Re: [asterisk-users] Xorcom PRI

[Steve Totaro]

Turn on PRI debugging and double check your cable.

On Mon, Nov 12, 2018 at 3:24 PM Jeff LaCoursiere 
wrote:

>
> I've been struggling for a few weeks now with the local telco trying to
> bring up a trunk that has been down for a year (hurricanes in the
> caribbean).  Box is a Dell R710, 16G RAM, Ubuntu 14.04.5 LTS, Dahdi
> 2.10.2-rc1, asterisk 13.23.1.  Xorcom Astribank w/ one T1/E1/PRI module,
> plugged into a USB 2.0 port on the Dell.  All of this was working *before*
> the storms last year with the same hardware/versions.
>
> Dahdi sees the astribank and loads firmware without issue:
>
> root@astbeach:~# dmesg | grep -i dahdi
> [661368.877090] dahdi: Version: 2.10.2-rc1
> [661368.880450] dahdi: Telephony Interface Registered on major 196
> [661368.963988] dahdi_transcode: Loaded.
> [661368.982746] INFO-xpp: FEATURE: with sync_tick() from DAHDI
> [661369.233471] INFO-xpd_pri: FEATURE: WITHOUT DAHDI_AUDIO_NOTIFY
> [661370.256053] dahdi_devices astribanks:xbus-00: local span 1 is already
> assigned span 1
> [661370.270028] dahdi_echocan_mg2: Registered echo canceler 'MG2'
> root@astbeach:~# lsusb
> Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
> Bus 006 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
> Bus 005 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
> Bus 001 Device 003: ID 0424:2514 Standard Microsystems Corp. USB 2.0 Hub
> Bus 001 Device 002: ID e4e4:1162 Xorcom Ltd. Astribank 2 series
> Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
> Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
> Bus 003 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
>
> The dahdi drivers are loaded, and the T1 layer has no alarms... telco also
> reports the line itself is "UP":
>
> root@astbeach:~# service dahdi status
> ### Span  1: XBUS-00/XPD-00 "Xorcom XPD [usb:X1067719].1: T1" (MASTER)
> ESF/B8ZS ClockSource
>   1 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   2 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   3 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   4 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   5 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   6 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   7 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   8 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>   9 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  10 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  11 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  12 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  13 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  14 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  15 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  16 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  17 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  18 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  19 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  20 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  21 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  22 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  23 T1 Clear   (In use) (EC: MG2 - INACTIVE)
>  24 T1 Hardware-assisted HDLC  (In use)
>
> asterisk chan_dahdi shows the T1 up with no alarms:
>
> astbeach*CLI> dahdi show status
> Description  Alarms  IRQbpviol CRCFra
> Codi Options  LBO
> Xorcom XPD [usb:X1067719].1: T1  OK  0  0  0  ESF
> B8ZS  0 db (CSU)/0-133 feet (DSX-1)
>
> but the PRI is down:
>
> astbeach*CLI> pri show spans
> PRI span 1/0: Down, Active
>
> I'm not really sure where to take it from here, and the telco has even
> less of a clue.  They brought out some gear that they hooked up to our
> cabling for the T1 and pretty quickly established a PRI, then placed and
> received test calls over it.  At that point they washed their hands of it,
> and logged as a "CPE issue"!
>
> Could it be that the storms damaged the Xorcom unit in such a way that the
> T1 can be up without alarms but the PRI signaling is broken?  Seems
> unlikely.
>
> I have included a few relevant config files below.  Note that the cabling
> wasn't in place when we ran dahdi_genconf, which is why it shows red
> alarm.  There is no red alarm now.
>
> /etc/dahdi/system.conf:
> # Autogenerated by /usr/sbin/dahdi_genconf on Fri Oct 12 11:34:27 2018
> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
> # your manual changes will be LOST.
> # Dahdi Configuration File
> #
> # This file is parsed by the Dahdi Configurator, dahdi_cfg
> #
> # Span 1: XBUS-00/XPD-00 "Xorcom XPD [usb:X1067719].1: T1" (MASTER) RED
> span=1,1,0,esf,b8zs
> # termtype: te
> bchan=1-23
> #dchan=24
> echocanceller=mg2,1-23
> hardhdlc=24
>
> # Global data
>
> loadzone= us
> defaultzone= us
>
> --

Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

[Steve Totaro]

Possibly the realm?

Thanks,
Steve

On Sat, Sep 2, 2017 at 3:58 AM, O. Hartmann  wrote:

>
> It might sound stupid and a kind of "noobish", but I have serious trouble
> with
> registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT
> box.
>
> The following is seen in the log and anything seems somehow "normal", my
> PBX tries to
> REGISTER, receives 401, and then  nothing more!
>
> I can't see why the REGISTER attempt dies that early (reason?). The only
> hint is:
>
> SIP/2.0 401 Unauthorized 1103003032F
>
> Can someone shed some light/help onto this?
>
> Thanks in advance,
>
> Oliver
>
> [...]
> [Sep 1 17:32:06] VERBOSE[100189] res_pjsip_logger.c: <--- Transmitting SIP
> request (829
> bytes) to UDP:213.20.127.47:5060 ---> REGISTER sip:sip.alice-voip.de
> SIP/2.0
> Via: SIP/2.0/UDP
> XXX.XXX.XXX.XXX:5060;rport;branch=
> From:
> ;tag=
> To:
>  Call-ID:
> yxyxyxyxyxyxyxyxyxyxyxyxyxyxy
> CSeq: 15095 REGISTER
> Contact: 
> Expires: 1800
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
> UPDATE, PRACK,
> REGISTER, REFER, MESSAGE Max-Forwards: 70
> User-Agent: Asterisk13
> Authorization: Digest username="491234567890", realm="ims.telefonica.de",
> nonce="xxx", uri="sip:sip.alice-voip.de",
> response="186x11yd22424424EDQb11133315b44ff1", algorithm=MD5,
> cnonce="BasjdasKFHKbfhhfkjhfjkSGHF", qop=auth, nc=0001
> Content-Length: 0
>
>
> [Sep 1 17:32:06] VERBOSE[100188] res_pjsip_logger.c: <--- Received SIP
> response (589
> bytes) from UDP:213.20.127.47:5060 ---> SIP/2.0 401 Unauthorized
> 1103003032F
> Via: SIP/2.0/UDP
> XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=
> xxx
> To: ;tag=
> xxx
> From: ;tag=y
> Call-ID:
> yxyxyxyxyxyxyxyxyxyxyxyxyxyxy CSeq: 15095 REGISTER
> Service-Route: 
> WWW-Authenticate: Digest
> realm="ims.telefonica.de",nonce="xx
> ",algorithm=MD5,qop="auth"
> Content-Length: 0
>
> [Sep 1 17:32:06] WARNING[100189] res_pjsip_outbound_registration.c:
> Temporal response
> '401' received from 'sip:sip.alice-voip.de' on registration attempt to
> 'sip:491234567...@sip.alice-voip.de', retrying in '30'
>
> [...]
>
>
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Re: [asterisk-users] Moving call DAHDI from channel X to Y.

[Steve Totaro]

I remember seeing something like this a long time ago.  If memory serves me
correctly it was a problem at the physical layer and a couple of the PRI
cables got flipped and plugged into the wrong port.  I had to change the
configs since I didn't have physical access to the box.

Thanks,
Steve

On Sun, Jul 30, 2017 at 9:34 PM, Daniel Harper  wrote:

> I am seeing the in the asterisk logs that channels (PRI ISDN)  are
> being moved ..
>
> [Jul 29 16:31:48] VERBOSE[16125] logger.c: -- Moving call
> (DAHDI/57-1) from channel 57 to 58.
>
> I then see the moved channels with a "0:" in front of it.
>
> [Jul 29 16:31:48] VERBOSE[26691] logger.c: -- Hungup 'DAHDI/0:58-1'
>
> Any ideas why this could be happening?
>
> I believe these messages are coming from chan_dahdi.c and the
> "pri_fixup_principle" function.
>
> --
> Cheers,
>
> Daniel
>
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>
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Re: [asterisk-users] Curious problem with NAT

[Steve Totaro]

On Sun, Jun 7, 2015 at 8:46 AM, Steve Totaro stot...@totarotechnologies.com
 wrote:



 On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello lucab...@lucabert.de
 wrote:

 Ashwin Surendran ashwin.surend...@now-health.com schrieb:

  What settings have you got for directmedia?
 
  Could you try
 
  nat=force_rport,comedia
  directmedia=no

 Tried. Peer always unreachable, call not possible... :(

 Other idea?

 Thanks
 Luca Bertoncello
 (lucab...@lucabert.de)



 Are you using the wifi on on the cellphone?  The peer IP is showing as
 192.168.200.3 which is not a routable address.  Unless things have changed,
 double NAT configurations do not work.

 Thanks,
 Steve T


You could try using your carrier's internet access instead of wifi.

OpenVPN for Android looks like it could work to eliminate your NAT issues
as well.

Thanks,
Steve T
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Re: [asterisk-users] Curious problem with NAT

[Steve Totaro]

On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello lucab...@lucabert.de
wrote:

 Ashwin Surendran ashwin.surend...@now-health.com schrieb:

  What settings have you got for directmedia?
 
  Could you try
 
  nat=force_rport,comedia
  directmedia=no

 Tried. Peer always unreachable, call not possible... :(

 Other idea?

 Thanks
 Luca Bertoncello
 (lucab...@lucabert.de)



Are you using the wifi on on the cellphone?  The peer IP is showing as
192.168.200.3 which is not a routable address.  Unless things have changed,
double NAT configurations do not work.

Thanks,
Steve T
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Re: [asterisk-users] Curious problem with NAT

[Steve Totaro]

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello lucab...@lucabert.de
wrote:

 Zitat von Steve Totaro stot...@totarotechnologies.com:

  Are you using the wifi on on the cellphone?  The peer IP is showing as
 192.168.200.3 which is not a routable address.  Unless things have
 changed,
 double NAT configurations do not work.


 Hi Steve,

 My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct
 in Internet.
 But maybe my Provider does a NAT, too...

 Very strange is, that I have a very poorly audio-quality, if I use my
 cellphone in my WLAN and connect to my Asterisk.
 With THE SAME USER, but from a PC in the same Network, the audio quality
 is perfect.

 Any idea?

 Thanks
 Luca Bertoncello
 (lucab...@lucabert.de)




Not without seeing some SIP debug output.

Thanks,
Steve T
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Re: [asterisk-users] Lost audio on forwarded calls

[Steve Totaro]

Asterisk does not need to care.  Is it SIP all the way through?

Thanks,
Steve T

On Fri, Oct 3, 2014 at 3:12 PM, Todd R. tjrl...@live.com wrote:

 OK, been messing with Asterisk for a long time and I have my opinion on
 where the issues lies but sometimes it's just nice to see what others think
 that can relate :-)

 Here goes..

 Inbound calls flow like this:
 Tier 1 Provider (SIP)  Asterisk 1.8  Name Brand PBX - Calls work fine

 Outbound calls flow like this:
 Name Brand PBX  Asterisk 1.8  Tier 1 provider (SIP) - Calls work fine


 Problem is being reported on that many (not all) calls have no audio when
 they are forwarded.

 Example of forwarded call:
 Inbound call comes in from Tier 1 Provider  Asterisk 1.8  Name Brand PBX

 Name Brand PBX then forwards the call back out to users cell phone:
 Name Brand PBX  Asterisk 1.8  Tier 1 provider

 No audio a large percentage of the time.


 It's my opinion that the Asterisk box only sees the forwarded call as a
 regular outbound call and forwards it on to the Tier 1 provider then to the
 users cell phone.

 I don't see how Asterisk even knows or cares if it was forwarded within
 the Name Brand PBX. The Name Brand PBX is the one making the connection of
 the inbound and outbound call. All other inbound and outbound calls are
 fine, audio is only lost when the Name Brand PBX connects the two calls and
 creates the forward.

 Thoughts?

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Re: [asterisk-users] PRI timing settings

[Steve Totaro]

PRI intense debug should show all you need to fix this.


On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net wrote:


 Sadly none of these changes have made any difference.  I'll report the
 resolution for posterity once we find it.

 Thanks,

 j


 On 08/20/2014 10:13 AM, Don Kelly wrote:

  It’s possible that Sprint is burping on the name. Try first dropping the
 “1.”  Then try dropping the name also, if necessary.



   --Don





 *From:* asterisk-users-boun...@lists.digium.com [
 mailto:asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere
 *Sent:* Wednesday, August 20, 2014 10:03 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* Re: [asterisk-users] PRI timing settings




 What about the text portion?  Should that never be sent?  I was indeed
 sending the '1', and I will remove that to see if it solves my problem, but
 I also have the company name in there.  I feel like a newb asking such
 questions, but I've never had this issue before :)

 Company 1NXXNXX

 Cheers,

 j

 On 08/20/2014 09:46 AM, Eric Wieling wrote:

  NXXNXX is the correct format of CallerID numbers in NANPA.   The
 leading 1 is not part of any NANPA phone number.   Toll free “area codes”
 are also not valid for CallerID.



 *From:* asterisk-users-boun...@lists.digium.com [
 mailto:asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff LaCoursiere
 *Sent:* Wednesday, August 20, 2014 2:41 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* Re: [asterisk-users] PRI timing settings



 On 08/20/2014 07:58 AM, Scott L. Lykens wrote:



 On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere 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




 This makes me curious... what *is* the simplest format possible for NANPA
 numbers?  I'm sure there must be a spec to conform to.  Can anyone point me
 to it?

 Cheers,

 j








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Re: [asterisk-users] T1 Card RED ALARM

[Steve Totaro]

Remember to always check your cables first.

Thanks,
Steve T


On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com wrote:


 Thank you Josh for your valuable reply. I will do try changing the server
 and let you know what happening.


 ~Arun


 On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com
 wrote:



 On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com
 wrote:

 Hello All,

 I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect
 T1 lines it goes in RED. When I do connect the same line on a different
 Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1
 Card for any hardware failures. I heard about loopback test , how helpful
 it is?

 Here are my configuration
 /etc/zaptel.conf
 span=1,1,0,esf,b8zs
 bchan=1-23
 dchan=24

 Zaptel Configuration
 ==
 SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 24 channels configured

 Thanks
 ~Arun


 It could still be some sort of system config issue, even if you think
 everything is configured the same.  Have you tried moving the T1 card from
 the Bad system to the good system?  That will at least help narrow down
 if it's a bad card / port, or a config issue.

 -Josh

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Re: [asterisk-users] VPN SIP Phone | PC Traffic

[Steve Totaro]

I did this with SNOM phones and a special firmware a while ago.  The trick
to get the VPN to extend to the PC port is bridge-utils.  Worked very well.
On Apr 9, 2014 7:40 AM, Positively Optimistic 
positivelyoptimis...@gmail.com wrote:

 We are using vpn routers to connect home users back to our office network.
   Basically, shipping a mikrotik router that 'calls home' and establishes a
 vpn connection for the pc and phone that are connected to the mikrotik...
 user plugs router in, plugs phone and computer into router, and that
 traffic is encapsulated back to our office... simple and straighforward.

 We would like to remove the router from the equation...  does anyone know
 of a SIP phone with a built in VPN client that can provide the tunnel for 
 *both
 the phone and the pc traffic*?  It would seem trivial to route a subnet
 down to the vpn client in the phone, that would be available to devices
 connected on the PC side of the telephone..  This would be tremendous for
 an at-home contact center agent..An added benefit would be to limit
 connections the connection on the PC side of the phone to a specific mac
 address..

 We're aware of the opportunity to use a softphone on the pc with a vpn
 client.   though, we're looking for a physical phone.

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Re: [asterisk-users] IAX2 Trunk Encryption

[Steve Totaro]

Wireshark.


On Fri, Apr 4, 2014 at 11:13 AM, Elliott W dig...@private-address.infowrote:

 Ok, I think I am 90%+ there.

 Note: the configuration or status is the same on both sides unless
 otherwise noted.

 I am using RSA keys for authentication and the calls are coming through as
 authenticated so I'm sure that part works.

 The peer shows the (E) next to the status in Asterisk Info for the IAX2
 peers

 The trunk configuration contains:
 encryption=yes

 So here is my question, Calls stop flowing when I use the directive:
 forceencryption=yes
 At the trunk level or higher does not matter, same effect.

 So my question comes down to, are my calls getting encrypted and why does
 this directive cause them to fail, AND how can I tell.

 Thanks.


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Re: [asterisk-users] IAX2 Trunk Encryption

[Steve Totaro]

Have you enabled IAX2 debugging and tried some test calls?

Thanks,
Steve T


On Fri, Apr 4, 2014 at 6:59 PM, Elliott W dig...@private-address.infowrote:

 That answered my question as to whether it WAS encrypted, I think, and the
 answer is no, the credentials are but all the rest is not.  That just
 leaves the question of what I need to do to get it encrypted..

 Thanks.


 On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro 
 stot...@totarotechnologies.com wrote:

 Wireshark.



 On Fri, Apr 4, 2014 at 11:13 AM, Elliott W 
 dig...@private-address.infowrote:

 Ok, I think I am 90%+ there.

 Note: the configuration or status is the same on both sides unless
 otherwise noted.

 I am using RSA keys for authentication and the calls are coming through
 as authenticated so I'm sure that part works.

 The peer shows the (E) next to the status in Asterisk Info for the
 IAX2 peers

 The trunk configuration contains:
 encryption=yes

 So here is my question, Calls stop flowing when I use the directive:
 forceencryption=yes
 At the trunk level or higher does not matter, same effect.

 So my question comes down to, are my calls getting encrypted and why
 does this directive cause them to fail, AND how can I tell.

 Thanks.




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[asterisk-users] (OT) Phones with STP, DHCP, and/or (T)FTP Issues

[Steve Totaro]

I remember having to turn off STP or set portfast on some switch ports to
some phones due to the boot sequence and timeouts of some phones a long
time ago.

Does anyone know which phones, if any still suffer from these problems?

I am setting up a lab and want to introduce this problem for the class.

Thanks,
Steve T
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Re: [asterisk-users] Need more meetme users -- hitting some limit

[Steve Totaro]

I found below here:  http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe

If you have too many conferences, one CPU may not be able to mix all the
audio and you will have audio problems even if there are 7+ other CPUs that
are essentially idle while waiting for one CPU to mix everything. You
should be able to handle 512 conference participants on a modern server
system without problem. The current trunk of *DAHDI linux limits the number
of open pseudo channels to 512 for this reason*. [1]

Thanks,
Steve T

[1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9610

The new ConfBridge module in the upcoming Asterisk 1.10 release may not
have this limitation.


On Fri, Mar 21, 2014 at 11:53 AM, Steve Edwards
asterisk@sedwards.comwrote:

 I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
 CentOS 6.5.

 The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.

 The primary application will be bridging groups of users using meetme().

 I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1
 behaving a bit more like a production box -- bridging calls (box2).

 The call file on box1 originates a call to box2 and then plays a 2 hour
 WAV file.

 The dialplan on box2 drops the call into a meetme, creating the room name
 from the last 2 digits of the current call count -- distributing the calls
 into 100 meetmes.

 When I run a script to create 500 call files on box1, box2 starts
 complaining at 312 calls, logging 'Unable to open DAHDI pseudo channel:
 Cannot allocate memory' on the console.

 From the 'callers perspective' the call is dropped between 'There are
 currently x other participants in the conference' and the 'beep-beep.'

 'top' says Asterisk is only using about 1/2 gigabyte of RAM.

 'top' says Asterisk is using about 250% of the CPU (4 physical, 8 logical
 cores).

 'ulimit' (added to /usr/sbin/safe_asterisk in the run_asterisk() function)
 says the open file limit is 397,006.

 'ls -l /proc/$(cat /var/run/asterisk/asterisk.pid)/fd | wc -l' says
 Asterisk only has 2,194 files open.

 'iftop' sees about 24Mb of bandwidth in each direction between the boxes.

 Using confbridge() I can easily get 3,000 calls (14,869 open files, 180Mb
 bandwidth), but I'd lose some functionality and have to re-write parts of
 my application.

 Any clues of what limit I'm hitting and how to increase it?

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000


 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Need more meetme users -- hitting some limit

[Steve Totaro]

On Fri, Mar 21, 2014 at 2:26 PM, Steve Edwards asterisk@sedwards.comwrote:

 On Fri, 21 Mar 2014, Steve Totaro wrote:

  I found below here:  http://www.voip-info.org/
 wiki/view/Asterisk+cmd+MeetMe

 If you have too many conferences, one CPU may not be able to mix all the
 audio and you will have audio problems even if there are 7+ other CPUs that
 are essentially idle while waiting for one CPU to mix everything. You
 should be able to handle 512 conference participants on a modern server
 system without problem. The current trunk of DAHDI linux limits the number
 of open pseudo channels to 512 for this reason. [1]


 With 312 calls distributed across 100 meetmes, 'top' shows 1 core at 32%,
 1 core at 6% and the rest basically idle.

 So it looks like meetme() is still a single CPU application, but I have
 plenty of CPU headroom.

 Coincidentally, 512 is my target. Any clues on how to get 200 more?


 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000



What does the console say for channels when you max out?  That limitation
has to be in the source code if in fact that is the limit you are bumping
into.

Thanks,
Steve T
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Re: [asterisk-users] Need more meetme users -- hitting some limit

[Steve Totaro]

Is there any good documentation on that process?


On Fri, Mar 21, 2014 at 3:36 PM, John Novack
jnov...@stromberg-carlson.orgwrote:


 Steve Edwards wrote:

 On Fri, 21 Mar 2014, Adrian Serafini wrote:

  Upgrade to 1.4?  hehe, I thought you were the self proclaimed 1.2
 luddite? I'm a big fan of older releases with 1 year plus of uptime.


 Yep, that's me :)

 I'm trying to make the leap from 1.2 to 11.8.1

  That is a HUGE leap
 Watch out for whiplash!

 John Novack

 --

 Dog is my Co-pilot



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Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

[Steve Totaro]

Gateway computers rejects calls like this.  I was informed that their
carrier rejects the calls because they cannot accurately bill.

