat /etc/asterisk/chan_dahdi_groups.conf
 ;;
 ; Do NOT edit this file as it is auto-generated by FreePBX. All
 modifications to ;
 ; this file must be done via the web gui. There are alternative files
 to make ;
 ; custom modifications, details at:
 http://freepbx.org/configuration_files ;
 ;;
 ;
 
 
 ; [span_1]
 signalling=pri_net
 switchtype=national
 pridialplan=national
 prilocaldialplan=national
 group=0
 context=from-pstn
 channel = 1-15

Is the PRI coming from the telephone carrier?  If so, shouldn't the signalling 
be pri_cpe?

Michael L. Young
(elguero)

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Re: [asterisk-users] Follow-me to my answering machine :-(

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Bryan Jacobs wrote:

 I wonder if all the cell providers let you do this?

I presume you mean turn off voice mail.  I don't know, but the first
time I called Verizon to have it done the gal I spoke with said it
couldn't be done.  So I said thanks, called in again, got another rep
and he said no problem.  In less than five minutes I was good to go.

Barry



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Re: [asterisk-users] Follow-me to my answering machine :-(

[Barry L. Kline]

Bryan Jacobs wrote:

 I can't just call the car - the car is my cell phone DID with a
 bluetooth kit.

I did this same thing you're attempting.  I have a desk set at home, a
Polycom in my office and my cell phone all being called at the same
time.  I called Verizon and had them disable voice mail on my cell phone
so that the only voice mail system I use is my Asterisk box.  I no
longer give out my cell phone number but only my home phone number and
allow Asterisk to do all of the heavy lifting.

Oh, and I set the caller*ID outbound to the caller*ID of the inbound
call so I can still see who it is.

Barry

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Re: [asterisk-users] Being attacked by an Amazon EC2 ...

[Erik L]

FWIW, we're seeing similar attacks. The below is what I posted on NANOG 
earlier, which summarizes Amazon's stellar abuse response. I've also received 
an off-list e-mail from someone who was getting hit with 6Gbps of traffic from 
them (and was not able to reach anyone there either).

Time to start blocking them at the edge. Let their customers complain to them 
instead.

-Original Message-
From: Erik L 
Sent: April 11, 2010 10:38
To: na...@nanog.org
Subject: Seeking Amazon EC2 abuse contact

Could someone from Amazon EC2 please contact me off-list regarding an abuse 
issue from one of their IPs? Alternatively, could someone please send me the 
contact details of someone there?

E-mailing the abuse e-mail listed in WHOIS per their instructions, including 
all pertinent data, results in an auto-reply indicating to use a form on their 
site. Submitting the form results in There has been an error while submitting 
your data. Please try again later. Calling their supposed NOC (as per WHOIS) 
results in You have reached the legal department at Amazon...please leave a 
message.

Thanks

-- 
Erik
Caneris Inc.
Tel: 647-723-6365
Fax: 647-723-5365
Toll-free: 1-888-444-8843
www.caneris.com

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Re: [asterisk-users] Dropped Calls

[Michael L. Young]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of JR Richardson
 Sent: Tuesday, March 30, 2010 6:55 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Dropped Calls
 
  I've written about this issue several times, but have not yet found any
  solution to it.  I am using asterisk 1.4.21.2 and zaptel 1.4.12.  Phones
  are primarily Snom 300's but I also have a couple of headset phones
  connected to Grandstream HT286 SIP adapters.  I have 8 offices, each has
  it's own asterisk server all running the same versions of asterisk and
  Zaptel.  Only difference is that one office uses a Digium TDM 8-port
  card and the other branches use 4-port Rhino cards with only 2 ports in
  use.  What happens is that periodically we will be in a call and the
  call will just drop.  It's usually within the first couple of minutes of
  the call.  The calls can be either incoming or outgoing.  The phenomenon
  affects both the Snoms and the Grandstreams.  Along with the dropped
  call issue, we periodically have a problem where a person we call or a
  person that calls in cannot hear the person in the our office, but the
  person in our office can hear the remote person fine.
 
