Re: [asterisk-users] pjsip endpoint reachable

[Thomas Ray]

On 2022-10-14, 11:31 AM, "asterisk-users on behalf of marek" 
 
wrote:

hi,

we are migrating from chan_sip to pjsip

i want logs like this about pjsip endpoints

[Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer 'endpoint22' is now 
Reachable. (15ms / 2000ms)

is it possible?

thanks

Marek


Marek-

This is all logged by the system. You can look for things like this:

[2022-10-13 19:53:11] VERBOSE[32593] res_pjsip/pjsip_configuration.c: Endpoint 
902 is now Unreachable
[2022-10-13 19:53:11] VERBOSE[32593] res_pjsip/pjsip_options.c: Contact 
902/sip:902@192.168.88.248:5060 is now Unreachable.  RTT: 0.000 msec
[2022-10-13 19:54:08] VERBOSE[32593] res_pjsip/pjsip_configuration.c: Endpoint 
902 is now Reachable
[2022-10-13 19:54:08] VERBOSE[32593] res_pjsip/pjsip_options.c: Contact 
902/sip:902@192.168.88.248:5060 is now Reachable.  RTT: 3.193 msec

[2022-10-02 02:47:44] VERBOSE[21527] res_pjsip_registrar.c: Removed contact 
'sip:410@192.168.88.251:5083' from AOR '410' due to expiration

[2022-10-02 01:09:30] VERBOSE[19080] res_pjsip_registrar.c: Added contact 
'sip:905@192.168.88.14:5060' to AOR '905' with expiration of 3600 seconds



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Re: [asterisk-users] Channel names with semicolons

[Thomas Ray]

 

 

From: asterisk-users  on behalf of 
"Joshua C. Colp" 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 

Date: Wednesday, September 7, 2022 at 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Channel names with semicolons

 

On Wed, Sep 7, 2022 at 11:17 AM Antony Stone 
 wrote:

On Wednesday 07 September 2022 at 11:44:54, Antony Stone wrote:

> Hi.

This is a follow-up to an email I posted earlier today to the list, although I 
haven't seen it come back yet.  If it's under moderation for some reason, I 
hope some kindly admin will release it :)

 

There's nothing in the moderator queue that I can see.

 


> I'm trying to deal with a problem regarding putting a call on hold and then
> later resuming it.  I am using chan_sip throughout, and Asterisk 16.



> The main thing which is puzzling me about this is that I see examples of
> both Local/number@context-0ce9;1 and Local/number@context-0ce9;2
> during the processing of the calls.
> 
> What is the significance of the number following the semi-colon?
> 
> I also see in verbose logging output:
> 
> [2022-09-07 09:37:57.310706] pbx VERBOSE[29148]: dial.c:598 in
> handle_frame: Local/number@context-0ce9;1 answered
> 
> [2022-09-07 09:37:57.310792] pbx VERBOSE[29155][C-1265]:
> bridge_channel.c:2252 in bridge_channel_internal_push_full: Channel
> SIP/Trunkname-2b55 joined 'simple_bridge' basic-bridge <7e260e93-
> abd4-48ea-96f1-33601165dba2>
> 
> [2022-09-07 09:37:57.310937] pbx VERBOSE[29149][C-1265]:
> bridge_channel.c:2252 in bridge_channel_internal_push_full: Channel
> Local/number@context-0ce9;2 joined 'simple_bridge' basic-bridge
> <7e260e93- abd4-48ea-96f1-33601165dba2>
> 
> 
> So, when the channel Local/number@context-0ce9;1 gets answered, the
> result is to bridge the channels Local/number@context-0ce9;2 and
> SIP/Trunkname-2b55

I see something very similar in the documentation about local channels at 
https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels - 
there are examples of both devices-ecf0;1 and devices-ecf0;2 but no mention of 
what the final digit means.

Can anyone enlighten me please?

 

A single channel can't do two things at once (you can't have a channel talking 
to Alice while also executing the Voicemail dialplan application for example) - 
so Local channels solve this by having 

two independent channels that exchange things back and forth internally. The ;2 
leg is the one that gets sent into the dialplan, while the ;1 leg is doing 
whatever dialed it decides to do with it. If you send audio to ;1 it then pops 
out of ;2, and vice versa.

 

--- 

Oh I had the channel numbers reversed in my previous example. So yeah, this.