It seems pretty silly with voip and number portability.

Thanks,
Steve T

On Mar 17, 2014 5:19 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Often it is P-Asserted-ID, but depends on the carrier.  You should be
asking your carrier how to do this.   Be careful, if the carrier doesn't
like your CID spoofing they might bill the call to a default number on the
account.

 I suspect it is the destination which is rejecting the call because toll
free numbers are not considered valid, not your carrier rejecting the call.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic
 Sent: Monday, March 17, 2014 4:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk
11.5.1

 In a multi-tenant environment, we are sending various CallerIDs outbound
from asterisk based on who the user is.  We have an insurance agency who
would like to present a toll free callerid.  This works..  unless they're
calling a toll free number.  In that case, occasionally, the call fails.
 However, should we send a correctly formatted npanxx of a local number,
the call completes.

 We have been advised that we can send the billing telephone number as a
separate header and the call will complete, all-the-while, presenting the
toll free number as the caller id.

 Does anyone know of the correct header required to provide this
functionality?



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Re: [asterisk-users] Remote extensions call drops after 20 seconds.

[Steve Totaro]

Check here:
http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0

Thanks,
Steve Totaro


On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.com wrote:

 Yes, well, really is Elastix.   Hmmm where I need to pt directmedia=no ?

 Thanks,


 On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote:

 Try ulaw instead of g729, set directmedia=no

 I see you are using FreePBX.  I cannot help further.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
 Sent: Monday, March 10, 2014 4:15 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: and...@telesip.net
 Subject: Re: [asterisk-users] Remote extensions call drops after 20
 seconds.

 Guys, hi. I have not solved the problem. Outgoing calls to remote
 extensions drops on 5-20 seconds. Incoming calls work perfectly.

 Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq

 Thanks,


 On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote:


 See sip.conf.sample in the Asterisk tarball for documentation of
 valid settings.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com

 Sent: Wednesday, December 18, 2013 9:30 PM
 To: and...@telesip.net; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: Re: [asterisk-users] Remote extensions call drops after
 20 seconds.


 I set canreinvite=very  in the remote extension, and now the call
 not drops. Valid solution?


 On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net
 wrote:


 On 12/18/13, 3:09 PM, alp...@gmail.com wrote:


 Hello. I have a problem with the configuration of
 a remote extensions. Calls are truncated at 20 seconds.

 I got my my NAT firewall properly configured.
 Here I attached my debug in CLI: http://pastebin.com/gh34E69f


 When the call is setup I see your Asterisk retransmitting
 the SIP/2.0 200 OK packet many times and getting no response.  The other
 end needs to receive the packet and generate an ACK.  You need to trace
 where that packet is going and figure out why it is not reaching its
 target, or if it is, then why is the ACK not making it back.  Thats your
 problem.


 Thank you!

 --

 Allan Porras

 http://allanPorras.com 
 http://www.AllanPorras.com
 Google Plus: http://goo.gl/BRkbX

 Twitter: @alpocr http://twitter/alpocr










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 http://www.cellroute.net

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 Allan Porras

 http://allanPorras.com http://www.AllanPorras.com Google Plus:
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 Twitter: @alpocr http://twitter/alpocr




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 http://allanPorras.com http://www.AllanPorras.com Google Plus:
 http://goo.gl/BRkbX

 Twitter: @alpocr http://twitter/alpocr



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 Twitter: @alpocr http://twitter/alpocr



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Re: [asterisk-users] send the calls from to servers

[Steve Totaro]

POTS?


On Thu, Dec 19, 2013 at 1:31 PM, Salaheddine Elharit 
salah.elharit...@gmail.com wrote:

 i ask about outbound calls not inbound round-robin

 best regards


 2013/12/19 Eric Wieling ewiel...@nyigc.com

 Inbound call hunting is handled by your carrier, not Asterisk.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit
 Sent: Thursday, December 19, 2013 12:52 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] send the calls from to servers


 I have this scenario


 In the first server 192.168.5.100 I have asterisk installed 1.4.43 and
  one diguim card with 2 ports: in the first port connection for the
 provider X : the second port of diguim card  the connection of the provider
 Y


 In the second server (the same configuration) 192.168.5.200 asterisk
 installed 1.4.43 and  one diguim card with 2 ports : the first port is
 empty the second port  the connection of the provider Y


 My question how can I do in order to send the calls of the second
 providers from the port 2 server 1 and port 2 server 2 ()if one of them is
 down I continue to send the calls from the other



  Thanks and regards

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Re: [asterisk-users] Asterisk TON number

[Steve Totaro]

On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani endri.stef...@plus.alwrote:

  Hi

 ** **

 Greeting to all you out there.

 ** **

 I am new at asterisk, I have been working with PLMN platforms
 telecommunication for 5 years with NSN and Huawei.

 We have recently built an asterisk PBX with Trixbox and connected it to
 our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are
 tons of information out there, except for the TON number.

 If you have worked in Telecommunication you will know the importance of
 TON flexibility. 

 All the posts online suggested to update under Chan_dahdi.conf:

 pridialplan = international

 prilocaldialplan = international

 or other TON value ,restart the platform and then trixbox1*CLI dialplan
 reload

 I have already done this with no success. Are there other changes I have
 to make in order to change dialplan?

 ** **

 ** **

 Br

 **


So what are you trying to do specifically?

Thanks,
Steve Totaro
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Re: [asterisk-users] Asterisk TON number

[Steve Totaro]

I asked you before.  What exactly are you trying to do that you cannot?  It
helps to be fairly detailed when asking a question to the list.  Include
error messages if you have any.

The dialplan and your ISDN configs are different things.  It sounds like
maybe you are having issues with with your dailplan and pattern matching.
 You can probably do all of that from the Trixbox GUI.  If you like
Trixbox, check out FreePBX since Trixbox is done.

Thanks,
Steve Totaro


On Wed, Sep 25, 2013 at 9:30 AM, Endri Stefani endri.stef...@plus.alwrote:

  Hi guys

 ** **

 Thanks a lot, I am just getting used to it, my telco managers J don’t
 trust stability for open source solution for voice(you give a headache to
 calling parties if lag is more than 250ms J ) and I want to prove them
 wrong. I have successfully integrated * with our system via ISDN and first
 calls look fast and clear but it is required to be flexible with TON number
 in order to be used. I tried unloading and loading chan-dahdi.conf like
 Hans suggested with no success TON was not changed.

 Here is my chan_dahdi.conf, is there anything else I should do in order
 for new pridailplan came into action :

 ** **

 ;

 ; DAHDI telephony

 ;

 ; Configuration file

 ** **

 [trunkgroups]

 ** **

 [channels]

 ** **

 switchtype = qsig

 context = pri_incoming

 group = 0

 signalling = pri_cpe

 channel = 1-15,17-31

 ** **

 ** **

 ** **

 ;language=en

 ;context=from-zaptel

 ;signalling=fxs_ks

 ;rxwink=300  ;

 ; Whether or not to do distinctive ring detection on FXO lines

 ;

 ;usedistinctiveringdetection=yes

 ** **

 usecallerid=yes

 hidecallerid=no

 callwaiting=yes

 usecallingpres=yes

 callwaitingcallerid=yes

 threewaycalling=yes

 transfer=yes

 cancallforward=yes

 callreturn=yes

 echocancel=yes

 echocancelwhenbridged=no

 echotraining=400

 rxgain=0.0

 txgain=0.0

 group=0

 callgroup=1

 pickupgroup=1

 immediate=no

 ** **

 ** **

 ;faxdetect=both

 faxdetect=incoming

 ;faxdetect=outgoing

 ;faxdetect=no

 ** **

 ;Include setup-pstn configs

 #include dahdi-channels.conf

 ** **

 group=1

 ** **

 ;Include PBXconfig configs

 #include chan_dahdi_additional.conf

 ** **

 ** **

 unknown:Unknown

  private:Private ISDN

  local:  Local ISDN

  national:   National ISDN

  international:  International ISDN

  dynamic:Dynamically selects the appropriate dialplan

  redundant:  Same as dynamic, except that the underlying number is not
 

  changed (not common)

 ** **

 pridialplan = international

 prilocaldialplan = international

 ** **

 ** **

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *$$ dave cantera
 (android asus)
 *Sent:* Wednesday, September 25, 2013 2:46 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion

 *Subject:* Re: [asterisk-users] Asterisk TON number

  ** **

 if you are a serious teleco guy, which it seems you are. you might
 consider dumping trixbox in the near future. while trixbox does provide a
 good entry level into the * world, there are limitations that will
 eventually hold you back from enjoying the full breadth of utility that *
 offers.

 food for thought,

 Dave Cantera
 (856)813-7098 mobile/txt
 david.cant...@ibsonecall.com

 Sent from my ASUS Pad

 Steve Totaro stot...@totarotechnologies.com wrote:

  ** **

 ** **

 On Wed, Sep 25, 2013 at 3:22 AM, Endri Stefani endri.stef...@plus.al
 wrote:

 Hi

  

 Greeting to all you out there.

  

 I am new at asterisk, I have been working with PLMN platforms
 telecommunication for 5 years with NSN and Huawei.

 We have recently built an asterisk PBX with Trixbox and connected it to
 our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are
 tons of information out there, except for the TON number.

 If you have worked in Telecommunication you will know the importance of
 TON flexibility. 

 All the posts online suggested to update under Chan_dahdi.conf:

 pridialplan = international

 prilocaldialplan = international

 or other TON value ,restart the platform and then trixbox1*CLI dialplan
 reload

 I have already done this with no success. Are there other changes I have
 to make in order to change dialplan?

  

  

 Br

  

 ** **

 So what are you trying to do specifically?

 ** **

 Thanks,

 Steve Totaro 

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Re: [asterisk-users] Asterisk TON number

[Steve Totaro]

So you are using QSIG and connecting your Asterisk box to a legacy PBX over
PRI E1?

Did you try unknown?
Do you need to use QSIG (over euroisdn for instance)?

Thanks,
Steve Totaro


On Wed, Sep 25, 2013 at 10:33 AM, Endri Stefani endri.stef...@plus.alwrote:

  Hi Steve,

 ** **

 There are no errors I need to be able to change TON(below my PRI debug )
 in international or subscriber. The change in chan_dahdi.conf did not do it
 

 ** **

  Calling Number (len= 8) [ Ext: 0  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)

Presentation: Presentation permitted, user
 number not screened (0)  '1000' ]

  [70 0c a1 30 30 36 36 39 31 31 30 30 30 30]

  Called Number (len=14) [ Ext: 1  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)  'xx' ]

 ** **

 ** **

 Br

 ** **

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro
 *Sent:* Wednesday, September 25, 2013 4:24 PM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk TON number

  ** **

 I asked you before.  What exactly are you trying to do that you cannot?
  It helps to be fairly detailed when asking a question to the list.
  Include error messages if you have any.

 ** **

 The dialplan and your ISDN configs are different things.  It sounds like
 maybe you are having issues with with your dailplan and pattern matching.
  You can probably do all of that from the Trixbox GUI.  If you like
 Trixbox, check out FreePBX since Trixbox is done.  

 ** **

 Thanks,

 Steve Totaro

 ** **

 On Wed, Sep 25, 2013 at 9:30 AM, Endri Stefani endri.stef...@plus.al
 wrote:

 Hi guys

  

 Thanks a lot, I am just getting used to it, my telco managers J don’t
 trust stability for open source solution for voice(you give a headache to
 calling parties if lag is more than 250ms J ) and I want to prove them
 wrong. I have successfully integrated * with our system via ISDN and first
 calls look fast and clear but it is required to be flexible with TON number
 in order to be used. I tried unloading and loading chan-dahdi.conf like
 Hans suggested with no success TON was not changed.

 Here is my chan_dahdi.conf, is there anything else I should do in order
 for new pridailplan came into action :

  

 ;

 ; DAHDI telephony

 ;

 ; Configuration file

  

 [trunkgroups]

  

 [channels]

  

 switchtype = qsig

 context = pri_incoming

 group = 0

 signalling = pri_cpe

 channel = 1-15,17-31

  

  

  

 ;language=en

 ;context=from-zaptel

 ;signalling=fxs_ks

 ;rxwink=300  ;

 ; Whether or not to do distinctive ring detection on FXO lines

 ;

 ;usedistinctiveringdetection=yes

  

 usecallerid=yes

 hidecallerid=no

 callwaiting=yes

 usecallingpres=yes

 callwaitingcallerid=yes

 threewaycalling=yes

 transfer=yes

 cancallforward=yes

 callreturn=yes

 echocancel=yes

 echocancelwhenbridged=no

 echotraining=400

 rxgain=0.0

 txgain=0.0

 group=0

 callgroup=1

 pickupgroup=1

 immediate=no

  

  

 ;faxdetect=both

 faxdetect=incoming

 ;faxdetect=outgoing

 ;faxdetect=no

  

 ;Include setup-pstn configs

 #include dahdi-channels.conf

  

 group=1

  

 ;Include PBXconfig configs

 #include chan_dahdi_additional.conf

  

  

 unknown:Unknown

  private:Private ISDN

  local:  Local ISDN

  national:   National ISDN

  international:  International ISDN

  dynamic:Dynamically selects the appropriate dialplan

  redundant:  Same as dynamic, except that the underlying number is not
 

  changed (not common)

  

 pridialplan = international

 prilocaldialplan = international

  

  

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *$$ dave cantera
 (android asus)
 *Sent:* Wednesday, September 25, 2013 2:46 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion


 *Subject:* Re: [asterisk-users] Asterisk TON number

  

 if you are a serious teleco guy, which it seems you are. you might
 consider dumping trixbox in the near future. while trixbox does provide a
 good entry level into the * world, there are limitations that will
 eventually hold you back from enjoying the full breadth of utility that *
 offers.


 food for thought,

 Dave Cantera
 (856)813-7098 mobile/txt
 david.cant...@ibsonecall.com

 Sent from my ASUS Pad

 Steve Totaro stot...@totarotechnologies.com wrote:

  

  

 On Wed, Sep 25, 2013 at 3:22

Re: [asterisk-users] Asterisk TON number

[Steve Totaro]

The call is being placed, is it not?  Again, I know you are trying to
change the TON but what are you trying to accomplish and what is failing.
 It seems like you are dialing 1000 and that is being sent on the wire.

Thanks,
Steve Totaro


On Wed, Sep 25, 2013 at 10:37 AM, Steve Totaro 
stot...@totarotechnologies.com wrote:

 So you are using QSIG and connecting your Asterisk box to a legacy PBX
 over PRI E1?

 Did you try unknown?
 Do you need to use QSIG (over euroisdn for instance)?

 Thanks,
 Steve Totaro


 On Wed, Sep 25, 2013 at 10:33 AM, Endri Stefani endri.stef...@plus.alwrote:

  Hi Steve,

 ** **

 There are no errors I need to be able to change TON(below my PRI debug )
 in international or subscriber. The change in chan_dahdi.conf did not do it
 

 ** **

  Calling Number (len= 8) [ Ext: 0  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)

Presentation: Presentation permitted, user
 number not screened (0)  '1000' ]

  [70 0c a1 30 30 36 36 39 31 31 30 30 30 30]

  Called Number (len=14) [ Ext: 1  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)  'xx' ]

 ** **

 ** **

 Br

 ** **

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro
 *Sent:* Wednesday, September 25, 2013 4:24 PM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk TON number

  ** **

 I asked you before.  What exactly are you trying to do that you cannot?
  It helps to be fairly detailed when asking a question to the list.
  Include error messages if you have any.

 ** **

 The dialplan and your ISDN configs are different things.  It sounds like
 maybe you are having issues with with your dailplan and pattern matching.
  You can probably do all of that from the Trixbox GUI.  If you like
 Trixbox, check out FreePBX since Trixbox is done.  

 ** **

 Thanks,

 Steve Totaro

 ** **

 On Wed, Sep 25, 2013 at 9:30 AM, Endri Stefani endri.stef...@plus.al
 wrote:

 Hi guys

  

 Thanks a lot, I am just getting used to it, my telco managers J don’t
 trust stability for open source solution for voice(you give a headache to
 calling parties if lag is more than 250ms J ) and I want to prove them
 wrong. I have successfully integrated * with our system via ISDN and first
 calls look fast and clear but it is required to be flexible with TON number
 in order to be used. I tried unloading and loading chan-dahdi.conf like
 Hans suggested with no success TON was not changed.

 Here is my chan_dahdi.conf, is there anything else I should do in order
 for new pridailplan came into action :

  

 ;

 ; DAHDI telephony

 ;

 ; Configuration file

  

 [trunkgroups]

  

 [channels]

  

 switchtype = qsig

 context = pri_incoming

 group = 0

 signalling = pri_cpe

 channel = 1-15,17-31

  

  

  

 ;language=en

 ;context=from-zaptel

 ;signalling=fxs_ks

 ;rxwink=300  ;

 ; Whether or not to do distinctive ring detection on FXO lines

 ;

 ;usedistinctiveringdetection=yes

  

 usecallerid=yes

 hidecallerid=no

 callwaiting=yes

 usecallingpres=yes

 callwaitingcallerid=yes

 threewaycalling=yes

 transfer=yes

 cancallforward=yes

 callreturn=yes

 echocancel=yes

 echocancelwhenbridged=no

 echotraining=400

 rxgain=0.0

 txgain=0.0

 group=0

 callgroup=1

 pickupgroup=1

 immediate=no

  

  

 ;faxdetect=both

 faxdetect=incoming

 ;faxdetect=outgoing

 ;faxdetect=no

  

 ;Include setup-pstn configs

 #include dahdi-channels.conf

  

 group=1

  

 ;Include PBXconfig configs

 #include chan_dahdi_additional.conf

  

  

 unknown:Unknown

  private:Private ISDN

  local:  Local ISDN

  national:   National ISDN

  international:  International ISDN

  dynamic:Dynamically selects the appropriate dialplan

  redundant:  Same as dynamic, except that the underlying number is
 not 

  changed (not common)

  

 pridialplan = international

 prilocaldialplan = international

  

  

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *$$ dave cantera
 (android asus)
 *Sent:* Wednesday, September 25, 2013 2:46 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion


 *Subject:* Re: [asterisk-users] Asterisk TON number

  

 if you are a serious teleco guy, which it seems you are. you might
 consider dumping trixbox in the near future. while trixbox does provide a
 good entry level into the * world

Re: [asterisk-users] Sip-Client / type=peer / Why can this client place calls?

[Steve Totaro]

On Tue, Sep 3, 2013 at 8:11 AM, Thorsten Göllner t...@ovm-group.com wrote:

 Hi,

 I am using Asterisk 11.5.1. As far as I understood, the following
 configuration allows a sip client only to receive calls (type=peer) but not
 to place calls 
 (http://www.voip-info.org/**wiki/view/Asterisk+sip+typehttp://www.voip-info.org/wiki/view/Asterisk+sip+type).
 Why can I place calls though with this config?

 sip.conf
 ...
 [thorsten]
 type=peer
 host=dynamic
 context=my_context
 nat=force_rport,comedia
 secret=...
 dtmfmode=rfc2833
 disallow=all
 allow=g722
 allow=g729
 allow=g729
 ...

 extensions.conf
 ...
 [my_context]
 exten = _X.,1,Dial(DAHDI/g1/${EXTEN},**60)
 ...

 Of course: when removing a valid context the client can not place the
 call. But I thought this behaviour can be controlled via type=peer?!

 Thanks in advance
 -Thorsten-


See if this is helpful.

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
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Re: [asterisk-users] Conference calls wont traverse my trunk

[Steve Totaro]

On Thu, Jun 27, 2013 at 9:53 AM, DadoMaker dadoma...@gmail.com wrote:

 My conference call wont go thru my SIP trunk.  I may be missing a dialplan
 configuration setting as my PCM phone to phone calls go over the (GSM) tunk.


 The server with the conference:
 exten = 5777,1,GoTo(conf-confDemo,join,1)
 [conf-confDemo]
 exten = join,1,ConfBridge(confDemo/S/1)

 The server from which some users dial in from:
 exten = 5777,1,Dial(SIP/$EXTEN}@200_PBX)

 Any insight appreciated.

 Thanks,

 Dado


Dado, subject sounds like a personal problem.  Sorry couldn't resist.

How about some CLI debug info while trying a call?

Thanks,
Steve T
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Re: [asterisk-users] I need a second opinion on a new phone system deployment

[Steve Totaro]

http://red-fone.com http://red-fone.com/products-new/fonebridge/ might be
a good place look and see if other ideas pop up.  They have good products.
 I am not affiliated with them, just a happy user on a couple of
deployments.


On Fri, Jun 14, 2013 at 11:43 AM, Nunya Biznatch
aster...@ihearbanjos.comwrote:

 Howdy All,
They say opinions are like belly buttons, everybody has one. (that's
 the clean version of the saying). So I'm asking for yours. I hope you see
 it as a fun exercise.

 I'm designing a phone system from the ground up. Will be about 1000-1300
 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23
 buildings. Userbase is emergency services organization, 24/7/365 operation.
 Down time is not an option, but blips are acceptable. Repair time is
 immediate. We need failover for the failover essentially. However, money is
 a major factor, so I have to do it all for nothing. So here's what I'm
 thinking. Please throw in your 2 cents.

 Network will be separate for phones. Fiber infrastructure available
 between buildings as well as copper. Internet access will be limited to a
 single administrative console on a temporary basis, and then only when
 remote 3rd party support is required. Access for 3rd party support will be
 supervised through remote access tools such as VNC, GoToMeeting, etc...
 etc... System will have zero access to local data network. This means all
 ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be
 specific to the phone system. Yes, I know some responders at this time will
 become fixated on me gaining this connectivity. It ain't gonna happen. It's
 not an option. Period, end of story. These are the parameters I must work
 within. Trying to fix that will be a non-starter.

 The phone system will upgrade an existing TDM-based system. Mitel SX2000
 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect
 at least a one to two-year transition, meaning we will have time to find
 problems,  work bugs, and learn over time, with minimized impacts. It also
 means we'll be supporting two systems for some time.