  All of the phones are on the same physical network as the asterisk
  server.  There is no NAT, no Firewall, VLAN, etc. between the phones and
  the server.   I have tried running sip debugs on the calls, but on the
  off chance that my logs catch either a drop or a one-way audio, the sip
  debug looks like just a normal call.
 
  Is there any setting that might cause both one-way audio and dropped
 calls?
 
  Thanks,
  Brent Davidson
 
 Join the club.  I've experienced the same with various strains on
 1.4.x above 1.4.21.1 (not an issue with this one that I have seen).
 This issue is truly random and debugging reveals nothing.  I run an
 all SIP environment with same results.  My solution was to downgrade
 to another version or switch to 1.2 or 1.6 depending on what features
 I need for the system.
 
 Sorry I couldn't be of any help, but I feel your frustration.
 
 JR
 --
 JR Richardson
 Engineering for the Masses
 

Is there a chance that you are using Realtime at all?  

I am just curious because I was having problems with dropped calls as well
and just discovered that it appears to be related to the database server.
If for some reason on the database server there is a table lock (which I am
investigating why) asterisk drops any PRI calls and SIP calls.  Everything
looked normal and the error messages never once suggest a problem with the
database server or Realtime.  I was looking everywhere else but at the
Realtime until I stumbled across it.  While doing some backups with FLUSH
READ LOCKS to a slave machine, which I changed asterisk to use a few months
back, I had dropped calls occur.  I later confirmed that asterisk seems to
hang / freeze during that period but once the database server releases the
locks, asterisk continues to function without any problems.  

This started to occur when we had an increase in call volume and an increase
in load on the db server.  I was using Realtime for extensions, sip peers
and CDR.  I had turned off using realtime for CDR (which we don't really use
anyway) and started to use a slave server instead of the master when
performing some maintenance on the master db server.  I left it that way
since I was just using it for extensions and sip peers and that had cleared
it up over the last few months until I ran my backup.

Not sure that helps but it is worth a shot in mentioning to you.

Regards,
Michael Young
(elguero)


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Re: [asterisk-users] Libtonezone

[Joseph L. Casale]

You could read the source code, but based on it's name I would say it is a 
library responsible for zone specific tone generation. Many parts of the world 
have different tone patterns than the U.S. and Asterisk is used worldwide. A 
better question is, why are you concerned by it?

I was building rpm's for dahdi w/ oslec using Anthony Messina's spec file
and he pulls in the shared object as a dep, but looking at digiums repo, it
isn't pulled in as a dep by any of the dahdi rpms?

Thanks!
jlc

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[asterisk-users] Libtonezone

[Joseph L. Casale]

Trying to find out what the libtonezone shared object built with dahdi-tools is
for, the default dahdi package installation from the Digium repo's pull it in,
so when is it needed?

Thanks,
jlc

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Re: [asterisk-users] Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

DHAVAL INDRODIYA wrote:

 where as I am using Asterisk 1.6.0.5 and my machine is using
 *safe_asterisk* script asterisk running

Why are you using such an old version in the 1.6.0 branch?

1.6.0.25 is current, upgrade to there and then worry about the problem
if it recurs.

Barry


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Re: [asterisk-users] Getting verbose or debug tracing in Asterisk

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Tim Culhane wrote:
 Here is my output of 'sip show peers'
 
 user1/user110.41.3.12   D   N  10434Unmonitored 
 user2/user210.41.3.12   D   N  65293Unmonitored 
 user3/user3(Unspecified)D   N  5060 Unmonitored 
 user4/user4(Unspecified)D   N  5060 Unmonitored 
 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0
 offline] 
 
 
 So,  does this mean  the registration worked?
 
 What is the difference between monitored and unmonitored?
 
 Tim


user1  user2 have registered.
user3  user4 have not

Unmonitored means that you have not specified  qualify=yes in the peer
 configuration.

Barry


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Re: [asterisk-users] SIP tunnel

[Scott L. Lykens]

 My idea is to use a well know port like port 80 (that is not blocked).
Skype for example uses this port.

If you are in a situation where the ISP/government is blocking VoIP you
are probably going to have to encrypt it to get it through, and that may
not even work. I have a client who has facilities in Belize where BTL
apparently employs quite sophisticated deep packet inspection... SIP or
IAX on any port combination would drop about half a second after the
media starts. IPSec over UDP/IKE were completely blocked as well. I
ended up using IPSEC over TCP as it was not interfered with.