 PBX is 97% serving your basic phone on the desk. Nothing special.
 Customers expect the usual list of features. There will be a goodly number
 of hints required for BLF on maybe 150 phones. There is one office of about
 30 phones in a call-center environment that will need that service. They
 would be considered low volume (but don't tell them that).

 My Skills... I am not a Linux kung fu master, but I have built and managed
 my share of Linux servers on mutiple Linux flavors. I am a DCAA, having
 been through formal training, and have been playing with Asterisk for
 years, but always in fits and spurts and never in a live environment so I
 am by no means a kung fu master there either. I have started dabbling with
 virtualizations via XEN, but I am not comfortable enough with it to go live
 this first round. I can see myself implementing it in about three years
 once we're totally comfortable with what we have, so I can then have time
 to get that skill sorted. I was a network engineer for the US no3. telecom
 for a number of years, 10-years in comm-electronics in the military before
 that. Telecom my entire career. I've got the kung-fu to handle the network
 side of the house, and having administrated multiple PBXs for decade-plus,
 I've got the concepts down.

 No plans to build databases for things like directories, etc... I'm not
 greatly confident in those skills, and to date, haven't found anything that
 really stands out that would make me require that. You may think otherwise,
 so please chime in. I say that, but at the same time I recognize I may
 require a GUI interface once fully deployed to allow lower-skilled people
 to follow the motions to complete simple moves, adds, and changes. I'm
 fighting the uphill battle that is the GUI is new, CLI is old mentality.

 System will use G.722 for VoIP Phones.

 So there's the groundwork. Here's the hardware plan.

 Plan is to build my own servers following industry standards (ATX) and
 using industry standard equipment. Why? Spares? Whether redundant or not, I
 will still have spares for the most common elements on the shelf so
 equipment can be returned to service as quickly as possible. This will also
 allow me to be comfortable with more basic server configurations and help
 keep cost down. For example, Servers with single power supplies vs. dual.
 Also, components will be standardized for all equipment to aid in supply
 requirements.

 First the layout.

 2-servers acting as gateways. Each handling 2 PRIs for outside trunks.
 They'll also handle the analog ports. Failover will be in the form of
 degraded trunk access if one should fail, but the second will be able to
 support services in degraded fashion.

 2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be
 capable of handling the load of the entire system, and the other will
 pickup when 

Re: [asterisk-users] asterisk fax in debian

[Steve Totaro]

On Thu, Jun 13, 2013 at 12:29 PM, vortex binary.vor...@gmail.com wrote:

 Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to
 send to email the voicemails.
 i would like to get rid of the analog fax machine and use asterisk to
 send/receive faxes.
 I do have a PSTN line with a SPA3102 adapter to interface it to asterisk.
 The number of the PSTN line is dedicated to faxing only. So i would like to:
 -receive faxes to asterisk and then send it as PDFs to an email address
 -Send from my PC a fax directly.

 is there any guide on how to do that since i got lost with all of it?



I would go with HylaFAX.  FAX is an art with any VoIP solution.  The best
art I have done and seen turned out to use HylaFAX.

Thanks,
Steve Totaro
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Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)

[Steve Totaro]

On Thu, Jun 13, 2013 at 9:23 AM, James Bensley jwbens...@gmail.com wrote:

 Hi All,

 I am looking for a way to troubleshoot issues with TDM (E1) trunks
 with a provider.

 Currently with SIP trunks I am using tcpdump to perform packet
 captures between our gateways and the SIP providers IPs, capturing
 traffic on all ports, to include both the SIP messages and the RTP
 stream.

 How can I achieve a similar result on TDM links connected to TDM cards
 in Asterisk servers, where by I can see the signalling (like the SIP
 message) and the audio stream (like the RTP stream) in my packet
 captures?

 If it helps, the end goal is to create something like the packet
 captures I am making so I can see the control signals and audio
 streams (in and out) for troubleshooting one way audio issues for
 example. So, am I sending audio to the TDM provider, are they sending
 it to me, have we both signalled correctly to start/stop sending
 audio, etc.

 Many thanks,
 James.


Is it PRI?  You can see PRI debug info on the console.  Extremely valuable
in troubleshooting.   http://www.voip-info.org/wiki/view/Asterisk+CLI

Zap channel commands

zap destroy channel: Destroy a channel
zap show channels: Show active zapata channels
zap show channel: Show information on a channel
zap show status: lists all the Zaptel spans. A span will apear here whether
or not its channels are configured with chan_zap.
zap show cadences: Show the configured ring cadences (available e.g with
Zap/1r2).
zap set swgain(= 1.6): set the (software) gain for a hannel. Temporary
equivalents of rxgain and txgain in zapata.conf.
zap set hwgain(=1.6): set the hardware gain for channels that support it.
zap set dnd(=1.6) set a channel's do-not-disturb mode on or off.


The following commands are available if the channel is built with support
for libpri:

pri debug span: Enables PRI debugging on a span
pri intense debug span: Enables REALLY INTENSE PRI debugging
pri no debug span: Disables PRI debugging on a span
pri show spans: List spans and their status.
pri show span: Information about a span.
pri show debug: show where debug is enabled.
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Re: [asterisk-users] A problem with IAX2

[Steve Totaro]

On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.comwrote:

 B.H.

 Hello!

 We have several Asterik boxes that are connected to PSTN using PRI cards
 and they are interconnected using IAX2 trunks so that incoming calls are
 delivered from PSTN to the servers they belong to.

 In past we were using asterisk 1.4 on the server that is receiving IAX
 connections and everything worked as expected. Recently, we have switched
 to a newer box with asterisk 1.8.22 and then we began to experience
 sometimes a strange problem:

 At some point of time, incoming IAX connections begin to get refused by
 the server and we get the following messages in the logs:

 WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from
 address X.X.X.X

 where X.X.X.X is the IP of the PSTN-IAX gateways and all the incoming
 calls start to be rejected.

 Direct PSTN calls (both incoming and outgoing) to the same server work OK.
 The only solution that helps is to kill the asterisk and restart it.

 All the servers are connected to the same LAN segment, with gigabit
 switch, there is no problems with the network. No packet loss.

 There's already bug report present with very similar issue, but it is
 suspended and, like stated there, the problem is very hard to reproduce.

 See: https://issues.asterisk.org/jira/browse/ASTERISK-21762


 --
 משיח NOW!


Use SIP and never look back.

Thanks,
Steve Totaro
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Re: [asterisk-users] OC3/STM-1 Line Card

[Steve Totaro]

Adtran MX2800 is rock solid.  Save some money and use NFAS.

Thanks,
Steve Totaro


On Sun, Jun 9, 2013 at 10:11 PM, Nick Khamis sym...@gmail.com wrote:

 Thank you so much for your responses!!! With this route we would have
 to manage so many * boxes with T1s, not to mention, the hit we would
 take on the MUX. Any decent DS/T3 cards out there?

 N.

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Re: [asterisk-users] Asterisk Calendar integration suggestions

[Steve Totaro]

Without knowing requirements, Sugar CRM seems to be the most supported.

Thanks,
Steve Totaro


On Thu, Apr 25, 2013 at 9:22 AM, j...@millican.us j...@millican.us wrote:

 Hello all,
 I am looking into building a calendar server (due to business requierments
 I can not use public hosted calender like Google), and am looking for
 suggestions based on experience with different calendar
 applications/servers available for Linux that you have integrated with
 Asterisk.  If you can give a quick, simple list of what worked and what
 didn't I would be very grateful.
 Thank You,
 John


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Re: [asterisk-users] Redirect incoming call to SIP trunk.

[Steve Totaro]

On Wed, Mar 6, 2013 at 3:48 PM, Administrator TOOTAI ad...@tootai.net wrote:
 Le 06/03/2013 17:57, Luis H. Forchesatto a écrit :

 Solved.


 Great, happy for you.

 What would be nice is to explain how you solve it for archives. Other people
 can run in the same problematic that yours and would be happy to see your
 way to get out of it


I would bet you that is exactly what he did.  This list has died off
so much because you can find almost every answer in the archives now.

Thanks,
Steve Totaro

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Re: [asterisk-users] Wierd question - Give me your opinion please

[Steve Totaro]

On Mon, Feb 4, 2013 at 11:11 PM, Jared Baxley jared.bax...@gmail.com wrote:
 Client - Not for Profit in the Middle of the Jungle/Rain Forrest

 Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding,
 and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge
 Podge of DYI wiring across remaining buildings. Phones - Total of about 50
 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will
 have to be analog due to the distance.

 Analog Extensions will be on Digium TDM2400 or Sangoma A400 Cards.

 Analog extensions WILL Hit a Surge Gate before the cards, and as much
 precaution on grounding protection and power protection is being taken as
 possible. The cards WILL BE PCI not PCI-e (They are being donated)

 A New Dell Power-edge Server will be acquired for the PBX

 HERE IS MY QUESTION

 Would you purchase a NEW TOWER Server with PCI slots to accommodate the
 cards,

 OR

 Purchase a NEW RACK MOUNT server for the PBX, and Buy/Build a Cheaper Server
 just for the analog extensions,


 I'm torn... The ease of management of one server, or the isolation of
 analog extensions scattered through the jungle on it's own server.

 Opinions?


For that number of analog stations, I would go with tried and true
channel banks.  Adtran or Adit would be my personal choice.  I would
probably consider not even using any IP phones to keep things very
simple.

Thanks,
Steve T

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Re: [asterisk-users] DIDForSale spam

[Steve Totaro]

Just ignore the very clear subject line.

On Thu, Jan 10, 2013 at 1:04 PM, Roy Abshire r...@coopvr.com wrote:
 I just ignore spam if I'm not interested and flag them so they go right into
 my trash folder.
 I think its more exhausting debating the issue on this forum.
 I got the email too from DIDForSale but now I'm getting alot more from this
 thread.

 It really didn't bother me as much as reading all the posts but that's just
 me...now back to Asterisk issues :)

 Co-op Vacation Rentals
 www.coopvr.com
 15218 Summit Ave
 Suite #300-354
 Fontana, CA 92336
 Phone/Fax (855) 760-COOP (2667)

 On 1/10/2013 9:26 AM, chris wrote:

 +1

 On Jan 9, 2013 8:59 PM, Don Kelly d...@donkelly.biz wrote:

 Jai,



 It should not be necessary for me to remove my email address from your
 list. It should not be on there to start with—we do not have, and have never
 had, a relationship that justified you sending me email.

 --Don

 Don Kelly

 PCF Corp
 People Come First
 651 842-1000
 651 842-1001 fax



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jai Rangi
 Sent: Wednesday, January 09, 2013 7:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] DIDForSale spam



 Guys,
 Since I am attached to did for sale:
 My apology to every one who received the DIDForSale 2012 Achievement
 email and you hated it.

 As a asterisk user my question will be.
 If some xyz company send you a so called spam email, what made you think
 that you should spam the mailing lists. I am sure we all get lots if spam
 emails every day. If you really got some time and talent, why don't you
 write some good tips and tricks about asterisk.

 Long story short We have a link where you can unsubscribe your email for
 any further communication.
 http://www.didforsale.com/unsubscribe.php  or Send me your email  address
 I will personally take care of that and will remove your email. This will
 take less than 5 seconds.
 I am sure there will be lot of arguments on why you should that and all. I
 will refrain myself on any further unproductive communication.

 Happy new year to you all.


 On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:

 +1 here.

 On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
 wrote:

 On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
  On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
  What were the senders IP(s)?
 
  Will have to look it up when I get home.
 
  Doug
 
  --
  Ben Franklin quote:
 
  Those who would give up Essential Liberty to purchase a little
  Temporary Safety, deserve neither Liberty nor Safety.
 
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  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  I have gotten hit with this twice so far. in March and Today:
 
  Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
  3/8/12
 
  DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
  1/9/13
 
  UGH, when I asked in March where he got my email he said:
 
  Hi Chris,
  We got your contact from the Internet. Let me know the good time to
  talk about this in detail.
  Thank you,
  -Rohit Dhaka
 

 Obviously by harvesting these lists.  I received 2 myself.

 Thanks,
 Steve T

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Re: [asterisk-users] DIDForSale spam

[Steve Totaro]

So what asterisk issue do you have?  Let's fix it.

On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler
rwhee...@artifact-software.com wrote:
 That does not solve any asterisk issue that I have.



 On 10/01/2013 1:32 PM, Carlos Alvarez wrote:

 On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote:

 It really didn't bother me as much as reading all the posts but that's
 just me...now back to Asterisk issues :)


 Sorry to add another, but for me, the main point is that this activity
 speaks to the character, ethics, and trustworthiness of the company doing
 it.  We all have spam filters.  I just also add the company to my do not
 buy/do not recommend list.


 --
 Carlos Alvarez
 TelEvolve
 602-889-3003



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 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102


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Re: [asterisk-users] DIDForSale spam

[Steve Totaro]

A tier one provider.

On Thu, Jan 10, 2013 at 3:44 PM, Carlos Alvarez car...@televolve.com wrote:
 Hopefully it's not, What is the best DID provider for Asterisk...


 On Thu, Jan 10, 2013 at 1:37 PM, Steve Totaro
 stot...@totarotechnologies.com wrote:

 So what asterisk issue do you have?  Let's fix it.

 On Thu, Jan 10, 2013 at 1:49 PM, Ron Wheeler
 rwhee...@artifact-software.com wrote:
  That does not solve any asterisk issue that I have.
 
 
 
  On 10/01/2013 1:32 PM, Carlos Alvarez wrote:
 
  On Thu, Jan 10, 2013 at 11:04 AM, Roy Abshire r...@coopvr.com wrote:
 
  It really didn't bother me as much as reading all the posts but that's
  just me...now back to Asterisk issues :)
 
 
  Sorry to add another, but for me, the main point is that this activity
  speaks to the character, ethics, and trustworthiness of the company
  doing
  it.  We all have spam filters.  I just also add the company to my do not
  buy/do not recommend list.
 
 
  --
  Carlos Alvarez
  TelEvolve
  602-889-3003
 
 
 
  --
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  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
  --
  Ron Wheeler
  President
  Artifact Software Inc
  email: rwhee...@artifact-software.com
  skype: ronaldmwheeler
  phone: 866-970-2435, ext 102
 
 
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Re: [asterisk-users] Streaming/Recording audio

[Steve Totaro]

On Wed, Jan 9, 2013 at 2:54 AM, Grant Bagdasarian g...@cm.nl wrote:
 Hello,



 For some reason I did not receive any replies related to my question by
 mail, but I found the topic back on the online mailing archives. I hope by
 supplying the same subject this email will be logged in my previously
 created topic instead of a new one. If it does not, I apologize.



 Regarding my second question, is the recorded stream also available in the
 dialplan after the recording has finished?



 Regards,



 Grant



I am not sure what you are asking.  What do you mean available in the
dialplan?  What are you trying to do?

Did you figure out your streaming question?  Not sure if it is
deprecated, but app_ices was how I did streaming many moons ago.

Thanks,
Steve T

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Re: [asterisk-users] Streaming/Recording audio

[Steve Totaro]

On Wed, Jan 9, 2013 at 4:16 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
 On Wed, Jan 9, 2013 at 2:54 AM, Grant Bagdasarian g...@cm.nl wrote:
 Hello,



 For some reason I did not receive any replies related to my question by
 mail, but I found the topic back on the online mailing archives. I hope by
 supplying the same subject this email will be logged in my previously
 created topic instead of a new one. If it does not, I apologize.



 Regarding my second question, is the recorded stream also available in the
 dialplan after the recording has finished?



 Regards,



 Grant



 I am not sure what you are asking.  What do you mean available in the
 dialplan?  What are you trying to do?

 Did you figure out your streaming question?  Not sure if it is
 deprecated, but app_ices was how I did streaming many moons ago.

 Thanks,
 Steve T

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices

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Re: [asterisk-users] DIDForSale spam

[Steve Totaro]

On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
 On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
 What were the senders IP(s)?

 Will have to look it up when I get home.

 Doug

 --
 Ben Franklin quote:

 Those who would give up Essential Liberty to purchase a little Temporary 
 Safety, deserve neither Liberty nor Safety.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


 I have gotten hit with this twice so far. in March and Today:

 Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
 3/8/12

 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
 1/9/13

 UGH, when I asked in March where he got my email he said:

 Hi Chris,
 We got your contact from the Internet. Let me know the good time to
 talk about this in detail.
 Thank you,
 -Rohit Dhaka


Obviously by harvesting these lists.  I received 2 myself.

Thanks,
Steve T

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Re: [asterisk-users] new user help required to build voice recorder with asterisk

[Steve Totaro]

Top post for the New Year.

Yes, if you might scale up to 60 or more simultaneous calls,
definitely look at OrecX or RTPTap because you will run into I/O
issues.  Not sure what current hardware can accommodate but it is best
not to find out.

Considering the very low cost of hardware these days compared with the
cost of possible downtime, poor audio, lost recordings or whatever
else you can assign a monetary value, I always suggest a separate
machine for Passive recording when dealing with more than a handful
of simultaneous calls.

Thanks,
Steve Totaro

On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
 With just one PRI card this should not be an issue, but for larger systems
 you may consider using something like Oreka to offload the I/O from the
 Asterisk server
 l.


 2012/12/31 Vinod Nadiadwala thinw...@gmail.com

 Hi,

 I am new to asterisk, i want to know that is it possible to use asterisk
 for build voice recording system.

 Scenario :
 ISDN PRI line (30 line)
 I want every incoming  outgoing call has to recorded, but without manual
 action. system has to auto receive the call.

 Please suggest, how should i start and with which hardware / cards it is
 possible.




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 Test-drive WombatDialer beta @ http://wombatdialer.com

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Re: [asterisk-users] new user help required to build voice recorder with asterisk

[Steve Totaro]

It depends on what you do with them.

Years ago, 60 calls would start to crap out audio on live calls and I
learned that the hard way on a production call center.  There was the
I/O of just SLIN, then converting to MP3, then transferring to a not
too forgiving SAMBA share.  Scheduling things for a slower times and
moving the MP3 conversion to the mass storage significantly helped
while scrambling to find the permanent solution.

People could increase those numbers with RAMDisk and other tricks but
just moving it off the Phone System makes more sense.

Why not engineer something to scale and last without knowing that you
will have to revisit it and quite possibly at the most inopportune
time, like when you just spent a good deal of money on an advertising
spot?

Thanks,
Steve T

On Wed, Jan 2, 2013 at 7:35 AM, Leandro Dardini ldard...@gmail.com wrote:
 I don't know how many I/O can be achieved on a modern hardware, but I don't
 think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of
 data. However can be a good idea to start loading a server and be prepared
 to share the load on another server.

 Leandro


 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com

 Top post for the New Year.

 Yes, if you might scale up to 60 or more simultaneous calls,
 definitely look at OrecX or RTPTap because you will run into I/O
 issues.  Not sure what current hardware can accommodate but it is best
 not to find out.

 Considering the very low cost of hardware these days compared with the
 cost of possible downtime, poor audio, lost recordings or whatever
 else you can assign a monetary value, I always suggest a separate
 machine for Passive recording when dealing with more than a handful
 of simultaneous calls.

 Thanks,
 Steve Totaro

 On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com
 wrote:
  With just one PRI card this should not be an issue, but for larger
  systems
  you may consider using something like Oreka to offload the I/O from the
  Asterisk server
  l.
 
 
  2012/12/31 Vinod Nadiadwala thinw...@gmail.com
 
  Hi,
 
  I am new to asterisk, i want to know that is it possible to use
  asterisk
  for build voice recording system.
 
  Scenario :
  ISDN PRI line (30 line)
  I want every incoming  outgoing call has to recorded, but without
  manual
  action. system has to auto receive the call.
 
  Please suggest, how should i start and with which hardware / cards it
  is
  possible.
 
 
 
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
  --
  Loway - home of QueueMetrics - http://queuemetrics.com
  Test-drive WombatDialer beta @ http://wombatdialer.com
 
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Re: [asterisk-users] new user help required to build voice recorder with asterisk

[Steve Totaro]

Mixmonitor also muxes the two sides of the conversation after hangup.
That is quite a bit of I/O for 60 simultaneous calls lasting an
average of 5-15mins

On Wed, Jan 2, 2013 at 9:59 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
 It depends on what you do with them.

 Years ago, 60 calls would start to crap out audio on live calls and I
 learned that the hard way on a production call center.  There was the
 I/O of just SLIN, then converting to MP3, then transferring to a not
 too forgiving SAMBA share.  Scheduling things for a slower times and
 moving the MP3 conversion to the mass storage significantly helped
 while scrambling to find the permanent solution.

 People could increase those numbers with RAMDisk and other tricks but
 just moving it off the Phone System makes more sense.

 Why not engineer something to scale and last without knowing that you
 will have to revisit it and quite possibly at the most inopportune
 time, like when you just spent a good deal of money on an advertising
 spot?

 Thanks,
 Steve T

 On Wed, Jan 2, 2013 at 7:35 AM, Leandro Dardini ldard...@gmail.com wrote:
 I don't know how many I/O can be achieved on a modern hardware, but I don't
 think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of
 data. However can be a good idea to start loading a server and be prepared
 to share the load on another server.

 Leandro


 2013/1/2 Steve Totaro stot...@asteriskhelpdesk.com

 Top post for the New Year.

 Yes, if you might scale up to 60 or more simultaneous calls,
 definitely look at OrecX or RTPTap because you will run into I/O
 issues.  Not sure what current hardware can accommodate but it is best
 not to find out.

 Considering the very low cost of hardware these days compared with the
 cost of possible downtime, poor audio, lost recordings or whatever
 else you can assign a monetary value, I always suggest a separate
 machine for Passive recording when dealing with more than a handful
 of simultaneous calls.

 Thanks,
 Steve Totaro

 On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri lenz.lo...@gmail.com
 wrote:
  With just one PRI card this should not be an issue, but for larger
  systems
  you may consider using something like Oreka to offload the I/O from the
  Asterisk server
  l.
 
 
  2012/12/31 Vinod Nadiadwala thinw...@gmail.com
 
  Hi,
 
  I am new to asterisk, i want to know that is it possible to use
  asterisk
  for build voice recording system.
 
  Scenario :
  ISDN PRI line (30 line)
  I want every incoming  outgoing call has to recorded, but without
  manual
  action. system has to auto receive the call.
 
  Please suggest, how should i start and with which hardware / cards it
  is
  possible.
 
 
 
 
  --
  _
  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
  New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
  --
  Loway - home of QueueMetrics - http://queuemetrics.com
  Test-drive WombatDialer beta @ http://wombatdialer.com
 
  --
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Re: [asterisk-users] Top Posting

[Steve Totaro]

On Wed, Jan 2, 2013 at 11:00 AM, Richard Kenner ken...@gnat.com wrote:
  I'm the opposite.  I'm likely not to scroll down 10 pages to see
  the comments at the end.

 Wouldn't need to if people trimmed their posts properly.

 Precisely (e.g., see above)! Indeed, my sense is that top-posting
 *discourages* properly trimming email and that's my main reason against it.
 If things were properly trimmed, the email would be short enough that it
 really doesn't matter that much if the new material is on the top or
 bottom, but people who top-post and don't trim create really hard-to-follow
 emails.


Not really true often times when people do the right thing and post
debug and conf files often required to get meaningful help.

Thanks,
Steve T

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Re: [asterisk-users] Top Posting

[Steve Totaro]

On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote:
 On 1/2/2013 11:30 AM, Richard Kenner wrote:

 If things were properly trimmed, the email would be short enough that it
 really doesn't matter that much if the new material is on the top or
 bottom, but people who top-post and don't trim create really
 hard-to-follow
 emails.

 Not really true often times when people do the right thing and post
 debug and conf files often required to get meaningful help.

 Yes, but if you put those at the end, where they belong, people reading
 the email can follow the thread quite easily and can ignore those if
 they don't need them.  Certainly only a tiny part of such, if any at all,
 should be included in a reply.

 Ok folks, could not stop myself any longer.   This pissing and moaning is
 foolish to say the least.  There was a post a while ago in the original
 hijacked thread by Steve Edwards that gave a link to the rules of the list
 at:
 http://www.asterisk.org/community/discuss/

 GO READ THEM!

 Directly before the list of Rules is:

 Show consideration. It's important to read the rules before posting on a
 mailing list.

 Sage advice if you ask me, and yes I know nobody actually asked me.

 It is not hard to follow the rules .  If the nice folks at Digium took the
 time to post rules we should at least TRY to follow them. If you do not like
 the rules you can always petition Digium to change them but, taking up
 bandwidth on the list in this all to frequent pissing match is a futile
 waste of time.

 Grow up, follow the rules, have a good day.
 JohnM


I became a list member way before any such rule and never had to click
through and agree to these update ToS.

I am grandfathered in.

Thanks,
Steve Totaro

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Re: [asterisk-users] Top Posting

[Steve Totaro]

On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote:
 On 1/2/2013 12:20 PM, Steve Totaro wrote:

 On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us
 wrote:

 On 1/2/2013 11:30 AM, Richard Kenner wrote:

 If things were properly trimmed, the email would be short enough that
 it
 really doesn't matter that much if the new material is on the top or
 bottom, but people who top-post and don't trim create really
 hard-to-follow
 emails.

 Not really true often times when people do the right thing and post
 debug and conf files often required to get meaningful help.

 Yes, but if you put those at the end, where they belong, people reading
 the email can follow the thread quite easily and can ignore those if
 they don't need them.  Certainly only a tiny part of such, if any at
 all,
 should be included in a reply.

 Ok folks, could not stop myself any longer.   This pissing and moaning is
 foolish to say the least.  There was a post a while ago in the original
 hijacked thread by Steve Edwards that gave a link to the rules of the
 list
 at:
 http://www.asterisk.org/community/discuss/

 GO READ THEM!

 Directly before the list of Rules is:

 Show consideration. It's important to read the rules before posting on a
 mailing list.

 Sage advice if you ask me, and yes I know nobody actually asked me.

 It is not hard to follow the rules .  If the nice folks at Digium took
 the
 time to post rules we should at least TRY to follow them. If you do not
 like
 the rules you can always petition Digium to change them but, taking up
 bandwidth on the list in this all to frequent pissing match is a futile
 waste of time.

 Grow up, follow the rules, have a good day.
 JohnM

 I became a list member way before any such rule and never had to click
 through and agree to these update ToS.

 I am grandfathered in.

 Thanks,
 Steve Totaro

 So Steve, can I steal this and send it to the IRS? The ATF? Local Police
 Department? G  Wouldn't that be nice!  Sorry couldn't  resist.

 JohnM


What the hell are you implying?  The local police love me, I am in
good standing with the ATF, FBI, DoD, DoS, USAID, DoE, DoL, and NSA.

IRS wants some money in April but don't they always? LOL.

Thanks,
Steve T

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Re: [asterisk-users] Asterisk as answering machine

[Steve Totaro]

Problematic at best.  Just make a phone an extension and allow that to
ring in a hunt group.

Thanks,
Steve Totaro

On Wed, Jan 2, 2013 at 3:30 PM, Ron Wheeler
rwhee...@artifact-software.com wrote:
 I have connected a PSTN line to a Digium FXO card.
 There is also an ordinary analogue phone attached to the same line.

 The Asterisk answers the line on the first ring.

 I would like it to wait for a few seconds so that someone can answer the
 PSTN line with an analogue phone.
 This would allow a person to directly pick up the line if they wanted to or
 if not, let it go to the Asterisk where it would be dispatched through the
 normal process.

 Currently, as soon as the analogue phone rings, the Asterisk PBX has already
 answered the call and starts the You have reached. Dial  and tries
 to dispatch the call.

 This makes it hard to carry on a conversation.

 Ron

 --
 Ron Wheeler
 President
 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: ronaldmwheeler
 phone: 866-970-2435, ext 102


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Re: [asterisk-users] Top Posting

[Steve Totaro]

On Wed, Jan 2, 2013 at 3:46 PM, jon pounder j...@inline.net wrote:
 On 01/02/2013 03:22 PM, Patrick Lists wrote:

 On 01/02/2013 06:20 PM, Steve Totaro wrote:

 I became a list member way before any such rule and never had to click
 through and agree to these update ToS.

 I am grandfathered in.


 Just looked it up. I see my first post back in April 2003, yours in
 September 2003 and Jon in March 2003. Wow you find something fun to play
 with and suddenly a decade has passed :-)


 Are you sure about that ? I know I was doing stuff with asterisk back in the
 LSS days and that was around 2001



The archives are a bit sketchy before Feb of 2003.  I would guess my
first dabble was circa 2001 and started making money from it in 2002.
Right around the debut of the 3COM NBX 100.

Thanks,
Steve T

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Re: [asterisk-users] Top Posting

[Steve Totaro]

Yeah.  I never really got the whole fanatical top vs bottom thing.
Whatever, I have answered way more than my fair share of free
questions (as in beer).  The person asking was always quite happy to
get a meaningful and helpful reply, no matter where it was in the body
of the content.

Why people get worked up over small things is beyond me.  Embrace the Chaos.

Thanks,
Steve T

On Sun, Dec 30, 2012 at 5:42 PM, jon pounder j...@inline.net wrote:
 On 12/30/2012 03:54 PM, Benny Amorsen wrote:

 Boy what an elitist attitude.

 I have been on this list far longer than most people - long before digium
 even existed and if you don't value what I have to say - well just don't
 read it.

 If you or your mail reader can't slice and dice a mailing list the way you
 want to see it well maybe its your opinions us top posters won't miss, since
 clearly you are lacking the skills to even have your tools format documents
 for you.




 Gergo Csibra csi...@gmail.com writes:

 Complaining about top posting on a list where's no moderation,
 no sanction if somebody top posting is pointless.

 There is a sanction. People like me will score top posters lower and
 soon not see their posts at all.

 It is often a quick way to see if it is worth responding to someone. If
 they top post, nothing of value is likely to come out of the
 conversation.

 So by all means, everybody who wants to, keep top posting.


 /Benny



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Re: [asterisk-users] Paging for Praying

[Steve Totaro]

On Fri, Dec 28, 2012 at 11:35 PM, John Novack
jnov...@stromberg-carlson.org wrote:

 Shaun Ruffell wrote:

 On Fri, Dec 28, 2012 at 06:41:38PM -0800, Steve Edwards wrote:

 On 12/28/2012 08:13 PM, Steve Edwards wrote:

 Please don't top-post. If you don't know what that means, please
 consult Google.

 On Fri, 28 Dec 2012, jon pounder wrote:

 Please stop saying don't top post, some of us prefer it that way.

 Besides being my preference, it is the documented rule of the
 mailing list:

 http://www.asterisk.org/community/discuss/

 Note Mailing List Rules, #5.

 For a walk down memory lane on top vs bottom posting on the Asterisk
 mailing lists:

 http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/254997

 I would add that the rule # 5 was added long after the first 4, by someone
 in charge after one of the many times this subject has popped up.

 Many of the same complainers  routinely do not remove the multi line
 footers, sometimes MANY of them, forcing those who really want to read a
 reply to wade through  the mess. Seems some can't be bothered to delete them

 I would also add that rules are made to be broken!

 Peg Leg O'Brien


 --

 Dog is my Co-pilot


Wow.  Didn't know there was a rule...  I never got the whole argument.
 If the flow is top posting, I top post.  If I am certain I can answer
a simple question with a simple answer, sometimes I will just top
post.

Totally off topic, apologies.  I know I have weighed in once or twice
but really never cared.  If the flow was totally borked and I didn't
care enough to follow the topic, it wasn't that important anyways.

Even more off topic.  Can someone smarter than me get the post totals
for each year?  I was #1 one year.  I am not even talking about
individual post counts though.  It just seems the list has died for
the most part.

Thanks,
Steve T

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Re: [asterisk-users] CHANNEL(t38passthrough) is 0

[Steve Totaro]

Do you have reinvite allowed?  That was an issue on one of my
installations if I am remembering correctly.  Any debug, logs, confs
that would help?

Thanks,
Steve Totaro

On Thu, Dec 27, 2012 at 12:15 PM, Eric Wieling ewiel...@nyigc.com wrote:
 Setting directmedia=no does not help.  The calls still go through and the fax 
 still fails after switching to T.38.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
 Sent: Thursday, December 27, 2012 10:49 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] CHANNEL(t38passthrough) is 0

 I'm trying to get T.38 passthrough working on Asterisk 1.8.18.1.  It isn't 
 working.   Calls go through and are answered, but the fax machines are unable 
 to communicate.   I checked the value of CHANNEL(t38passthrough) and it seems 
 to always be 0.   One side is Level 3 T.38 TN and the other side is an Adtran 
 NetVanta with POTS ports.   What would cause Asterisk to not offer t38 
 passthrough on a channel?

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Re: [asterisk-users] IAX2 over OpenVPN connection.... working but

[Steve Totaro]

On Sun, Dec 9, 2012 at 2:54 PM, Stephen Brown stephen.brow...@gmail.com wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 So a friend of mine and I setup a static key based point to point
 OpenVPN connection from my box to his for the express intent of carrying
 IAX traffic encrypted.

 His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His
 PBX is located at 172.30.1.48 and mine is at 10.0.30.2. We had an
 existing working IAX trunk in place prior to the VPN, and after we
 brought the VPN up we set the host= parameter within Asterisk
 accordingly on each end to match the local IP's and discovered it did
 not work. The trunk remained in an UNKNOWN status on each end, even
 though we could ping each box locally, SSH, and even SIP worked.

 Here's where I am baffled and I am hoping someone with intricate
 knowledge of this implementation may be able to explain it to me. What
 we had to do to get this working was to set the host= parameter to the
 respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
 172.10.1.2 in his case. Calls flow normally now and we cannot understand
 how or why. I would have assumed with a destination of either LAN as
 defined by the routing table it would have left out on the OpenVPN
 connection by default, and what's even more strange is that IAX is the
 only protocol that does not appear to function as intended.

 Any takers? :)


 -BEGIN PGP SIGNATURE-
 Version: GnuPG v2.0.17 (MingW32)

 iEYEARECAAYFAlDE7GcACgkQ3sJXNEncx7is9QCcCciMYFJ7ZXjYxuHC2EYD0PZY
 waAAniNNx8GuC5To7ajlGR5sYs3yftFK
 =lcWJ
 -END PGP SIGNATURE-



First, not so much of an answer but more of a question.  Why use IAX2
in your scenario?  SIP would seem to be very logical in this case if
you already tested it and it works.

IAX2 really only has merits where NAT and multiple ports are an issue.
 It has been known to create many problems and headaches.

Since OpenVPN negates the multiple ports over the web, and NAT isn't a
problem from what you have stated, why even bother with IAX2?

To cleanly solve your issue, create an OpenVPN tunnel directly between
the boxen with the same IP/subnet scheme.  That is what I would do, as
each tunnel is a subinterface of sorts, there is no need to keep the
addressing scheme of your LANs.  SIP and IAX2 should both work for you
(I still suggest SIP).  Creating a separate subnet for your OpenVPN
connection will arguably also add a bit of security between networks.

What does your IPtables look like?  Maybe you are blocking IAX?  Turn
of debugging and post verbose.

Thanks,
Steve T

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Re: [asterisk-users] PRI can receive calls but cannot dial out

[Steve Totaro]

: 0  Info 
 transfer capability: Speech (0)
 PRI Span: 4   Ext: 1  Trans mode/rate: 64kbps, 
 circuit-mode (16)
 PRI Span: 4 User information layer 1: A-Law 
 (35)
 PRI Span: 4  [18 01 81]
 PRI Span: 4  Channel ID (len= 3) [ Ext: 1  IntID: Implicit  BRI  Spare: 0  
 Preferred  Dchan: 0
 PRI Span: 4ChanSel: B1 channel
 PRI Span: 4  ]
 PRI Span: 4  [6c 06 21 80 34 30 35 33]
 PRI Span: 4  Calling Party Number (len= 8) [ Ext: 0  TON: National Number 
 (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 PRI Span: 4  Presentation: Presentation 
 allowed, User-provided, not screened (0)  '4053' ]
 PRI Span: 4  [70 0a 80 36 35 36 36 36 30 34 39 39]
 PRI Span: 4  Called Party Number (len=12) [ Ext: 1  TON: Unknown Number Type 
 (0)  NPI: Unknown Number Plan (0)  'XX' ]
 PRI Span: 4 q931.c:6291 q931_setup: Call 32774 enters state 1 (Call 
 Initiated).  Hold state: Idle
 -- Called DAHDI/g2/XX
 PRI Span: 4 T303 timed out.  cref:32774
 PRI Span: 4
 PRI Span: 4  DL-DATA request
 PRI Span: 4  Protocol Discriminator: Q.931 (8)  len=32
 PRI Span: 4  TEI=0 Call Ref: len= 1 (reference 6/0x6) (Sent from originator)
 PRI Span: 4  Message Type: SETUP (5)
 PRI Span: 4 TEI=0 Transmitting N(S)=11, window is open V(A)=11 K=1
 PRI Span: 4
 PRI Span: 4  Protocol Discriminator: Q.931 (8)  len=32
 PRI Span: 4  TEI=0 Call Ref: len= 1 (reference 6/0x6) (Sent from originator)
 PRI Span: 4  Message Type: SETUP (5)
 PRI Span: 4  [04 03 80 90 a3]
 PRI Span: 4  Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info 
 transfer capability: Speech (0)
 PRI Span: 4   Ext: 1  Trans mode/rate: 64kbps, 
 circuit-mode (16)
 PRI Span: 4 User information layer 1: A-Law 
 (35)
 PRI Span: 4  [18 01 81]
 PRI Span: 4  Channel ID (len= 3) [ Ext: 1  IntID: Implicit  BRI  Spare: 0  
 Preferred  Dchan: 0
 PRI Span: 4ChanSel: B1 channel
 PRI Span: 4  ]
 PRI Span: 4  [6c 06 21 80 34 30 35 33]
 PRI Span: 4  Calling Party Number (len= 8) [ Ext: 0  TON: National Number 
 (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 PRI Span: 4  Presentation: Presentation 
 allowed, User-provided, not screened (0)  '4053' ]
 PRI Span: 4  [70 0a 80 36 35 36 36 36 30 34 39 39]
 PRI Span: 4  Called Party Number (len=12) [ Ext: 1  TON: Unknown Number Type 
 (0)  NPI: Unknown Number Plan (0)  'XX' ]
 PRI Span: 4 T303 timed out.  cref:32774
 PRI Span: 4 q931.c:6180 t303_expiry: Call 32774 enters state 0 (Null).  Hold 
 state: Idle
 PRI Span: 4 Fake clearing.  cref:32774
 PRI Span: 4 q931.c:9551 pri_internal_clear: alive 1, hangupack 1
 Span 4: Processing event PRI_EVENT_HANGUP(6)
 -- Span 4: Channel 0/1 got hangup, cause 18
 PRI Span: 4 q931.c:7092 q931_hangup: Hangup other cref:32774
 PRI Span: 4 q931.c:6849 __q931_hangup: ourstate Null, peerstate Null, 
 hold-state Idle
 PRI Span: 4 Destroying call 0xb85c61d0, ourstate Null, peerstate Null, 
 hold-state Idle
 -- Hungup 'DAHDI/i4/XX-6'
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Auto fallthrough, channel 'SIP/4053-0003' status is 'CHANUNAVAIL'
 -- Executing [h@company:3] Hangup(SIP/4053-0003, ) in new stack
   == Spawn extension (company, h, 3) exited non-zero on 'SIP/4053-0003'

 Note that incoming calls via this PRI work correctly.

 Asterisk 11.0.1
 latest libpri and dahdi.

 Thanks,

 Vieri


Why don't your span numbers match?  1-4 but you have 3-6 in your .conf.

Have you tried to loop the ports or spans back to another port?

Set two ports for cpe and the other two for net, then crossover cable
to connect cpe to net.  Spans should come up and you should be able to
simulate the telco and test everything out in both directions.

Finally, call Digium and your telco if you are able to do the above
with no problems.

Thanks,
Steve Totaro

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Re: [asterisk-users] Hacked by Microsoft?

[Steve Totaro]

On Wed, Nov 28, 2012 at 7:45 PM, J Gao j...@veecall.com wrote:
 This morning someone tried to make sip call through my Asterisk. My server
 just drop these calls and record them in CDR with IP address:

 2012-11-28 06:30:51 SIP/216...  10001000 1000
 Hangup   999011972592249388 ANSWERED00:01   Hacker:
 168.63.67.239
 2.  2012-11-28 06:30:49 SIP/216...  10001000 1000
 Hangup   88011972592249388  ANSWERED00:01   Hacker:
 168.63.67.239
 3.  2012-11-28 06:30:46 SIP/216...  10001000 1000
 Answer   99011972592249388  ANSWERED00:02
 4.  2012-11-28 06:30:43 SIP/216...  10001000 1000
 Answer   1011972592249388   ANSWERED00:02
 5.  2012-11-28 06:30:39 SIP/216...  10001000 1000
 Hangup   2011972592249388   ANSWERED00:00   Hacker:
 168.63.67.239
 6.  2012-11-28 06:30:33 SIP/216...  10001000 1000
 Hangup   7011972592249388   ANSWERED00:01   Hacker:
 168.63.67.239
 7.  2012-11-28 06:30:30 SIP/216...  10001000 1000
 Answer   8011972592249388   ANSWERED00:03
 8.  2012-11-28 06:30:27 SIP/216...  10001000 1000
 Hangup   9011972592249388   ANSWERED00:06   Hacker:
 168.63.67.239
 9.  2012-11-28 06:30:25 SIP/216...  10001000 1000
 Answer   011972592249388   ANSWERED00:07

 Now I noticed something interesting: The hacker's IP address: 168.63.67.239

 whois gave me:
 NetRange:   168.61.0.0 - 168.63.255.255
 CIDR:   168.61.0.0/16, 168.62.0.0/15
 OriginAS:
 NetName:MSFT-EP
 NetHandle:  NET-168-61-0-0-1
 Parent: NET-168-0-0-0-0
 NetType:Direct Assignment
 RegDate:2011-06-22
 Updated:2012-10-16
 Ref:http://whois.arin.net/rest/net/NET-168-61-0-0-1

 OrgName:Microsoft Corp
 OrgId:  MSFT-Z
 Address:One Microsoft Way
 City:   Redmond
 StateProv:  WA
 PostalCode: 98052
 Country:US
 RegDate:2011-06-22
 Updated:2011-06-22
 Ref:http://whois.arin.net/rest/org/MSFT-Z


 hmmm Did I just hacked by Micro$oft?

 Gao


http://iplocation.truevue.org/168.63.67.239.html

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Re: [asterisk-users] Wireshark AMI Dissector

[Steve Totaro]

On Fri, Oct 26, 2012 at 2:52 AM, Olle E. Johansson o...@edvina.net wrote:

 23 okt 2012 kl. 22:31 skrev Kristian Kielhofner k...@kriskinc.com:

 Hello everyone,

  Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector?

  Decode as telnet and display filter telnet.data kind of work but TCP
 reassembly can't happen without a better understanding of the
 protocol...

 No, but that's a very cool idea. Would be great to have.

 Cheers,
 /O


Very cool.  Check this page.  It may be out of date but maybe you can
find more on the site.  I didn't dig into it yet.

https://github.com/adhearsion/adhearsion/wiki/AMI-Protocol-Notes

Thanks,
Steve T

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Re: [asterisk-users] high capacity analog - sip gateway

[Steve Totaro]

That is just silly.  You mean to say that the Adtran and the Adit
units are not as reliable as these new devices.  No way.

Get Adtrans, buy a four port T1 card or even better get the redfone
device and do HA Linux between to boxes, you have immediate failover.
http://www.red-fone.com/products-new/80.html

I seriously doubt any product on the market is as solid, tried, and
true as the traditional channel bank.