If the ISP or government are not the problem, only firewalls... IIRC in
a typical NAT setup you could have the client register to you using IAX
- This will keep the port open through the NAT device so you can send
calls to them without them having to map ports in their firewall.

sl

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Re: [asterisk-users] E71

[Scott L. Lykens]

Not sure it is relevant, however, I have an E52 I use occasionally with
my * and I've found that without an active SIM in the phone the SIP
profile will ring silently.

 

I'm sure there's a way to fix it I just haven't been bothered enough to
work on it.

 

sl

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of YC Nyon
Sent: Monday, February 08, 2010 7:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] E71

 

hi,

 

I'm been successful in making calls to another local extension using
Nokia E71. However calling the E71 from another ext. (X-lite) is not
successful. There is a ringing tone from the caller side but the E71 is
silent. 

Tried disabling the NAT (dunno whether that helps).

Instructions where from
http://www.geek.com/articles/mobile/feature-voip-with-nokia-e71-how-to-2
008095/

 

Am i missing something? 

 

 

 

 

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Re: [asterisk-users] 911, location

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

mir shahnawaz wrote:
 Hi there,
 
 I am running a PBX under asterisk 1.6. I have few FXO analogue lines
 connecting to PSTN. These lines are in a hunt group. I trying to make
 my extensions to dial 91, but this is a bit scary, I mean if somebody
 make an emergency call after hours and without completing call is not
 able to tell his/her location. How can I make 911 call center to know
 the exact location of my extension. I think its possible by having
 DID's but I am looking for other options too. I would appreciate your
 valuable ideas and suggestions.

If you're using POTS lines to make the call to 911 they'll know the
location, if the POTS lines come into the building that you're calling
from.  Are you saying that these lines are located in a different location?

Barry


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Re: [asterisk-users] AGI perl script set timeout within script?

[Scott L. Lykens]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Steve Edwards
 Sent: Thursday, January 07, 2010 10:30 PM

 What about:
 
 1) Fixing the slow responding DNS server?
 
 2) Tweaking /etc/resolv.conf options?
 
 3) Setting up a caching name server on your Asterisk host?

I'm out of my element on the rest of the thread but have had great
success in solving various DNS woes with Asterisk (and several other
apps) by simply installing and configuring dnsmasq as a local resolver
talking to my primary DNS servers. If the box is up it can talk to a DNS
server (itself) and get a response (nxdomain at worst) to allow the app
to move on instead of waiting around...

sl

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Re: [asterisk-users] Dahdi and oslec

[Joseph L. Casale]

I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup 
configured.

Anthony,
I appreciate the pointer, and I do have a build environment but am not 100%
sure how to accomplish this under CentOS with your files. Can you elaborate
a bit to get me started?

Thank you very much!
jlc


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Re: [asterisk-users] Dahdi and oslec

[Joseph L. Casale]

I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but 
you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm
tool to build from an svn checkout if you already have a build setup 
configured.

Anthony,
So this script builds them with the dahdi-tools-libs package requirement, I
thought the fedora spec built all of these? Any idea?

Thanks!
jlc

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Re: [asterisk-users] Really Silly Question From Total Newbie

[Barry L. Kline]

UIT DEVELOPMENT wrote:

 Sorry for what might seem as really silly questions, but I am not sure
 how to proceed.
 
 Thanks in advance for any insight that you folks can provide!

Hello Mike.

Welcome to the wonderful world of Asterisk.  Before you sludge through a
GUI and all the attendant bad habits that can produce, I suggest that
you download what we consider to be the Bible of Asterisk.  The infobot
on IRC says:

thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN
0-596-51048-9) --- Order yours at
http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF
http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at
http://astbook.asteriskdocs.org;

Download that and read the first few chapters.  It will make your
Asterisk experience a lot more enjoyable and will help you understand
what you're doing.

This list, and the IRC channel #asterisk, are good resources when you
finally get to the point where you're stuck and need some help.