You can pickup these channel banks very cheap used, and often find
them in telco closets that have been abandoned.

Thanks,
Steve Totaro

On Thu, Oct 25, 2012 at 4:29 PM, jon pounder j...@inline.net wrote:
 On 10/25/2012 04:21 PM, Justin Killen wrote:

 just talking in general terms here I have found this sort of hardware is not
 the most reliable, and the more physical devices you spread it across the
 more fault tolerant you are of a single fault taking down a big chunk of
 your users.

 I wouldn't go more than a 24port device and for 100 users I would get 5 or 6
 of them depending on the exact numbers and have one as a hot spare that can
 just be swapped in quickly if one of the others dies.

 my analog stuff is all on spa or pap2t right now and I find that working
 out better for me than T1 card and channel bank was in the past, but the
 cabling is not as neat and tidy. Its a lot easier pill to swallow when 2
 extensions die than 24 for me.


 I’m looking for an fxs - sip gateway/router/switch for about 100 existing
 analog phones.  I’d like to get this done cheaply, but I want to make sure
 that whatever we buy works well with asterisk as well.  As far as I can
 tell, digium make no such device.  The only ones I’ve been able to find with
 a 48 port capacity are these two:



 Sangoma Vega 5000 50 FXS + 2 FXO Gateway
 (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs)

 Realtone WSS120 VoIP Gateway
 (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description)





 Does anyone have any experience with either of these products/vendors, or
 any suggestions for a different piece of hardware?



 Thanks

 -Justin



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Re: [asterisk-users] high capacity analog - sip gateway

[Steve Totaro]

On Thu, Oct 25, 2012 at 5:01 PM, Chris Bagnall
aster...@lists.minotaur.cc wrote:
 On 25/10/12 9:49 pm, Justin Killen wrote:

 What would be the advantage of using 100 single units vs. just buying VoIP
 phones?  That doesn't seem very cost effective to me in the long run.


 In older buildings with existing single pair cabling, there might not be a
 great deal of choice.

 We were faced with a similar scenario at a hotel in Lincolnshire a few
 months ago - listed building, lots of old pre-ethernet cable, no likelihood
 of being able to replace the cable.

 In answer to the OP, I concur with suggestions for 24-port channel banks -
 you really don't want one or two devices responsible for all 100 extensions.
 I would not encourage individual SPA or PAP units - it'd be an administative
 (and cabling) nightmare - it's bad enough with a dozen of the things.

 Kind regards,

 Chris
 --
 This email is made from 100% recycled electrons


Wow, single pair?  never came across that before.  You can run 10bastT
on CAT3.  The worst I have found where Amphenol cables, one per
station.  Not really bad since it is a 25 pair cable.

Thanks,
Steve T

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Re: [asterisk-users] How to tie orders taken to specific CDR records

[Steve Totaro]

On Thu, Oct 25, 2012 at 12:18 PM, Mitch Claborn mitch...@claborn.net wrote:
 Our phone operators work off of an Asterisk queue.  They take calls from
 customers and take orders with our back end systems.  What I need to be able
 to do is tie the orders taken to the specific CDR record that reflects the
 call from which the order originated.

 The typical/sample CDR table doesn't have a primary key.  I can add an
 auto-generated PK, but the CDR is not written until the call ends, when the
 orders have already been placed.  (Even if the CDR was written earlier,
 could I retrieve the generated PK from it in the dialplan somehow?)

 Is there some combination of fields in the CDR that might uniquely identify
 a specific call?

 Open to any and all ideas.


 --

 Mitch

I have done this with SIPCallID and sound files.  Not a true GUID but
enough combinations to not run into problems.

I have also used RDNIS to stuff info since it is mostly never used.
You can add DB functions to your dialplan beyond what the standard
queue logs contain or is written to the DB by the queue app.

http://www.voip-info.org/wiki/view/Asterisk+variables

Thanks,
Steve T

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Re: [asterisk-users] high capacity analog - sip gateway

[Steve Totaro]

On Thu, Oct 25, 2012 at 5:35 PM, jon pounder j...@inline.net wrote:
 On 10/25/2012 05:09 PM, Carlos Alvarez wrote:

 On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen
 jkil...@allamericanasphalt.com wrote:

 Cost and ease of deployment, yes.  At this specifc location we are
 currently using Centrex lines (ATT hosted) and are looking for a way to
 move into something cheaper without throwing away the existing phones.  I
 like the idea of using a channel bank – I’ll look into that as an option as
 well.


 You should be able to also connect the Centrex lines to the channel banks, I
 believe.


 Best to check the specs of the actual phones, around here some of them are
 norstar phones that I am pretty sure are some sort of isdn (bri) thing
 rather than being a pure analog device.  Better still take one of them and
 plug it in a raw analog line someplace and see what you get.


 I always advocate throwing out old analog phones as they will be a pain, but
 understand if you absolutely cannot.  Just keep in mind you can get a decent
 VoIP phone for $60 that is very likely to be nicer than what they have now
 and do much more.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


Very true.  If they have lots of lights and a display, they are most
likely digital phones.  What kind of PBX and phones do you have.
Before digital, phones needed 25 pair to control the phone's various
lights, lines, mwi.

Thanks,
Steve Totaro

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Re: [asterisk-users] high capacity analog - sip gateway

[Steve Totaro]

On Thu, Oct 25, 2012 at 5:09 PM, jon pounder j...@inline.net wrote:
 On 10/25/2012 05:01 PM, Steve Totaro wrote:

 That is just silly.  You mean to say that the Adtran and the Adit
 units are not as reliable as these new devices.  No way.


 I have had channel banks fail yes, and I stick by my assertion that failing
 a small $50 box is a lot less painful on the wallet and users than a channel
 bank with most of the extensions on it, this changes as the scale goes up
 though.

 I would only use a channel bank where the size can justify at least 3 of
 them, and I would never use a T1 based one again I would use the ethernet to
 FXS ones.

 I use a combination of analog and voip phones and there are various reasons
 for each being the type it is, one solution doesn't always fit everything,
 even within a single system.


Of course one solution doesn't fit everything but most likely would in
the case at bar (providing it is truly an analog system).

T1 cards are dirt cheap and channel banks are too.  They are also
modular, so if a card goes out, you lose 4 extensions.  If the chassis
goes out, then you lose all but it is all solid state.  I mean this
stuff is the mainstay of telephony.

A SIP FXS box will obviously have a substantial lower mean time
between failures.  Look it up.

What brand did you see fail and what failed?  I have only had cards
blow out and that is EXTREMELY rare.

Thanks,
Steve Totaro

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Re: [asterisk-users] Fully utilise all PRIs in a DAHDI group

[Steve Totaro]

On Wed, Oct 17, 2012 at 8:43 AM, Raj Mathur (राज माथुर) 
r...@linux-delhi.org wrote:

 Hi,

 Our client has DAHDI groups with 4 PRIs in each group (one 4-port
 interface per group), up to 6 groups per server.  When we dial, we can
 specify the group to be used for dialling, and our dial plan
 automatically distributes calls over multiple servers and multiple
 groups within a server.

 The way Asterisk dials by default is to use the lowest-numbered free
 line in a group to place a call.  This is technically fine.  However,
 what it means for our client is that the first couple of PRIs in a group
 tend to get the bulk of calls, the other two remain more-or-less
 unutilised.  This is a problem, since there are call commitments to the
 Telco for each PRI line.  The Telco tends to get all soggy and hard to
 light if some of the PRIs are used way below committed call levels.

 One solution is to group at the individual PRI level, so the load
 balancing automatically takes care of fair utilisation of each PRI.
 However, for various reasons we'd prefer not to do this.

 Another solution would be if Asterisk could choose a random (or LRU or
 LCU or round-robin or any other scheme) PRI within a group when
 dialling.  Any roughly fair way to distribute calls to PRIs within a
 DAHDI group would be fine.  Is there some way to achieve this?

 Asterisk 1.8.8 on Debian Squeeze.

 Regards,

 -- Raj
 --
 Raj Mathur  || r...@kandalaya.org   || GPG:
 http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
 It is the mind that moves   || http://schizoid.in   || D17F


Taken from the wiki searching with the exact terms you used.
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels

Thanks,
Steve Totaro
Dialing a GroupIn the Zap Channel Module's configuration file
(zapata.confhttp://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf),
you can define groups of Zap channels that get treated as a single channel
as far as the Dial
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dialcommand
is concerned. You specify which of four methods the Zap channel module is
to use to select a non-busy channel from the channel group by prefixing the
group number with one of the letters *g*, *G*, *r*, or *R*:


   - *g*: select the lowest-numbered non-busy Zap channel (aka. ascending
   sequential hunt group).
   - *G*: select the highest-numbered non-busy Zap channel (aka. descending
   sequential hunt group).
   - *r*: use a round-robin search, starting at the next highest channel
   than last time (aka. ascending rotary hunt group).
   - *R*: use a round-robin search, starting at the next lowest channel
   than last time (aka. descending rotary hunt group).


The round-robin searches make the Zap channel module start looking for an
available channel from a different channel number each time. For each
channel group, the Zap channel module keeps track of the last round-robin
start point, and this time starts checking availability from either the
next (lowercase *r*)) or the previous uppercase *R* channel in the group.
Which channel it actually finds available (if any) does not affect the
starting point for the next round-robin search. Calls to the
Dialhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Dial command
using ordinary (*g* or *G*) group selections do not affect future
round-robin starting points either.

For example, if you have defined channel group 2 as containing Zap channels
1, 2, 5 and 8, and the last round-robin search for this group (group 2)
began searching from channel 5, this is the order of searching that the Zap
channel module will use for the four possible selection methods:


   - Dial(Zap/g2...): Looks in order 1, 2, 5, 8
   - Dial(Zap/G2...): Looks in order 8, 5, 2, 1
   - Dial(Zap/r2...): Looks in order 8, 1, 2, 5
   - Dial(Zap/R2...): Looks in order 2, 1, 8, 5
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Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.

[Steve Totaro]

On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote:

 I was following Digium's instructions to the letter to install g729. but
 upon telling asterisk to load the module, the system hung.

 after a few minutes later a CTRL-C and attempted to run the command again.
 Same result. any g729 show command returns nothing... no error no results.

 Reboot the server and asterisk will not process calls. Freepbx shows the
 following.

 [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language dir
 fr for directory, not installed on system, skipping
 [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) -
 retreive_conf failed to get engine information and cannot configure up a
 softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
 [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) -
 [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not
 applied

 This seems to indicate that the g729 module is working

 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding
 module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc.
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied
 under a commercial license granted by Digium, Inc.
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full
 license text supplied by the accompanying
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: register utility, or
 ask for a copy from Digium.
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes
 software developed by the OpenSSL Project
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL
 Toolkit. (http://www.openssl.org/)
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006
 The OpenSSL Project

 [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
 action G729LicenseStatus
 [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
 action G729LicenseList
 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID:
 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05
 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license
 'G729-XXX' providing 40 channels
 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40
 G.729 licenses

 How do i roll this back? Just delete codec_g729a.so ?


You can do a noload in modules.conf.  This doesn't appear to be the problem
though.  It may be.  Did you try saving a change in FreePBX and applying it?

It seems more like a FreePBX config error that should be overwritten by
FreePBX database to flat files.

Thanks,
Steve Totaro
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Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.

[Steve Totaro]

On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro 
stot...@totarotechnologies.com wrote:



 On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote:

 I was following Digium's instructions to the letter to install g729. but
 upon telling asterisk to load the module, the system hung.

 after a few minutes later a CTRL-C and attempted to run the command
 again. Same result. any g729 show command returns nothing... no error no
 results.

 Reboot the server and asterisk will not process calls. Freepbx shows the
 following.

 [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language
 dir fr for directory, not installed on system, skipping
 [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) -
 retreive_conf failed to get engine information and cannot configure up a
 softwitch with out it. Error: ERROR-UNABLE-TO-PARSE
 [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) -
 [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not
 applied

 This seems to indicate that the g729 module is working

 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding
 module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc.
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied
 under a commercial license granted by Digium, Inc.
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full
 license text supplied by the accompanying
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: register utility, or
 ask for a copy from Digium.
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes
 software developed by the OpenSSL Project
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL
 Toolkit. (http://www.openssl.org/)
 [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006
 The OpenSSL Project

 [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
 action G729LicenseStatus
 [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered
 action G729LicenseList
 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID:
 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05
 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license
 'G729-XXX' providing 40 channels
 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40
 G.729 licenses

 How do i roll this back? Just delete codec_g729a.so ?


 You can do a noload in modules.conf.  This doesn't appear to be the
 problem though.  It may be.  Did you try saving a change in FreePBX and
 applying it?

 It seems more like a FreePBX config error that should be overwritten by
 FreePBX database to flat files.

 Thanks,
 Steve Totaro


See here
http://www.freepbx.org/forum/freepbx/users/apply-configuration-changes-errors-with-failed-to-get-engine-info-retreive-conf

Very similar problem with FreePBX, your G729 looks fine.

Check permissions and ownership of any files you changed.

Thanks,
Steve Totaro
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Re: [asterisk-users] Odd cracking with SIP-DAHDI

[Steve Totaro]

On Tue, Oct 16, 2012 at 1:31 PM, Richard Kenner ken...@gnat.com wrote:

 We recently set up a SIP trunk between an office in NY running Asterisk and
 an office in Paris (running Alcatel).  All works fine if a SIP phone on the
 NY system talks to the Paris PBX.  But if something on DAHDI (a PRI or
 MeetMe) talks to the Paris PBX, there's a low-volume crackling.  This isn't
 clipping because it also occurs when there's no legitimate sound.  It's
 sort of a mild version of what you used to get when a POTS pair had a
 ground short.  This occurs no matter what size originates the call.

 pings show round trip times of around 100ms, ranging from around 200 to 80
 ms.  Packet loss is zero.  The fact that SIP-SIP works fine suggests the
 issue isn't related to IP issues.

 I tried adding a jitter buffer, but that didn't make a difference.

 I've tried this sending just ULAW and G722 and allowing everything, but no
 difference.  The SDP that comes back from Paris doesn't list any audio
 codecs and is:

 v=0
 o=default 1350406175 1350406175 IN IP4 10.10.22.246
 s=Asterisk PBX 10.7.1
 c=IN IP4 10.10.22.246
 t=0 0
 m=audio 32000 RTP/AVP 0 101
 a=sendrecv
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=ptime:20
 a=maxptime:90
 m=video 0 RTP/AVP 31 34 34 98 99 104
 a=rtpmap:31 H261/9
 a=rtpmap:34 H263/9
 a=rtpmap:34 H263/9
 a=rtpmap:98 h263-1998/9
 a=rtpmap:99 H264/9
 a=rtpmap:104 MP4V-ES/9
 a=sendrecv

 Does anybody have any ideas as to what I should look at next?


cat proc/interrupts?

 http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards

Thanks,
Steve T
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Re: [asterisk-users] Odd cracking with SIP-DAHDI

[Steve Totaro]

On Tue, Oct 16, 2012 at 4:23 PM, Richard Kenner ken...@gnat.com wrote:

  I seem to recall seeing somewhere recently where there was a bugfix
  for ulaw/alaw conversion which would cause poor audio.

 Hmm.  You mean:

 https://issues.asterisk.org/jira/browse/ASTERISK-1323

 That was quite old, but that is what the noise sounds like.

  Have you tried updating your Asterisk to the latest of whatever
  major version you are running?

 I'm running 10.7.1, which is pretty new.  I'd prefer not to upgrade unless
 I know it'll fix it because of the work involved.


I would look for an answer and possibly ask your question on this site.
http://www.alcatelunleashed.com/search.php?keywords=SIP+trunking+with+Asterisk


Thanks,
Steve Totaro
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Re: [asterisk-users] SIP, Polycom, Asterisk - VPN

[Steve Totaro]

On Wed, Oct 3, 2012 at 10:22 AM, eherr email.eherr9...@gmail.com wrote:

 I am trying to configure the following scenario but have failed.

 ** **

 Currently, I have an Asterisk box sitting on a Static Public IP address in
 my office.

 ** **

 I have a remote office with 3 Polycom IP335s that are registering back to
 my local office’s publically address Asterisk box.

 ** **

 The remote office Polycom phones are getting IP information from an RV042
 and using the local ISP for internet access.

 ** **

 I want to set up a VPN on the remote side.

 ** **

 Has anyone done this? Does it make sense to do this?

 ** **

 Thanks,

 --E


Setup OpenVPN between the two sites.  A small solid state appliance can
handle this easily.  Don't worry about IAX2 as was suggested, SIP is just
fine.

I have used the WRT54G wireless router with one of the Linux firmwares.  I
have even run Asterisk on these little gems.

Some SNOM phones have a Linux/OpenVPN firmware and you can actually bridge
the WAN/LAN ports and use the phone as a gateway.

Thanks,
Steve Totaro
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Re: [asterisk-users] Who said asterisk is not to the task

[Steve Totaro]

On Sat, Sep 29, 2012 at 6:49 AM, Markus unive...@truemetal.org wrote:

 Am 29.09.2012 10:49, schrieb resea...@businesstz.com:

  [tz-ivr01 ~]# uptime
   11:00:32 up 776 days, 10:49,  3 users,  load average: 3.06, 3.05, 2.57
 Sharing is caring


 Is that a Quad Core CPU in your box?

 PS: Yes, Asterisk is great. :)


High load avg.  Is that Asterisk 1.2?
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Re: [asterisk-users] using analog phones

[Steve Totaro]

On Sun, Aug 19, 2012 at 8:37 PM, Noam Birnbaum
n...@maccentricsolutions.com wrote:
 Hi folks,

 A client wants to keep their old Inter-Tel KTS analog phones for budget
 reasons. Two questions:

 1. How could they use these with FreePBX?
 2. Would they be losing any features that they currently have with their
 analog PBX?

 Thanks!


 Noam Birnbaum
 Mac Daddy
 http://www.maccentricsolutions.com
 877.luv.macs x666
 tweet @noamb

 Tech support — 877.luv.macs or supp...@maccentricsolutions.com


I have done a good deal installations where analog handsets, headsets,
or many faxes just made sense.  I have used T1 cards to connect to 24
port channel banks to provide true analog dialtone.

I have also used SIP based channel banks with great success but never
bothered messing with the fax issues.

FreePBX should give you way more functionality than their current PBX,
sometimes it just takes a little creativity and doing things
differently.  Always get the details of what they currently use.  An
admin assistant or secretary is usually the person that knows what
bells and whistles everyone uses.  That is the person that you really
want to befriend for a smooth implementation.

Finally, I have installed a handful of Inter Tel/Mitel PBXen along the
way, and the phones I installed were infact digital and proprietary.
Check the model number of the phones to see if they are truly analog,
which I doubt.  In that case, you can still use a FreePBX box, but you
are going to have to integrate it with the Inter Tel, I usually put
Asterisk in front and just timeout your ring to VM a little bit
shorter on the Asterisk box than the Inter Tel.

With phone prices what they are, and a very good Polycom (or whatever
you like) SIP phone so cheap, I would seriously consider ditching the
phones if you find that they are digital.

Thanks,
Steve Totaro

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Re: [asterisk-users] Video conferencing?

[Steve Totaro]

On Thu, Jul 26, 2012 at 2:22 PM, Jonathan Rose jr...@digium.com wrote:

 Ken D'Ambrosio wrote:
  From: Ken D'Ambrosio k...@jots.org
  To: asterisk-users@lists.digium.com
  Sent: Wednesday, July 25, 2012 1:24:50 PM
  Subject: [asterisk-users] Video conferencing?
 
  Hi, all.  I'm 99% sure that Asterisk technically *supports*
  videoconferencing -- at least, as a conduit -- but are there products
  out there that leverage that?  I've been tasked with bringing
  videoconferencing internal to my company, and had been coming up
  empty
  looking for standalone solutions, when I suddenly realized that my
  favorite PBX software might be able to help out.
 
  Thanks much for any pointers you might be able to give me,
 
  -Ken

 I haven't used much in the way of video myself, but Asterisk 10 did some
 major overhauls to the confbridge application and I believe you can perform
 video teleconferencing with basically any SIP device/program that allows
 for a video stream, at least as long as it supports one of the video
 formats
 that Asterisk uses (h263 or h264). You could do some basic testing for this
 by running a few softphones connected to your Asterisk box. Jitsi does
 video
 calling as does Linphone... though I've personally had some trouble setting
 up h264 to work with Linphone. Good luck.

 --
 Jonathan R. Rose
 Digium, Inc. | Software Engineer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct +1 256 428 6139

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


I am deploying OpenMeetings on EC2 right now, you might want to check it
out.

Can be standalone or as I am doing, setting it up with Asterisk
https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html

Here are some useful videos and various howtos
https://www.google.com/#q=openmeetingshttps://www.google.com/#q=openmeetings+site:youtube.comsa=Xei=vfEWUJjAB-200QGI4oH4CAved=0CKgBENsBhl=enbav=on.2,or.r_gc.r_pw.r_cp.r_qf.,cf.osbfp=1f871a50839302aebiw=1280bih=576

Looks like it will be pretty cool.

Thanks,
Steve T
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Re: [asterisk-users] app_rpt

[Steve Totaro]

On Sat, Mar 10, 2012 at 11:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Fri, Mar 09, 2012 at 03:10:50PM -0600, Kevin P. Fleming wrote:
  On 03/09/2012 02:56 PM, Josh Freeman wrote:
  The most current patched Asterisk, along with the most current app_rpt,
  can be found at
  
  http://svn.ohnosec.org/svn/projects/allstar/astsrc-1.4.23-pre/trunk/
 
  I'm really trying to avoid fanning the flames here, but if that code
  is *really* based on 1.4.23, and hasn't been kept up to date with
  the Asterisk 1.4 releases, then that means it contains a number of
  security vulnerabilities that users should be aware of. Some of them
  are user enumeration vulnerabilities, but others (like AST-2011-010,
  AST-2011-005, AST-2011-001, and maybe more) are more serious.

 http://patch-tracker.debian.org/package/asterisk/1:1.4.21.2~dfsg-3+lenny5
 Or:

 http://anonscm.debian.org/viewvc/pkg-voip/asterisk/branches/lenny-security/debian/patches/

 Those are the patches for the Asterisk package in Debian 5.0 (Lenny). It
 is based on 1.4.21.2 (though with some extra patches: part of the
 bristuff patch). At least for a while I tried to check every security
 fix to see if it applies to Lenny.