Regards,

Barry

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Re: [asterisk-users] Dahdi and oslec

[Joseph L. Casale]

  git clone http://git.tzafrir.org.il/git/dahdi-extra.git
  cd dahdi-extra
  make gen-patch

And use the generated dahdi_linux_extra.diff . It includes OSLEC and
some other things. See the Makefile there for more information. The
patch should be applied with -p1 .

This repository includes the extra DAHDI drivers currently included
directly in the Debian package.

Tzafrir,
Thank you very much for this. It's been ages since I had to do this,
and previously I was downloading a recent kernel source and copying
drivers/staging/echo to the dahdi source, then modifying the dahdi
kbuild and adding an echo kbuild. This really isn't an area I am all
that familiar with, but should I assume this patch includes the source
for that recent kernel echo code, and as a result I could apply this to
Jason Parkers srpm for dahdi-linux-2.2.0.2, then rebuild the whole
set to leverage the kmod under CentOS?

Thanks again!
jlc

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Re: [asterisk-users] Dahdi and oslec

[Joseph L. Casale]

Basically - yes. It's an extra patch to add to your source RPM. Are you
familiar with modifying them?

Tzafrir,
Vaguely, I would very graciously take any suggestions you could provide:)
The whole dahdi package routine has change since the last time I used it,
was shortly Jason Parker started providing the dahdi linux/tools.

From what I can tell so far, I can continue to use his user tools unchanged
but I need to apply this patch to the tar file in the 
dahdi-linux-2.2.0.2-1_centos5.src.rpm
and rebuild it, but that , `dahdi-linux` pulls in:

dahdi-firmware
dahdi-firmware-oct6114-064
dahdi-firmware-oct6114-128
dahdi-firmware-tc400m
kmod-dahdi-linux
kmod-dahdi-linux-fwload-
yum-kmod

That of which contain dahdi-firmware and kmod-dahdi-linux-fwload-vpmadt032 
which don't have
srpms available to me.

I'm just unclear on how the patching of the dahdi-linux rpm affects the rest.

Thanks for any guidance!
jlc

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Re: [asterisk-users] Dahdi and oslec

[Joseph L. Casale]

atrpms.net also provides packages for RHEL5, if those would work.

http://atrpms.net/dist/el5/

Just on my way to work on this server now, this would be great! That
way I don't have to work all night:) Does the atrpms ones finally do oslec?

Thanks!
jlc

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Re: [asterisk-users] Dahdi and oslec

[Joseph L. Casale]

I don't use them myself, but I was thinking that the RHEL5 spec files might be 
another place to look for what you need to build with OSLEC included, more 
specifically for CentOS.  I just tried taking a look at ATrpms, but the site 
is having some connection issues at the moment.

How about this -- another CentOS repo:
http://www.zultron.com/2009/03/dahdi-rpms/

This TDM410p card is making my life miserable, it works like crap and kernel
panics several different systems. At this point, I am just going to get a
Linksys SPA3102 and be done with this nightmare...

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[asterisk-users] Dahdi and oslec

[Joseph L. Casale]

Looking at the source in the rpms from the asterisk package site
appears that oslec is not built and enabled for the kmod rpms.

Anyone know an existing repo or have direction on how to enable
this to built for those rpms?

Thanks,
jlc

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Re: [asterisk-users] MYSQL queries from dial plan

[Scott L. Lykens]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Neeraj Chand
 Sent: Monday, January 04, 2010 1:17 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] MYSQL queries from dial plan

[mysql dialplan function]

 This currently takes about 4 seconds to complete.
 
 If I run two simultaneous queries, this goes up to about 9 seconds for
 both queries to complete.
 
 Is there a way that I can bring this time down?

How long does the query take when executed at the MySQL command line?

In my experience there is no perceptible Asterisk-related delay in
executing MySQL from the dialplan versus ODBC versus the MySQL command
line. Is DNS involved or do you access MySQL by IP?

A long time ago we had a poorly written LCR routine that ran for about 3
seconds on a large set of tables. A little bit of intelligence and
indexing has brought that down to 0.3 seconds on old single-core
hardware.

sl

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[asterisk-users] Dahdi causes panic on server restart

[Joseph L. Casale]

Not sure how to go about troubleshooting this, did a fresh install
of CentOS 5.4x86 with a netinstall iso off the base and update repo
followed by a install of dahdi-lniux/tools from the digium/asterisk
repo, ran genconf on my single fxo tdm410p and rebooted, ran fxotune,
rebooted and now this panic every time it restarts...

Any known issues with the version of Dahdi in the repo's? Server is an
HP DL380G4

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[asterisk-users] Dahdi install issues

[Joseph L. Casale]

After using the CentOS repo's at digium to install dahdi Linux  tools, I got 
this:

  Installing : kmod-dahdi-linux-fwload-vpmadt032
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs 
unknown symbol voicebus_transmit
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs 
unknown symbol vpmadtreg_unregister
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs 
unknown symbol voicebus_get_handlers
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs 
unknown symbol voicebus_set_handlers
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs 
unknown symbol vpmadtreg_register
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs 
unknown symbol voicebus_get_pci_dev

Anything to worry about?
Thanks,
jlc

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Re: [asterisk-users] Mixing commercial/SVN Asterisk

[Scott L. Lykens]

 Subject: Re: [asterisk-users] Mixing commercial/SVN Asterisk
 
 On Dec 16, 2009, at 10:08 AM, Richard Kenner wrote:
 
  Am I correct that if I'm running an -rc or from an SVN release tree
  that there's no way I can use any commercial add-ons from Digium,
such
  as Skype, Cepstral, or G.729?
 
 No, happily not correct.  :-)
 
 Digium tries to make their add-ons work with all major releases of
Asterisk.
 You should be able to use all the named proprietary add-ons with 1.4
or 1.6.x
 versions of Asterisk.  SVN TRUNK releases of course are not guaranteed
to

[snip]

I don't mean to be rude in calling you out about it, however, I've been
waiting for three months for an appearance of Fax for Asterisk that is
compatible with 1.6.1.5+. Several previous requests for timelines to
this list have resulted in responses indicating it was being worked on
that week each time with no further information or public release made.

It is frustrating to me as we are encouraged to upgrade due to security
issues but if we want to use this particular Digium product we cannot. I
have chosen to upgrade as we have not purchased Fax for Asterisk and as
we are unable to evaluate it I doubt we will. (Not to be snarky but I
don't think I'm interested in buying a product that has been
incompatible with its intended companion's official release for nearly
four months now with no public progress on making it compatible,
especially when those releases are suggested to resolve security
issues.)

sl

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Re: [asterisk-users] Can't get G.729 to work...

[Barry L. Kline]

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Ben Schorr wrote:

 I’ve got G.729 loaded in the modules on the Asterisk server and on the
 Polycom phones I’ve set G.729 to be the first preference of codec, but
 still when I go SIP SHOW CHANNELS during active calls it still shows
 “(ULAW)” (G.711) as the codec in use.

How about in sip.conf?

Barry
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Re: [asterisk-users] Asterisk Queue Dialplan

[Barry L. Kline]

Daniel Stefanus wrote:
 Hi,
 I want to reconfigure my asterisk dialplan.I have a problem.I have 4
 agents in a queue.How is the configuration for the asterisk dialplan if
 I want to have only 4 agents maximum who can receive the phone,so if the
 fifth caller try to entering the queue they will be noted by my IVR that
 all our agents are busy?Thank you so much for this millis,it really
 helpful especially for a newbie like me.
 
 Best Regards,
 Daniel
 

What do you want to have happen?

Normally you put the caller into the queue and when one of your agents
become available the caller will be sent to him.

If you don't want to put the fifth caller into the queue then I'd
suggest looking at the GROUP* functions to keep count of the number of
callers.

Barry

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Re: [asterisk-users] IVR Prompt Recording

[Barry L. Kline]

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David Gibbons wrote:
 This may belong on -biz, but does anyone have experience with a decent and 
 cheap IVR/prompt recording house?
 
 Are decent and cheap mutually exclusive?
 
 A nice *sounding* lady would be nice... you can keep any burly voice studios 
 to yourself :)
 

I use Allison.  www.theivrvoice.com

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