 --
   Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir


I don't use Debian, but since this is a fork, the patches may break app_rpt
again like DAHDI did.

I may fire up a Debian Lenny VM and see if the fork with the patches match
up and work, and then if app_rpt and app_radio compile or throw an error.

The latest all in one ISO uses CentOS 5.7.

Thanks,
Steve Totaro
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Re: [asterisk-users] app_rpt

[Steve Totaro]

2012/3/9 Paul Belanger pabelan...@digium.com

 On 12-03-09 03:18 AM, Márkus Béla wrote:

 how can I add/enable app_rpt module to Asterisk 1.8?

  Make sure DAHDI is installed.  However, there is a patch on
 reviewboard[1] that will see this module be removed from asterisk.

 The code is out-dated and no longer maintained within asterisk.

 [1] 
 https://reviewboard.asterisk.**org/r/1764/https://reviewboard.asterisk.org/r/1764/
 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org


Here are details on 1.4 I have not done 1.8.

 Unfortunately, things are in somewhat of a mess.

There are major logistical hurdles with getting app_rpt code back into
the main Digium source tree. In addition, the latest versions of
asterisk have broken some of the code which app_rpt.c depends on, The
best thing to do at this point in time is to download the files.tar.gz
patched version of Asterisk from http://dl.allstarlink.org/installcd
and unpack it in /usr/src. Configure and compile zaptel, libpri, and
asterisk just like you would be downloading the sources from asterisk.org.

Once you have this version running, you can download the latest
app_rpt.c from:

http://svn.digium.com/view/asterisk/team/jdixon/chan_usbradio-1.4/apps

and install it in /usr/src/asterisk/apps and recompile asterisk.
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Re: [asterisk-users] app_rpt

[Steve Totaro]

On Fri, Mar 9, 2012 at 8:52 AM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:



 2012/3/9 Paul Belanger pabelan...@digium.com

 On 12-03-09 03:18 AM, Márkus Béla wrote:

 how can I add/enable app_rpt module to Asterisk 1.8?

  Make sure DAHDI is installed.  However, there is a patch on
 reviewboard[1] that will see this module be removed from asterisk.

 The code is out-dated and no longer maintained within asterisk.

 [1] 
 https://reviewboard.asterisk.**org/r/1764/https://reviewboard.asterisk.org/r/1764/
 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org


 Here are details on 1.4 I have not done 1.8.

  Unfortunately, things are in somewhat of a mess.

 There are major logistical hurdles with getting app_rpt code back into
 the main Digium source tree. In addition, the latest versions of
 asterisk have broken some of the code which app_rpt.c depends on, The
 best thing to do at this point in time is to download the files.tar.gz
 patched version of Asterisk from http://dl.allstarlink.org/installcd
 and unpack it in /usr/src. Configure and compile zaptel, libpri, and
 asterisk just like you would be downloading the sources from asterisk.org.

 Once you have this version running, you can download the latest
 app_rpt.c from:

 http://svn.digium.com/view/asterisk/team/jdixon/chan_usbradio-1.4/apps

 and install it in /usr/src/asterisk/apps and recompile asterisk.


There may be a working version of app_rpt.c for 1.8 in Jim Dixon's repo but
I doubt it.  Worth a look, I guess.

Not sure why you need 1.8 for a radio/repeater controller.

Thanks,
Steve T
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Re: [asterisk-users] app_rpt

[Steve Totaro]

On Fri, Mar 9, 2012 at 4:10 PM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 03/09/2012 02:56 PM, Josh Freeman wrote:

 The most current patched Asterisk, along with the most current app_rpt,
 can be found at

 http://svn.ohnosec.org/svn/**projects/allstar/astsrc-1.4.**23-pre/trunk/http://svn.ohnosec.org/svn/projects/allstar/astsrc-1.4.23-pre/trunk/


 I'm really trying to avoid fanning the flames here, but if that code is
 *really* based on 1.4.23, and hasn't been kept up to date with the Asterisk
 1.4 releases, then that means it contains a number of security
 vulnerabilities that users should be aware of. Some of them are user
 enumeration vulnerabilities, but others (like AST-2011-010, AST-2011-005,
 AST-2011-001, and maybe more) are more serious.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org



Kevin,

You are not fanning any flames, that is a good point and anyone that
deploys this technology should have to read a disclaimer as
to vulnerabilities.  I am well aware that there have been some serious
security issues in those earlier versions.

As for an Asterisk Box, or probably better described by what It is used
for, a Repeater or Base Station Controller Boxen, I have them locked down
in IPTables and in Asterisk.  There are usually not more then a dozen or so
RoIP conncted repeaters.

In my case, I only open one port for OpenVPN and I define the other
repeaters by host=IP.  As far as Soft Radios and Autopatch that function
is taken care of by a real Asterisk server that is more of a PBX and
faces the world, not the Repeater Controller, again, one entry defined by
IP over OpenVPN.  Bridged or routed, they non-routeable IPs.  The RoIP VPN
is only accessible through that tunnel, which is dedicated for that purpose.

I am very mindful of security, especially dealing with DoD, but pretty much
apply the same kind of security on any implementation.

Obviously, these security issues should be patched, but I feel that in my
implementations, things are very secure.

Thanks,
Steve T
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[asterisk-users] Fwd: Do you know how Asterisk came to be?

[Steve Totaro]

Apologies for the top post, something is screwed up with my email client,
will fix it soon.

What a BS story that I have debunked many times.  A used Key System could
be purchased for a few hundred dollars, a much better investment then
writing your own PBX from scratch.

A company that is supposed to provide linux services wasting it's time on
such a huge undertaking would go under very quickly.  Where is the revenue
stream?

Adtran was behind it from the start.  I have posted about this and it can
be found with other comments to provide more backing.  After making my
theory on the list, a former employee of digium who did actual hardware and
firmware  engineering verified my theory (anonymous for now, I don't
remember if he gave me permission to use his name but Mark will certainly
know M.P.)

http://web.archiveorange.com/archive/v/8Et0ZDHt1VEHkPySqlzs

Then this took place several years later which made things concrete in my
mind.

http://www.voip-news.com/feature/digium-asterisk-shuffle-adtran-013007/

There are a few articles that address my theory and back it up with
more evidence.

Also, check google maps for the offices of Digium and Adtran, unless they
moved, they were next door neighbors practically.

Shady stories like this cast a shadow over the reputation of Digium.  I
appreciate all that Digium has done but let's be realistic, the story is BS
and they are broadcasting it.  My duty is to call out BS for what it is.

Same deal with Vyatta and Cisco, just do some digging.

Thanks,
Steve T

-- Forwarded message --
From: Shea Caughron scaugh...@digium.com
Date: Thu, Mar 8, 2012 at 8:20 AM
Subject: Do you know how Asterisk came to be?
To: stot...@asteriskhelpdesk.com


**
View this email on your mobile device or
onlinehttp://app.email.digium.com/e/es?s=491e=26elq=816d47c4aa404f27ae6a0f05a7415092
http://www.asterisk.org/?link_id=headelq=816d47c4aa404f27ae6a0f05a7415092elqCampaignId=36
http://www.asterisk.org/?link_id=sideelq=816d47c4aa404f27ae6a0f05a7415092elqCampaignId=36Hello
Steve,

Origin 
storieshttp://app.email.digium.com/e/er?s=491lid=1190elq=816d47c4aa404f27ae6a0f05a7415092are
all the rage these days, and while perhaps the origin of Asterisk
isn’t
as exciting as the genesis of Wolverine, it’s still a pretty interesting
tale.

Way back in 1999, Mark Spencer had just started Linux Support Services
(LSS), an innovative small business that offered support for the Linux
operating system.  This was the height of the “Dot Com” era, and many
start-up businesses were taking advantage of the open source operating
system.  LSS took off, and as it grew, Mark found that he needed a phone
system.

Back in those days, phone systems were 100-percent proprietary.  They were
also expensive.  Not wanting to take out a loan for a phone system he would
probably outgrow in a matter of months, Mark decided to build his own PBX.
Unlike proprietary phone systems,* Mark’s solution was flexible software
that took advantage of the power (and price point) of Linux*. Mark named
the project “Asterisk,” a reference to the wildcard character.

Within a year, the Dot-com bubble popped and the demand for Linux support
dried up.  Fortunately for Mark, interest in his software PBX had
exploded.  Linux Support Services quickly pivoted to focus on the growing
demand for hardware and services related to Asterisk.  The groundswell of
interest in an open source telephony system grew into the Asterisk
Community with thousands of developers and users who pitched in, providing
patches, enhancements and valuable feedback. What started as a pragmatic
solution to a cash-flow problem, turned into a revolution.

By 2003, the business had been renamed “Digium” and was well on its way to
becoming the *world’s leading purveyor of telephony interface hardware*.

In the nearly 13 years since Mark released the initial Asterisk code, the
PBX market has undergone a massive shift.  *Open standards now rule* what
was once a proprietary market.  Expensive, limited proprietary PBX hardware
has given way to commodity computers running powerful software.  Digium has
grown from being a niche player to competing with the biggest names in the
PBX market.

So, there you have it.  That’s how it all started.  By the way, if you have
an interesting story about how
Asteriskhttp://www.asterisk.org/?elq=816d47c4aa404f27ae6a0f05a7415092elqCampaignId=36or
other open source software changed your life, we would love to hear
it.


 Shea Caughron
Digium, Inc. | Customer Development Manager
+1 256-428-6190 Check us out at
www.digium.comhttp://www.digium.com?elq=816d47c4aa404f27ae6a0f05a7415092elqCampaignId=36
www.asterisk.orghttp://www.asterisk.org?elq=816d47c4aa404f27ae6a0f05a7415092elqCampaignId=36
Follow 
http://app.email.digium.com/e/er?s=491lid=532elq=816d47c4aa404f27ae6a0f05a7415092
uhttp://app.email.digium.com/e/er?s=491lid=532elq=816d47c4aa404f27ae6a0f05a7415092

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote:

 On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
 stot...@asteriskhelpdesk.com wrote:
 
 

 [...]
  Yes, I have had no problems with Grandstream first gen ATAs, configured
 with
  server and credentials and shipped off, they just work.

 We use the HT-286, the server is on a public IP the nat setting on
 asterisk is set to yes and without port re-direction the ATAs have
 never connected from a private network, so I honestly find this SIP
 plug and play very hard to believe. But if it is true, then maybe you
 can actually help us figure out all the NAT issues we've had with SIP
 for the past 5 years. Perhaps, it is simply ignorance on our side and
 we have something fundamentally wrong in our set-up somewhere that may
 be have been causing these issues with NAT.

 Our set-up is fundamentally public and private Asterisk servers
 running on FreeBSD. Versions may vary from FBSD 7 thru 8.2 and
 Asterisk 1.4 and 1.6. We are planning to upgrade every server to FBSD
 8.2 and Asterisk 1.8 but we are in that process right now. Some
 Asterisk run in jails so I can understand the NAT issues there may be
 caused by the server itself. I honestly *love* your OpenVPN idea but I
 have to find a cheap ATA that could run as an OpenVPN client.

 Taking the simplest example a simple Asterisk 1.6 server on a public
 IP running on the base system (not in a jail):

 We run an operation that spans several countries including Canada, the
 US and the Latin American Andean region. As examples, with Canadian
 ISPs such as Rogers and Bell  we have always had to redirect the ports
 and use STUN server for the HT-286 to register to the Asterisk server.

 In the US we have the same problem with Comcast networks, so I don't
 understand how you say that you plug a Grandtream SIP ATA to a Comcast
 router and it just works. However, in a couple of NOLA countries the
 ISP's routers actually give public IPs, so if the SIP ATAs are
 connected directly to the ISP router, or in the DMZ then it just works
 as you say, BUT if the ATA is connected behind the firewall, or to a
 WiFi router, then we've _allways_  had to redirect ports. In every
 sigle customer we have had to send instructions on how to redirect
 ports, and of course to configure firewall if present.

 I just don't understand how you and other here say that a SIP ATA can
 just work. On the contrarty, with IAX2 using cheap AG-188N from
 Atcom they are just plug and play when shipped with a standard conf,
 and we have none of the quality issues you are referring to. We do
 have some call drops however, and some hangup problems but they don't
 affect our clients as much as having to deal with NAT issues.

 We may not run 15K extensions like you but I think we have a pretty
 good testing ground and have dealt with a fair share of NAT problems
 with SIP, that you and others here apparently don't have, so I am as
 amazed by your likeness of SIP as perhaps you are amazed as our
 likeness of IAX.


If you can post some SIP debug info from an ATA trying to register without
any redirection and also the relevant portions of your sip.conf, I am sure
I can help.

Do it from a new location with an el cheapo home router, Linksys WRTXXX.

If I cannot help you in a few emails, we can take this offline.

Actually paste your entire sip.conf in pastebin or something, as well as
sip debug.

Also the configs of your ATAs.

I think you have over-engineered to the point of creating problems.  This
is very common.  My philosophy is KISS

Thanks,
Steve T
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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 02/29/2012 08:22 AM, Alejandro Imass wrote:

  We use the HT-286, the server is on a public IP the nat setting on
 asterisk is set to yes and without port re-direction the ATAs have
 never connected from a private network, so I honestly find this SIP
 plug and play very hard to believe. But if it is true, then maybe you
 can actually help us figure out all the NAT issues we've had with SIP
 for the past 5 years. Perhaps, it is simply ignorance on our side and
 we have something fundamentally wrong in our set-up somewhere that may
 be have been causing these issues with NAT.


 The number of 'plain' SIP endpoints deployed behind consumer-grade NAT
 devices talking to Asterisk servers on public IP addresses is in the
 millions, if not the tens of millions. As has already been posted, Asterisk
 itself handles all the far-end NAT traversal duties necessary for this to
 work; neither the remote endpoint nor the NAT device need to do anything
 special, nor do they require any configuration.

 Rather than post a lengthy exposition on how widespread your network is
 and how technically astute your people are, you would probably accomplish
 much more to setup a simple test scenario as has been previously suggested,
 and if it does not work for you, post the details of the scenario and the
 failure here.


 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org


Agreed with one exception, the endpoint behind the NAT DOES need to be
setup correctly to keep the router from seeing inbound traffic to the
device as unsolicited and drop it.  That is a function of the router but
keep alives from Qualify on the Asterisk side, and setting the device to
register every few minutes will keep that mapping open and alive, letting
traffic pass as solicited.

Thanks,
Steve Totaro
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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote:

 On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
 stot...@asteriskhelpdesk.com wrote:
  Agreed with one exception, the endpoint behind the NAT DOES need to be
 setup
  correctly to keep the router from seeing inbound traffic to the device as
  unsolicited and drop it.  That is a function of the router but keep
 alives
  from Qualify on the Asterisk side, and setting the device to register
 every
  few minutes will keep that mapping open and alive, letting traffic pass
 as
  solicited.

 We use qualify=yes on Asterisk and a few months ago turned OFF the
 keep-alive feature on all SIP clients on our entire system.  This is
 working fine, and we did it because of a strange bug/behavior with
 certain versions of Cisco SPA series firmware.


 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


So you turned it off on the phones but use it on the Asterisk side?

Do you set a value or just use qualify=yes?

I had many problems with qualify over VSAT as ping times and jitter are
crazy.  700ms ping times were considered Good from the IZ in Iraq to
Equinix data center in VA, it took some tweaking to find the right value so
a phone that was Reachable was not labeled Unreachable, I did want
phones that were truly unreachable to be marked as such, more to spot
patterns and act on them or with the vendor.

Did you submit a bug report?  If it is easy to reproduce and you feel like
helping out, report it.  I do not report issues if there is a simple way to
do the same thing, but I know I should.

What does the debug or strange behavior look like?  Probably a variance in
the RFC implementation.

Thanks,
Steve T
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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Wed, Feb 29, 2012 at 1:26 PM, Alejandro Imass a...@p2ee.org wrote:

 On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com
 wrote:
  On 2012-02-29 15:25:49 +, Alejandro Imass said:
 
  We use SIP and IAX interchangeably, but had less hassle with IAX. The
  topic of the discussion on this thread was that SIP is so awesome and
  that IAX is a peice of crap.
 
 
  The original question (mine) was that my sound quality when using IAX was
  bad; with SIP the sound quality was great. Critically, I mentioned that I
  wanted to use IAX; I even said I was willing to do some self torture to
  get IAX working properly.
 

 Yeah, I wasn't referring particularly to the original post, just the
 way the thread turned against IAX like if it's not a viable solution
 and my point all along has been that for *us* IAX2 endpoints have
 worked better and easier to configure than SIP ones. Then it turned
 into a pissing contest, like you say, it happens in every list with
 the topic this or that.

 Again, as I pointed out to Steve above, and after reading all of your
 responses, our SIP/NAT woes seem obviously ignorance on our part, but
 that doesn't shadow the fact that IAX2 is working great for us with
 el-cheapo endpoints like Atcom's AG-188N and I would wish that many
 more manufacturers supported IAX2.

 We are happy with IAX and honestly never even had the need/curiosity
 to deal with the many SIP/NAT problems where sometimes it works great,
 and other times is a real pain in the ass that takes huge amounts of
 support to fix, and unhappy customers. On the other hand, IAX  took
 some engineering efforts at first, but the support issues are
 practically non-existent.

 --
 Alejandro Imass


I always posted that my view was based on experience.

My nieces and I made a viable home phone system out of strings and paper
cups

It is a real pain when you grow so large and then have to switch over to
SIP, might as well go with an Industry Standard then code that is and has
always been broken since it's inception.  You will find IAX2 trunking
issues dating back to 2005 and all sorts of IAX2 related problems since I
started way before Asterisk 1.0.  They have never got it right, SIP either,
but at least SIP is compliant enough to work just about all the time unless.

Try IAX, the predecessor of IAX2.

My alternator is currently not charging my battery enough for nightime
driving unless I turn off the radio and A/C.  It is fine without the extra
variables.  This is nothing new.

Knowing that when the demand rises, my battery will die and the vehicle
will falter and eventually stall means I am going to replace the
alternator.  Say I need my High Beams or to charge something via cig
lighter, I will end up stranded and need to take emergency action.

I could buy a used alternator, but I have no past experience with it and
have no idea how it will perform.

My choice of proper course of action is to put in something that is known
by all to work, maybe a bad unit, but backed by an immediate exchange.  I
will replace the battery and inspect other potential problem areas and
eliminate them as well.

Now I will have averted any problems down the road by doing it the right
way rather than hopping along on something that has been borken since day
one.

If you are going to do the job, do it right from the start so that you can
grow or change with ease and use real recognized standards.

If you are just playing around, do whatever.  Actually do whatever, and
learn the hard way, I don't care, just trying to help.

Thanks,
Steve Totaro
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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

People around here either hate me or love me.  I post experience and am
accused of bragging or whatever.  As a reader, I want to know who is giving
me advice and what it is based on.

$40k/wk of long distance through VoicePulse.  I have the invoices, that is
high usage, others attack me for posting information like this, I think I
know why but I don't care.

You have to have thick skin on these lists, the more technical, the more
you better have done your homework or get flamed.

It is from years of experience, not outsmarting anyone.  It took me months
to figure out that it just doesn't work well and as you can see, all of the
posts are very dated.  Nobody outsmarted anyone, just pure experience and
experience of MANY other people that use Asterisk.  Many did not wish to
make waves and emailed me directly that they either came to the same
conclusion or that they switched due to my suggesting and had no more
problems.

Digium and Digium FanBoys will argue that IAX2 is the best thing since
sliced bread.

Digium will ALWAYS tow the party line.  It was either Flemming or Lesher
that actually posted that it was in an official release so it couldn't have
bugs.  That was the end of listening to Digium about IAX2.  That statement
was archived with my reply of how ridiculous the statement was.  It is all
on the mailing list.

The compensation thing is very true, people drink the cool-aide about IAX2
and it sounds great.  Then it turns out that they go to production, and
audio sucks, customers are complaining.  It becomes a huge problem
obviously to an ITSP or any call center.

As I said, my experience is dated, but I have been one of the most prolific
people in the Asterisk community, I spoke at Astricon in 2007 on Large Call
Center Track and was the #1 poster for the year, a year or two ago.  I
predate most of Digium Staff.

I do this stuff in the real world, over VSAT or whatever connectivity you
can think of, my experience is real, not a developer in the world of code.

To answer your question, maybe you can spend time and get it to work
correctly, I have no idea, but why?

Why not just use SIP and be done with it.

Also realize that the dated posts have replies that are ridiculous like
VoicePulse is probably laying people off right now as we speak.

If a challenge drives you and you have tons of time to possibly never
figure it out and go to SIP, then by all means, do it.

If you want it to just work, use OpenVPN to get your single port, don't
believe the Digium party line and replies about using OpenSER or whatever
it is called now.  I get past the firewall and NAT issues with OpenVPN.

My standard now is Vyatta with NTOP, Asterisk, Webmin installed.  I only
use SIP and use OpenVPN.

I build Asterisk from source and menuconfig, I remove all that is not
needed, including IAX2.  I do download all the sound files in different
languages and codecs.

It just works.  I like things that just work.

Thanks,
Steve Totaro

On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com wrote:

 Ok Steve, obviously you’ve outsmarted at least this poster.  On the one
 hand, IAX2 has purchased things for you (won’t go as far as saying it
 bought your Mercedes), but on the other hand it is being dropped by
 providers as we speak. So are you saying it can be a good thing if you have
 the time and skill level to pursue it, but beginners should leave it alone?
 

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro
 *Sent:* Tuesday, February 28, 2012 3:59 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Same provider - IAX sounds bad, SIP
 sounds great

 ** **

 PSS

 ** **

 http://bit.ly/ywiwzt

 On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro 
 stot...@asteriskhelpdesk.com wrote:

 Google or click this link http://bit.ly/ywiwzteve  Steve Totaro IAX and
 then stop wasting your time,  go with SIP even if you need to create VPN
 tunnel(s).

 ** **

 Forget IAX2 and save yourself time you will never get back.

 ** **

 IAX2 has put tens of thousands of dollars in my pockets from the DoD, DoS,
 prime contractors to ITSPs around the world.

 ** **

 Thanks for IAX2 Digium!

 ** **

 Thanks,

 Steve Totaro

 ** **

 On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford ttelford.gro...@gmail.com
 wrote:

 I've tried turning jitterbuffer off - doesn't make a difference. (And why
 should it? The Jitterbuffer only applies to incoming calls, doesn't it?)**
 **



 On 2012-02-28 21:12:48 +, Noah Engelberth said:

 I'd try turning off the jitterbuffer and see if that makes things better.
  I just traced a similar call quality issue transferring calls incoming
 DAHDI on one * box to another * box, and turning off the jitterbuffer on
 the side that couldn't hear (in my case, the * box with the DAHDI lines,
 as the DAHDI callers couldn't hear the remote

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

Roger That, I am an IC.  I contract with the Government to little ten phone
shops.  From VA/MD/DC area, I have been contracted and flown in to many
large call center locations that were CONUS and OCONUS.

My facebook is Steve Totaro in Reston VA.  LinkedIN is more accurate, but
my resume speaks the truth.

Thanks,
Steve Totaro

On Tue, Feb 28, 2012 at 5:44 PM, Troy Telford ttelford.gro...@gmail.comwrote:

 On 2012-02-28 22:17:37 +, Danny Nicholas said:

  Ok Steve, obviously you've outsmarted at least this poster.  On the one
 hand, IAX2 has purchased things for you (won't go as far as saying it
 bought your Mercedes), but on the other hand it is being dropped by
 providers as we speak. So are you saying it can be a good thing if you have
 the time and skill level to pursue it, but beginners should leave it alone?


 I understood Steve to mean the following:
 - Secure locations like IAX. There's only one port to monitor or allow
 through a firewall, which is pretty compelling.
 - Aforementioned locations can't get IAX to work well.
 - So they hire Steve to get IAX to work properly, and he makes money.

 At least, that's my take.

 --
 Troy Telford



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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

And the dude arrives talking about penis..

On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez car...@televolve.comwrote:

 I have no interest in the penis-measurement competition firing up
 here, but I'll say that we have 100% abandoned IAX from all of our
 systems due to a myriad of issues.  These days it offers no real
 advantages in our opinion.


 On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro
 stot...@asteriskhelpdesk.com wrote:
  People around here either hate me or love me.  I post experience and am
  accused of bragging or whatever.  As a reader, I want to know who is
 giving
  me advice and what it is based on.
 
  $40k/wk of long distance through VoicePulse.  I have the invoices, that
 is
  high usage, others attack me for posting information like this, I think I
  know why but I don't care.
 
  You have to have thick skin on these lists, the more technical, the more
 you
  better have done your homework or get flamed.
 
  It is from years of experience, not outsmarting anyone.  It took me
 months
  to figure out that it just doesn't work well and as you can see, all of
 the
  posts are very dated.  Nobody outsmarted anyone, just pure experience and
  experience of MANY other people that use Asterisk.  Many did not wish to
  make waves and emailed me directly that they either came to the same
  conclusion or that they switched due to my suggesting and had no more
  problems.
 
  Digium and Digium FanBoys will argue that IAX2 is the best thing since
  sliced bread.
 
  Digium will ALWAYS tow the party line.  It was either Flemming or Lesher
  that actually posted that it was in an official release so it couldn't
 have
  bugs.  That was the end of listening to Digium about IAX2.  That
 statement
  was archived with my reply of how ridiculous the statement was.  It is
 all
  on the mailing list.
 
  The compensation thing is very true, people drink the cool-aide about
 IAX2
  and it sounds great.  Then it turns out that they go to production, and
  audio sucks, customers are complaining.  It becomes a huge problem
 obviously
  to an ITSP or any call center.
 
  As I said, my experience is dated, but I have been one of the most
 prolific
  people in the Asterisk community, I spoke at Astricon in 2007 on Large
 Call
  Center Track and was the #1 poster for the year, a year or two ago.  I
  predate most of Digium Staff.
 
  I do this stuff in the real world, over VSAT or whatever connectivity you
  can think of, my experience is real, not a developer in the world of
 code.
 
  To answer your question, maybe you can spend time and get it to work
  correctly, I have no idea, but why?
 
  Why not just use SIP and be done with it.
 
  Also realize that the dated posts have replies that are ridiculous like
  VoicePulse is probably laying people off right now as we speak.
 
  If a challenge drives you and you have tons of time to possibly never
 figure
  it out and go to SIP, then by all means, do it.
 
  If you want it to just work, use OpenVPN to get your single port, don't
  believe the Digium party line and replies about using OpenSER or
 whatever it
  is called now.  I get past the firewall and NAT issues with OpenVPN.
 
  My standard now is Vyatta with NTOP, Asterisk, Webmin installed.  I only
 use
  SIP and use OpenVPN.
 
  I build Asterisk from source and menuconfig, I remove all that is not
  needed, including IAX2.  I do download all the sound files in different
  languages and codecs.
 
  It just works.  I like things that just work.
 
  Thanks,
  Steve Totaro
 
  On Tue, Feb 28, 2012 at 5:17 PM, Danny Nicholas da...@debsinc.com
 wrote:
 
  Ok Steve, obviously you’ve outsmarted at least this poster.  On the one
  hand, IAX2 has purchased things for you (won’t go as far as saying it
 bought
  your Mercedes), but on the other hand it is being dropped by providers
 as we
  speak. So are you saying it can be a good thing if you have the time and
  skill level to pursue it, but beginners should leave it alone?
 
 
 
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
 Totaro
  Sent: Tuesday, February 28, 2012 3:59 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds
  great
 
 
 
  PSS
 
 
 
  http://bit.ly/ywiwzt
 
  On Tue, Feb 28, 2012 at 4:56 PM, Steve Totaro
  stot...@asteriskhelpdesk.com wrote:
 
  Google or click this link http://bit.ly/ywiwzteve  Steve Totaro IAX
 and
  then stop wasting your time,  go with SIP even if you need to create VPN
  tunnel(s).
 
 
 
  Forget IAX2 and save yourself time you will never get back.
 
 
 
  IAX2 has put tens of thousands of dollars in my pockets from the DoD,
 DoS,
  prime contractors to ITSPs around the world.
 
 
 
  Thanks for IAX2 Digium!
 
 
 
  Thanks,
 
  Steve Totaro
 
 
 
  On Tue, Feb 28, 2012 at 4:30 PM, Troy Telford 
 ttelford.gro...@gmail.com
  wrote:
 
  I've tried turning jitterbuffer

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

They said the same thing in 2005, 2008, now  Every release.

You never answered the question as to why you don't want to use SIP.  Is
there a reason, or do you just want to torture yourself?

Thanks,
Steve T

On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford ttelford.gro...@gmail.comwrote:

 On 2012-02-28 21:22:44 +, Kevin P. Fleming said:


 A serious bug with IAX2 trunking in recent versions of Asterisk (you did
 not mention what version you are using) was just resolved last week. You
 should test with 'trunk=no' to see if that is the cause of your problem;
 it seems very likely.


 For the record: 1.8.8.2~dfsg-1 (via Debian packages).

 I've tried trunk=no, and it might have made a difference (I'll have a
 better idea after some more testing.)
 --
 Troy Telford




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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

BTW, Trunking was the other selling point of IAX2 besides using 1 port
which is easily a DDOS target and also probably still
an implantation problem of using one thread and one proc for all calls.

Trunking allowed for less overhead then SIP since all the overhead for the
concurrent calls were combined into one stream.

Without trunking, you only have the single port thing.  It is quite easy to
open the correct ports for SIP, some just have GUI with a SIP checkbox,
IPTables is simple and there are tons of howtos.

Thanks,
Steve T

On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:

 They said the same thing in 2005, 2008, now  Every release.

 You never answered the question as to why you don't want to use SIP.  Is
 there a reason, or do you just want to torture yourself?

 Thanks,
 Steve T


 On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford 
 ttelford.gro...@gmail.comwrote:

 On 2012-02-28 21:22:44 +, Kevin P. Fleming said:


 A serious bug with IAX2 trunking in recent versions of Asterisk (you did
 not mention what version you are using) was just resolved last week. You
 should test with 'trunk=no' to see if that is the cause of your problem;
 it seems very likely.


 For the record: 1.8.8.2~dfsg-1 (via Debian packages).

 I've tried trunk=no, and it might have made a difference (I'll have a
 better idea after some more testing.)
 --
 Troy Telford




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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

Hey Alex,

Hope you are well.

Just a piece of advice.  Many or most people do not know the real
definition of ghetto and take it as a negative, poor, racial, black,
connotation.

Your vocabulary and and ability to articulate correctly can get you in
trouble sometimes.

Anyone that thinks that the word Ghetto means anything above, or a racial
slur should look up the true definition.

It isn't even an insult to the Asterisk Community.  By definition, the
Asterisk Community is an online Ghetto.

Just wanted to clear that up before someone tries to label you.

Thanks,
Steve T

On Tue, Feb 28, 2012 at 6:37 PM, Alex Balashov abalas...@evaristesys.comwrote:

 IAX is not supported or taken seriously outside the Asterisk ghetto, and
 that's good enough reason not to use it, IMHO.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

Wow Wikipedia was the only place that had the original meaning and not the
slur or slang meaning.

 A *ghetto* is a section of a city predominantly occupied by a group who
live there, especially because of social, economic, or legal issues. The
term was originally used in Venice http://en.wikipedia.org/wiki/Venice to
describe the area where Jews http://en.wikipedia.org/wiki/Jews were
compelled to live.

On Tue, Feb 28, 2012 at 6:55 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:

 Hey Alex,

 Hope you are well.

 Just a piece of advice.  Many or most people do not know the real
 definition of ghetto and take it as a negative, poor, racial, black,
 connotation.

 Your vocabulary and and ability to articulate correctly can get you in
 trouble sometimes.

 Anyone that thinks that the word Ghetto means anything above, or a
 racial slur should look up the true definition.

 It isn't even an insult to the Asterisk Community.  By definition, the
 Asterisk Community is an online Ghetto.

 Just wanted to clear that up before someone tries to label you.

 Thanks,
 Steve T

 On Tue, Feb 28, 2012 at 6:37 PM, Alex Balashov 
 abalas...@evaristesys.comwrote:

 IAX is not supported or taken seriously outside the Asterisk ghetto, and
 that's good enough reason not to use it, IMHO.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Tue, Feb 28, 2012 at 7:07 PM, Alejandro Imass a...@p2ee.org wrote:

 On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro
 stot...@asteriskhelpdesk.com wrote:

 [...]

  Without trunking, you only have the single port thing.  It is quite easy
 to

 Nope. The main reason _we_ use IAX is because it's easier for NAT


And it is easier for NAT because it uses one port as I stated, next


  open the correct ports for SIP, some just have GUI with a SIP checkbox,

 It may be true for you but it's certainly not the truth.

 - SIP requires redirection of ports if behind a NAT which is about 99%
 of home users, whether behind a WiFi router or an ISP private network.


Um, not when the server is on a public IP and the phones are configured
correctly.


 - SIP requires far more set-up and support effort and it's not a valid
 choice for a simple to use home-phone. (a) ISP routers change IPs
 frequently, (b) the router may change the ATA's private IP rendering
 the port redirection broken.


What about Magic Jack or Vonage?  The phone registers regularly with the
server so that negates everything above.

I don't do simple home setups, but they are simple home setups, your words,
not mine.  I have only had to redirect ports if the server is behind a NAT.
 Get a SNOM 370, flash with OpenVPN, run as a client and no problems, not
that there would be anyway.  I have placed 20 business phones behind NAT
with no special configuration and no issues but a bad phone or two in two
years

I have hostage negotiators with OpenVPN and a softphone on their laptops,
they travel the world and never have problems except maybe bandwidth.


 - A public SIP (w/o a VPN) requires careful control (e.g.
 contactpermit in Asterisk) to limit the IPs that can connect to the
 public box. Else you will get serous harm from things like SIPVicious
 attacks.


This can easily be mitigated by running on nonstandard ports.  Fail2Ban,
and a ton of other products can help, but yes, you are correct.  A
competent Admin is required to check logs daily and configure things
correctly.


 ISP change their IPs frequently so maintaining your user/ip
 list is almost impossible.


I use IP=dynamic with no problems but people tying to guess a password that
is the extension and MAC of the phone.  Dictionary attack is nothing.  With
a Gig pipe and fail2ban, no problems.

Also, I don't know where you live but I got Comcast@home when it first came
out and my IP has never changed.  ISPs in this area say dynamic but they
are static, at least the big two, Verizon and Comcast for home use.


 IAX2 was very vulnerable as well up to 2009
 but many things in this regard have changed and are much better.
 Granted, these security issues are common for both SIP and IAX2 but
 IMHO it's easier to manage with IAX.


Security was never really the issue if you read the thread.  It is about
voice quality.



 - In a NAT scenario SIP requires a couple of redirected ports per
 extension, which is a no-go for SMB installations requiring several
 ATAs without going to the extent of installing a more powerful
 equipment than a simple ATA.


Not in my experience, phone registers with server on public IP, no problems
except some obscure setting on a firewall.  Easy enough to google away.


 - You may use OpenVPN with SIP as you said but requires a PC which is
 not an option for a simple VoIP business that delivers something like
 Vonage, just plug it and it works.


Wrong, the SNOM 370 works great with OpenVPN.  You just contradicted
yourself as far as plug and play.

The SNOM 370 can also act as a bridge over the VPN tunnel using the LAN
port so the whole office is behind either split tunnel or direct VPN.

Any other SIP phone behind the SNOM with VPN bridging will also be on the
VPN as well as workstations.


 AFAIK there is no port redirection
 or any special configuration to use Vonage and it works almost on any
 network set-up (I don't use Vonage but know people that do). So if
 something like Vonage is using SIP it's probably using a VPN software
 like you recommend.


Magic Jack is pure SIP, no VPN


 Anyway, the point is that SIP and IAX2 have both pros and cons and I
 don't consider IAX2 to be a broken bat like you state. On the
 contrary, I think it works pretty well, and we use both SIP and IAX2
 targeted to simple Home, SOHO and SMBs that just want to plug it and
 work. We get that with IAX2 and not with SIP so from our experience is
 completely the opposite of what you say.


That is fine, I added disclaimers and small shops.  I deal in the 15,000
calls a day minimum realm, so we live in different worlds.  Two cups and
and a string work too


 --
 Alejandro Imass



 IAX2 is supported on cheap ATAs by several chineese companies and they
 work quite well.

  IPTables is simple and there are tons of howtos.
 
  Thanks,
  Steve T
 
 
  On Tue, Feb 28, 2012 at 6:29 PM, Steve Totaro 
 stot...@asteriskhelpdesk.com
  wrote:
 
  They said the same thing in 2005, 2008, now

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez car...@televolve.comwrote:

 On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass a...@p2ee.org wrote:
  works. This cannot be done with SIP and off the shelf cheap ATAs,
  period.

 We do it, so cannot seems to be a strong word.  It's not perfect,
 but our IAX problems outnumbered the SIP problems by at least double.
 Your mileage clearly varies.


Yes, I have had no problems with Grandstream first gen ATAs, configured
with server and credentials and shipped off, they just work.



  Also, respect netiquette and don't top post and use derogatory remarks
  and keep your discussion technical.

 Unfortunately, top-posting has become normal on this list.  I'm just
 fitting into how others were quoting the conversation.  If you find my
 remarks derogatory, I don't particularly care.


I follow the direction of the conversation.  If people are top posting,
then I follow suit, bottom, then I bottom post, inline as we are, then that
is what I do.  When in Rome



 --
 Carlos Alvarez
 TelEvolve
 602-889-3003

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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[Steve Totaro]

Roger That!

On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez car...@televolve.comwrote:

 On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass a...@p2ee.org wrote:
  Please expand as to how you set-up a SIP ATA behind a common home
  router set-up, without port redirection and/or use of a SIP proxy
  and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN
  support) it _cannot_ be done.

 Go buy a WRT-54G or nearly any consumer-class router and just plug in
 a SIP device.  Done.  It works.  We *never* work on customer routers
 and very rarely have to tell them to reconfigure their router at all.
 My own home configuration is an Airport Extreme with zero
 configuration.  So either these are very old routers you're having a
 problem with, or buggy SIP devices, or something else.

  You should care.

 Hmm, let me check the reading...

 http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png

  don't drive a Yugo but if I did I could easily be offended by the
  pejorative use of the brand.

 It's a piece of junk and everyone knows it, including the owners, so who
 cares?

 --
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Re: [asterisk-users] [asterisk-dev] SIP, NAT, security concerns, oh my!

[Steve Totaro]

On Sun, Oct 23, 2011 at 10:06 PM, Andrew Latham lath...@gmail.com wrote:
 OpenVPN is the solution to all NAT issues.  With at least the SNOM 370
 supporting it and the phone can be setup as a OpenVPN gateway as well,
 for very small offices, it is a great phone.  Linux based and you
 install their OpenVPN firmware and then setup the PC port on the phone
 to bridge (that is what I do anyways) traffic, plug it into a switch,
 configure whatever, no split tunnel, and completely secure site to
 site VPN and no NAT issues, I would do this with something like a five
 workstation office at most.

 Alot of bang for the Buck with the SNOM 370.  If you already know
 OpenVPN it is a breeze, and there are tons of howtos specific to the
 SNOM, documentation is good too.

 This could also be done with any number of other solutions from the
 WRT54GS whatever, or just a little boxen for VoIP over the VPN tunnel,
 and other traffic out the default gateway.  I just like to secure
 small remote sites so I can monitor, administer, and enforce network
 usage policy.  That is coming from a Private Military Company
 background.  I don't want any data not going through a voice or data
 tunnel to Equinix.  Then some small Top Secret installation in a
 remote area doesn't wind up infecting their little LAN.

 Set it up and put it in a fly-away quarter sized rugged rack with
 casters.  This approach has saved days and days of troubleshooting
 with people who cannot understand me by language or technology or
 whatever.  It took a bit of work to plan the whole thing out, mesh the
 systems to route over the tunnel with fault tolerance, but certainly a
 worth the time.

 Short, OpenVPN can get you around all SIP/NAT/Security issues, since
 the tunnel is on a singe port, the big idea behind IAX2 but much
 better, it is still SIP.

 You can lock down everything using OpenVPN to prevent problems and
 allow simple management of global networks.  All traffic passes
 through a few devices, giving you almost total security at a few key
 points.

 Vyatta paid version in a VM or Bare Metal is my internet facing
 firewall.  It is so powerful, cheap, and the dev team there is great.
 They have helped me directly a number of times.

 I like to have NTOP, Webmin and Asterisk on most of these boxen, but I
 don't want to install a bunch of extra junk beyond the Vyatta ISO and
 the packages I find handy.

 That is my approach until IPV6 ever come out, or some other variant.

 Thanks,
 Steve Totaro

 Thanks,
 Steve Totaro

 I use a lot of Zentyal for OpenVPN plus networking fun.  I did hear
 from a snom engineer that they got the openvpn working with a limited
 functionality on the snom 300 and other models. Direct to you email
 because I wanted to mention your double signature...

 --
 ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~


Andrew,

I have one client setup to add the signature and the others are
manual.  On occasion, there is a double signature.  Thanks for
pointing it out, but content of the post is issue.  My signature is
extremely minimal, not spammy like many.

I don't know how you can have Limited Functionality with OpenVPN.
Not sure who you talked to or when, but it works great.

Read some of the howtos and see that you can use one phone to create a
site to site VPN using bridge-utils.

For real networking fun, I would not use a phone, the OpenVPN is just
for a what I said, one off sites or totally mobile hard phones.

Vyatta has training and 24/7 support for $1k per server.  The project
you pointed out looks cool from a couple of screenshots, I will load
it up on VMWare if there is an image.

Vyatta paid version is so cheap, a great business plan, the backing of
former Cisco execs, and is very robust.  GUI needs some work, that is
why I put NTOP and Webmin on it, but their engineer, the main guy, an
exec, and myself have had conference calls about additional
functionality.  They don't want to incorporate and will not support
other projects and are working on their own GUI, I totally understand.

I just ask for the tools to build from source and not mess anything
up from Vyatta's and my viewpoint.  Beyond that, I completely
understand where their demarc is from stock software to whatever I
build.

Just to point out that Vyatta is Sand Scrit and means Open, I
thought that was cool.  They are very similar to Asterisk, at least as
far as having a commercial and open source offering.  I see Vyatta
staying the course, having legs, and not going the way of vapor.

Many places are using it now, I was surprised by some of the fortune
500s, and the job descriptions I get with Vyatta listed along with
Cisco.

Thanks,
Steve Totaro

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[asterisk-users] **OT** Fwd: oFono 1.0 has been released

[Steve Totaro]

When oFono launched, I announced the project to other projects that it
may compliment.

oFono has hit the 1.0 Milestone and has some serious backing if you
missed my post a year or so ago and never heard of it.

Check it out...

Thanks,
Steve Totaro

-- Forwarded message --
From: Marcel Holtmann mar...@holtmann.org
Date: Sun, Oct 16, 2011 at 2:25 PM
Subject: oFono 1.0 has been released
To: of...@ofono.org


Hello everybody,

I am pleased to announce that we have released oFono 1.0 this week. This
marks a major step for oFono and we consider it fully feature complete
for 2G and 3G telephony.

oFono is released under GPL version 2 and is 100% open source. It
includes support for the majority of data modem vendors and also full
voice telephony support for Infineon (now IMC), ST-Ericsson, Nokia/ISI
and also Calypso/Freerunner.

All standard features including voice calls, supplementary services,
text messaging, USSD, SIM Toolkit, network registration, multiple GPRS
contexts and many more have been integrated with easy to use D-Bus APIs.

With our 1.0 out of the door, we will continue to improve our CDMA
support and also integrate LTE into oFono. So stay tuned for new
features.

The tarballs are not yet available due to the security breach on
kernel.org, but will be uploaded as soon as service has been restored.

Regards

Marcel


___
ofono mailing list
of...@ofono.org
http://lists.ofono.org/listinfo/ofono

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Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?

[Steve Totaro]

Going with the flow on top posting.

You just need an SBC
http://www.voip-info.org/wiki/view/Session+Border+Controller setup
correctly.

Thanks,
Steve T

On Sun, Oct 2, 2011 at 3:14 AM, Sam Govind govoi...@gmail.com wrote:
 Hey,
 Why do you think using OpenSIPs is not going to work for you ? You can
 always add SIP trunks on openSips and based upon which trunks getting the
 call you can LB or/and FO to as many asterisk servers as you want !
 Regards,
 -Sammy

 On Sun, Oct 2, 2011 at 7:12 AM, Michelle Dupuis mdup...@ocg.ca wrote:

 If one server is supposed to carry the full load of the other during
 failure, then you have to size each server to handle  100% load - so load
 balancing is pointless.

 Checkout haast at www.generationd.com and read  the docs on how it does
 failover...certainly good for ideas.

 
 From: asterisk-users-boun...@lists.digium.com
 [asterisk-users-boun...@lists.digium.com] On Behalf Of Tobias Steen
 [tobias.st...@s2.se]
 Sent: Saturday, October 01, 2011 6:30 PM
 To: Asterisk Users List
 Subject: [asterisk-users] Make asterisk cluster appear and operate as a
 single server?

 Hi,



 I'm trying to plan a system of clustered asterisk machines where a number
 of SIP trunks will be hosted on the platform. Each trunk will be hosted for
 a specific customer who owns it and therefore payment is handled directly
 between the customers and their trunk-providers, each trunk will have about
 50-200 simultaneous calls.



 No SIP phones will be directly connected to the platform, my thought is
 that the asterisk machines should only receive incoming and make outgoing
 calls through the trunks, and then connect the calls with each other.



 To make this scalable and have the option of running an infinite number of
 sip-trunks, I need a good way to load-balance my asterisk servers and
 implement failover support and also be able to add / replace the machines in
 the cluster in a safe and reliable way.



 I'm have some experience building single asterisk solutions but I have
 never worked with load balancing of multiple asterisk machines.



 Is it possible to configure all trunks on a single asterisk setup which is
 then reflected over a cluster of asterisk machines? If I have a cluster of
 machines, I guess I need some kind of front-end application / system? I will
 then also need to be able to connect calls between the machines, the calls
 to be connected with each other will always be incoming and outgoing on the
 same trunk.



 In other words, I want to create a large cluster of asterisk machines to
 appear and operate as a single asterisk server.



 I've looked at projects like OpenSIP but it feels like this is not really
 what I need?



 I really appreciate if someone can help me get on the correct path here, I
 need all the feedback I can get.





 Thanks in advance!





 Best regards

 Tobias

\


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Re: [asterisk-users] Fax from FXS to PRI

[Steve Totaro]

On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett adamli...@plexicomm.netwrote:

 If I have a 4 port Digium FXS card and a single port PRI card on the same
 asterisk box, is it expected that I'd be able to plug a fax machine into the
 analog FXS port and have no problems sending or receiving faxes?  Our
 connection to the Telco is on the PRI obviously.

 I don't recall the specific card models that we have, but I can check if it
 matters.

 Does the version of asterisk or Zaptel matter?

 My related question is this: In the scenario described above does the audio
 pass directly from one card to the other through the PCI bus or does it have
 to somehow be processed by software?


Nobody can say for sure.  It is not a supported configuration.  I can tell
you that I have had great success and wasted days messing around with this
configuration.

It is usually the other side's fax machine, a cheap all in-one, or it is TX
and RX gains, or IRQs, or..

Questions to ask are
1.  Is this for your system or are you installing for someone else?  You
could look very bad if the proper expectations are not set.  It may take a
great deal of trial and error to get to an acceptable level, if you can even
do that based on need.

2.  Needs, if fax is part of the lifeblood, then this route may not be the
best.  If it doesn't hurt to ask someone to resend or whatever, then go for
it.

Just remember the gotchas, IRQs, TX RX gain settings, echo can when bridged
=no.

Again, it has never been a supported configuration by Digium, and everyone
that has dealt with faxing in Asterisk especially on different systems will
tell you that you won't know until you try.  And even then, is it worth days
of your time trying to get it as close to a POTS line as possible?

Another issue I have run into are the Digium FXS daughter boards getting
fried somehow.  I punch on a 66 block now and put on surge protection after
frying six modules in as many years.  Something like this
http://www.digitaltele.com/ProductInfo.aspx?productid=HCO

I have never had to do that on the FXS side but now it is just standard for
all single pairs I do.  I will know in the next year or two if it helps.  No
idea what is frying the daughterboards.  Must be the fax machine.

Thanks,
Steve T

Thanks,
Steve T
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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Steve Totaro]

See comments inline.

On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote:

 I'm about to start building my asterisk server and I can't seem to find
 anything that discusses the pros and cons of installing the OS (Fedora 15)
 as console only or GUI, ie install KDE as well.


If you want an OS that is going to be supported a year from now, don't use
Fedora.

Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much
beta RHEL.  It's EOL is one year from my understanding.

You want to install the very minimum as most people would agree, why do you
think you need a GUI.

Best practice is to only install the bare minimum on a server.


 So, other than a bit of disk space, is there any reason why I shouldn't
 install KDE when I set it up ?


It has and will cause issues.  I have installed KDE or whatever but booted
to init 3 for a couple of machines.  I could go to init 5 if I had to, but I
never did had to.  I don't see a single pro, but there are many cons.

What benefit do you get from KDE?  Why do you want it.  Is this just going
to be an asterisk server or a desktop?



 Is there any great disadvantage to running the server in init level 5 (ie
 KDE, xorg, etc) running in the background, but not being logged in, versus
 init level 3 ? (Or whatever they call these things these days..., ie F15
 uses systemd...)

 FWIW, my server hardware will sit on a server rack in the utility room.  I
 might drag a display and keyboard down there once in a while to troubleshoot
 and/or do maintenance, but mostly I'd ssh in and probably use a remote
 desktop app to work on it.


How does remote desktop help you over an SSH CLI?


 FWIW, I'm OK doing things via the CLI, but sometimes its really nice to
 have graphical tools.


Ok, I can understand, I used to be like this for a while.  I am a huge fan
of Webmin for a GUI.  It allows for almost everything and for me, it is
better than KDE or anything else.  It is just a webpage with tools
attached.  No big potential problem there.


 I look forward to your input.

 Thanks

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Steve Totaro]

On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:

 See comments inline.

 On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote:

 I'm about to start building my asterisk server and I can't seem to find
 anything that discusses the pros and cons of installing the OS (Fedora 15)
 as console only or GUI, ie install KDE as well.


 If you want an OS that is going to be supported a year from now, don't use
 Fedora.

 Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty
 much beta RHEL.  It's EOL is one year from my understanding.

 You want to install the very minimum as most people would agree, why do you
 think you need a GUI.

 Best practice is to only install the bare minimum on a server.


 So, other than a bit of disk space, is there any reason why I shouldn't
 install KDE when I set it up ?


 It has and will cause issues.  I have installed KDE or whatever but booted
 to init 3 for a couple of machines.  I could go to init 5 if I had to, but I
 never did had to.  I don't see a single pro, but there are many cons.

 What benefit do you get from KDE?  Why do you want it.  Is this just going
 to be an asterisk server or a desktop?



 Is there any great disadvantage to running the server in init level 5 (ie
 KDE, xorg, etc) running in the background, but not being logged in, versus
 init level 3 ? (Or whatever they call these things these days..., ie F15
 uses systemd...)

 FWIW, my server hardware will sit on a server rack in the utility room.  I
 might drag a display and keyboard down there once in a while to troubleshoot
 and/or do maintenance, but mostly I'd ssh in and probably use a remote
 desktop app to work on it.


 How does remote desktop help you over an SSH CLI?


 FWIW, I'm OK doing things via the CLI, but sometimes its really nice to
 have graphical tools.


 Ok, I can understand, I used to be like this for a while.  I am a huge fan
 of Webmin for a GUI.  It allows for almost everything and for me, it is
 better than KDE or anything else.  It is just a webpage with tools
 attached.  No big potential problem there.


 I look forward to your input.

 Thanks


I have been using Vyatta (paid for with phone support.)

It makes for the most powerful Asterisk platform you can imagine.  There is
a learning curve but I love what I have put together.  There are howtos
everywhere and if you buy licenses, you get excellent support and online
training courses.

It is a very firewall/Router.  It handles everything from OpenVPN, awesome
security features, IPS, and even QoS, wireshark.

I put webmin and NTOP on these machines as well.  Vyatta has become my new
platform for Asterisk.

Check it out http://www.vyatta.org/documentation

There is very little you cannot do, but don't have to use the features if
you don't want to.

Vyatta is also a company like Asterisk.  Vyatta is the baby of former
bigtime corporate Cisco guys.  Asterisk is the baby of former Adtran execs.

Thanks,
Steve T

Thanks,
Steve T
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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[Steve Totaro]

I gu

On Thu, Aug 25, 2011 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Thu, Aug 25, 2011 at 12:39:14AM -0400, Steve Totaro wrote:

  I used the Asteirsk System() app to call lynx with a special URL.  The
 URL
  contains all the authentication, recipient, and SMS body.  Calling that
 URL
  via System(), as I said, I like lynx, causes an SMS to be sent.  Kannel
 is
  extremely customizable.

 Slightly off-topic: why not use CURL()?

 --
   Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir


Because my first inplementation was before Curl() was part of Asterisk.

I am very familiar with Lynx and it wjust works.  Never an issue.

Don't enable or use apps on an Asterisk system if there is another, reliable
app that can be used.

I keep Asterisk's role to a minimum.  I only load the apps that are needed
for the implementation.  I usually build them all, but, either I do a noload
or rename the .so.

I also try to put other functions of different machines, to segregate the
mission critical, or at least the as much Asterisk from other features.
 Databases, fast-agi, HylaFax, recording calls, and whatever else.

It is just the way I do things, budget providing of course.  I want the core
being as stripped down OS, apps, and Asterisk as possible.

Thanks,
Steve T
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Re: [asterisk-users] SIP client on a mobile?

[Steve Totaro]

On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote:

 Steve Totaro wrote:

  Just use a SIP client on your phone.  Many providers have multiple
  failover paths for inbound calls.
 
  This thread morphed from a nice home phone system into something
  completely different.

 Yup.

   For my situation, DISA is pointless except for road warriors who
   call all over the world, from anywhere, they can call into the corp
   system, get dialtone and skip the whole process of expense reports
   for work
   related calls.  It makes things less complex, not more.
 
  Using DISA also means getting a corp caller id, not a mobile.
 
  Yes, spoofing provides that.
 
 
   Maybe if you explain your situation and how your plan works, but
   for me, personally, DISA would be a an added cost and complication.
  
   The only purpose I can think of for myself could be accomplished by
   spoofing caller id.
 
  How is that done from a mobile?  Sofar that has been my main reason
  for using DISA - cost is not a real issue.
 
  SIP client.  Spoof card, yes it is DISA, but you don't have to do
  anything but use the card.

 Steve, even if I could get SIP clients for our phones, doesn't this mean
 using a data connection rather than just voice?  That would make it a
 lot pricier than the current setup with DISA (which is largely free).


 /Per Jessen, Zürich


A Wifi connection?  I guess that wifi is not like it is here.  I can get on
highspeed wifi anywhere I go in the DC Metro area for free.  Just driving
around, there is always an open access point.  When driving around, I pick
up thousands of APs in a couple miles and don't have any protection at all.

I would suspect that most road warriors have high speed data needs?  Not
sure what business you are in, but having fast internet (relatively
speaking) is a must to do work.  I am not saying to use the data supplied
from phone, if that is what you are thinking.

If your phones don't have SIP, then use callback.  You call your company, go
through whatever you seutp in the dialplan, and the phone system calls you
back as well as calling the other party.

You edited out much of the context of the conversation to support your
side.  I don't play games like that...

SIP client on the phone was an option.  Was the original question about
using DISA to save money?  Yes it was.  Now you are stating that it is
largely free.

Callback is a great solution when outbound cell phone calls quite a bit more
than your cutrate VoIP provider.  As I said, many countries do not charge
for inbound calls.

I am still clueless what your point is/was but if it is almost free then,
stick with it.  Still clueless why you posted if it almost free.

Thanks,
Steve Totaro
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Re: [asterisk-users] SIP client on a mobile?

[Steve Totaro]

On Thu, Aug 25, 2011 at 3:06 AM, Steve Totaro 
stot...@totarotechnologies.com wrote:



 On Thu, Aug 25, 2011 at 2:27 AM, Per Jessen p...@computer.org wrote:

 Steve Totaro wrote:

  Just use a SIP client on your phone.  Many providers have multiple
  failover paths for inbound calls.
 
  This thread morphed from a nice home phone system into something
  completely different.

 Yup.

   For my situation, DISA is pointless except for road warriors who
   call all over the world, from anywhere, they can call into the corp
   system, get dialtone and skip the whole process of expense reports
   for work
   related calls.  It makes things less complex, not more.
 
  Using DISA also means getting a corp caller id, not a mobile.
 
  Yes, spoofing provides that.
 
 
   Maybe if you explain your situation and how your plan works, but
   for me, personally, DISA would be a an added cost and complication.
  
   The only purpose I can think of for myself could be accomplished by
   spoofing caller id.
 
  How is that done from a mobile?  Sofar that has been my main reason
  for using DISA - cost is not a real issue.
 
  SIP client.  Spoof card, yes it is DISA, but you don't have to do
  anything but use the card.

 Steve, even if I could get SIP clients for our phones, doesn't this mean
 using a data connection rather than just voice?  That would make it a
 lot pricier than the current setup with DISA (which is largely free).


 /Per Jessen, Zürich


 A Wifi connection?  I guess that wifi is not like it is here.  I can get on
 highspeed wifi anywhere I go in the DC Metro area for free.  Just driving
 around, there is always an open access point.  When driving around, I pick
 up thousands of APs in a couple miles and don't have any protection at all.

 I would suspect that most road warriors have high speed data needs?  Not
 sure what business you are in, but having fast internet (relatively
 speaking) is a must to do work.  I am not saying to use the data supplied
 from phone, if that is what you are thinking.

 If your phones don't have SIP, then use callback.  You call your company,
 go through whatever you seutp in the dialplan, and the phone system calls
 you back as well as calling the other party.

 You edited out much of the context of the conversation to support your
 side.  I don't play games like that...

 SIP client on the phone was an option.  Was the original question about
 using DISA to save money?  Yes it was.  Now you are stating that it is
 largely free.

 Callback is a great solution when outbound cell phone calls quite a bit
 more than your cutrate VoIP provider.  As I said, many countries do not
 charge for inbound calls.

 I am still clueless what your point is/was but if it is almost free then,
 stick with it.  Still clueless why you posted if it almost free.

 Thanks,
 Steve Totaro


I am not sure why people try to prove me wrong, but they do.  On rare
occasions, I am wrong, I am also big enough to admit it.

To answer your question, and get on the same terms, VoIP (or data as you
prefer) would probably be cheaper.  Isn't that the whole reason behind
VoIP?  You say voice, does that mean your provider's voice service?

Depending on the cost of inbound and out abound calls on a cell are the key
here.

Is it next to nothing to call a foreign country from your cell?  Is it much
more expensive than rates at the office.  Generally, I think outbound calls
from an office are much lower than cell phone charges.

I was paying a $40k plus weekly for long distance calls from Iraq to mostly
Fiji, Uganda, Peru.  That was with VoicePulse, all 703 DIDs around the
world.  Voicepulse gave me great rates because $40k a week is not chump
change.  I wonder what the cost of cell phone calls would amount to?

My international rates for outbound cell phone calls are beyond a rip-off.

Thanks,
Steve Totaro
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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[Steve Totaro]

So in other worlds you had nothing to contribute to this thread.

On Thu, Aug 25, 2011 at 2:44 AM, Per Jessen p...@computer.org wrote:

 Steve Totaro wrote:

  VoIP mostly aside, a couple more thoughts.
 
  I am not sure I understand your reasoning for DISA or how it is
  cheaper.

 The only reason we use DISA is to spoof the caller id.  The OP also
 wanted to save costs, which is also possible (as someone already
 confirmed).  DISA does save some cost for me too, but it is immaterial.

 The call from the mobile to the asterisk box is free or flat fee due to
 calling groups offered by our provider.  The outgoing call is charged
 at regular fixnet prices, much cheaper than mobile ditto.

  You can buy a card that accepts SIMs as FXO and FXS.
  For your reasoning, a card of such nature is required.  Populate  it
  with different SIMs or whatever that are in calling groups or whatever
  you were trying to say.

 You've lost me, I have no idea what you're talking about.

  Just use callback back and some logic to reduce your costs.
  Call back will allow you to use the corp identity, and  LCR will cut
  costs over DISA.
 
  The system calls you back after you make a call.  Then the call is
  placed. There is a very brief outbound cell phone call, followed by a
  an inbound call from the server that you initiated with call back.

 OK, I see.  I haven't looked at that, but it sounds more complicated
 than using DISA, and I'm not convinced it would be any cheaper.  (it's
 important that the scheme be easy to use from the mobile end).

  Inbound to a cell is generally less expensive that oubound on a cell,
  sometimes completely free.

 Yes, inbound to a mobile is free as long as you're not roaming. However,
 with our calling group setup, it doesn't matter who (fix or mobile)
 originates the call, the cost is the same.


 /Per Jessen, Zürich

 --
 http://www.spamchek.com/ - your spam is our business.


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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[Steve Totaro]

On Thu, Aug 25, 2011 at 4:36 AM, Skyler skchopper...@gmail.com wrote:

 Steve,

 On Thu, 2011-08-25 at 00:39 -0400, Steve Totaro wrote:
 ...

  For fax, I use Hylafax and for text, I use Kannel.  These are WAY more
  powerful than Asterisk apps.  With Kannel, I used the Bluetooth GSM
  modem to send SMS from my cell.  Kannel is awesome as is HylaFAX
 
  I used the Asteirsk System() app to call lynx with a special URL.  The
  URL contains all the authentication, recipient, and SMS body.  Calling
  that URL via System(), as I said, I like lynx, causes an SMS to be
  sent.  Kannel is extremely customizable.  I once had ten cell phones
  for for SMS modems.  My findings with t-mobile were that each phone
  could send an SMS once a second.  With ten, using chan_bluetooth, I
  could send ten SMS per second using ten phones.  Kannel is very well
  developed.  Chan_mobile is incredible.
 
  The same is true with HylaFAX.
 
  Thanks,
  Steve T

  I'm looking at using Kannel for a project here. Would you mind if I
 contacted you off list with some getting started questions?

 Skyler


Skyler,

I would be glad to help within reason.  Since it is not Asterisk and I use
app System() and Lynx as the glue, it wouldn't fit asterisk user's list
anyways.  I use fast AGI for most of the SMS variables.

Helping within reason is good for my karma, too much and I need to be
compensated.  At the very least, thanked publically ;

Like the old Italian saying, I give my friends just enough so that they
need me, but not too much so that they dont

I have quite a bit of experience with Kannel and the code.

Hit me up and let's see what help I can provide.

Thanks,
Steve T
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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[Steve Totaro]

On Wed, Aug 24, 2011 at 2:42 AM, Per Jessen p...@computer.org wrote:

 Linuxguy123 wrote:

  My original post didn't mention it, but I would like my home system to
  be Asterisk based.
 
  Has anyone figured out how to minimize cell charges when on the road
  via making calls via the home phone system ?

 Yep, look up DISA:

 http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA


 /Per Jessen, Zürich

 --


Just curious how DISA would help with cell phone usage charges.  I have
unlimited data, voice, and text for $40/mo so I don't much care about the
cost, just the brain cancer.

In the Mano River Area of Africa and I assume elsewhere, where everything is
SIM Based and inbound calls are free even if you use up your balance, for a
month after activating the SIM, I could see some benefit.

But at least here, if you are on a per minute plan, how would DISA help?
Obviously, different countries and carriers do things differently, but I
don't pay for anything extra, no roaming, nothing.

When overseas, I buy a phone card or a SIM.

Maybe if toll free didn't count as minutes, you could setup a TF VoIP number
on your Asterisk box and save on your cell phone.

For my situation, DISA is pointless except for road warriors who call all
over the world, from anywhere, they can call into the corp system, get
dialtone and skip the whole process of expense reports for work related
calls.  It makes things less complex, not more.

Maybe if you explain your situation and how your plan works, but for me,
personally, DISA would be a an added cost and complication.

The only purpose I can think of for myself could be accomplished by spoofing
caller id.

Thanks,
Steve T
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Re: [asterisk-users] Looking for ideas for nice home phone system

[Steve Totaro]

NBX100 by Polycom.  True plug and play.  Can be IP but uses MAC on the LAN
except for the PBX.

It may be discontinued but there is plenty on Ebay.  Cheap, scales well,
tons of options.

On Wed, Aug 24, 2011 at 10:34 AM, C F shma...@gmail.com wrote:

 The 824 is NOT discontinued.

 On 8/23/11, John Novack jnov...@stromberg-carlson.org wrote:
 
 
  C F wrote:
  On Tue, Aug 23, 2011 at 5:21 PM, John Novack
  jnov...@stromberg-carlson.org  wrote:
  snip
 
  What do you mean by MD?
 
 
  MD is a common telephony term for Manufacture Discontinued
 
  John Novack
 
  --
 
  Dog is my Co-pilot
 
 

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