 

 

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Re: [asterisk-users] Channel names with semicolons

[Thomas Ray]

From https://wiki.asterisk.org/wiki/display/AST/Channels

"The primary exception is with Local Channels. In the case of local channels, 
you'll typically have two local channel legs, one that is treated as outbound 
and the other as inbound. In this case both are really inside Asterisk, but one 
is executing dialplan and the other is not. The leg executing dialplan is the 
one treated as inbound."

In your case, context-0ce9;1 is the inbound channel because you did 
Dial(Local/number@context) and context-0ce9;2 is the outbound channel 
because it did the Dial to another destination. Simply, the numbers represent 
each leg of a local channel.

Tom


On 2022-09-07, 10:18 AM, "asterisk-users on behalf of Antony Stone" 
 wrote:

On Wednesday 07 September 2022 at 11:44:54, Antony Stone wrote:

> Hi.

This is a follow-up to an email I posted earlier today to the list, 
although I 
haven't seen it come back yet.  If it's under moderation for some reason, I 
hope some kindly admin will release it :)

> I'm trying to deal with a problem regarding putting a call on hold and 
then
> later resuming it.  I am using chan_sip throughout, and Asterisk 16.



> The main thing which is puzzling me about this is that I see examples of
> both Local/number@context-0ce9;1 and Local/number@context-0ce9;2
> during the processing of the calls.
> 
> What is the significance of the number following the semi-colon?
> 
> I also see in verbose logging output:
> 
> [2022-09-07 09:37:57.310706] pbx VERBOSE[29148]: dial.c:598 in
> handle_frame: Local/number@context-0ce9;1 answered
> 
> [2022-09-07 09:37:57.310792] pbx VERBOSE[29155][C-1265]:
> bridge_channel.c:2252 in bridge_channel_internal_push_full: Channel
> SIP/Trunkname-2b55 joined 'simple_bridge' basic-bridge <7e260e93-
> abd4-48ea-96f1-33601165dba2>
> 
> [2022-09-07 09:37:57.310937] pbx VERBOSE[29149][C-1265]:
> bridge_channel.c:2252 in bridge_channel_internal_push_full: Channel
> Local/number@context-0ce9;2 joined 'simple_bridge' basic-bridge
> <7e260e93- abd4-48ea-96f1-33601165dba2>
> 
> 
> So, when the channel Local/number@context-0ce9;1 gets answered, the
> result is to bridge the channels Local/number@context-0ce9;2 and
> SIP/Trunkname-2b55

I see something very similar in the documentation about local channels at 

https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels - 
there are examples of both devices-ecf0;1 and devices-ecf0;2 but no mention 
of 
what the final digit means.

Can anyone enlighten me please?


Antony.

-- 
Never automate fully anything that does not have a manual override 
capability. 
Never design anything that cannot work under degraded conditions in 
emergency.

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

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Re: [asterisk-users] TCP dial via proxy

[Thomas Ray]

The answer is chan_pjsip. You can do this with chan_pjsip. There’s no real 
support for chan_sip anymore. It’s dead, it’s going away. No fixes or updates 
will be accepted against it as of this point.

 

From: asterisk-users  on behalf of 
Dovid Bender 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 

Date: Thursday, July 21, 2022 at 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] TCP dial via proxy

 

David,

 

We had this exact "issue" in the past and were not able to figure out how to do 
it. Where we wanted tcp we prefixed the sip URI with "force_tcp". So:

Dial(SIP/1234@1.1.1.1//2.2.2.2)

became:

Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2)

On Kamailio's side in the FORWARD block we added:

# HACK for forcing TCP
if ($oU != $null && $(oU{s.len}) != 0) {
$var(prefix) = $(oU{s.substr,0,9});
if ($var(prefix) == "force_tcp") {
$rU = $(oU{s.substr,9,0});
add_uri_param( "transport=tcp" );
$fs = "tcp:" + $Ri + ":5060";
}
}

 

 

 

On Wed, Jul 20, 2022 at 10:47 PM David Cunningham  
wrote:

Hello,

 

We have an Asterisk dial which sends the call via a proxy using //, for example:

 

Dial(SIP/${EXTEN}@peer_address//proxy_address)

 

Does anyone know how we can make the SIP to the proxy use TCP? We tried making 
proxy_address match a peer in sip.conf with "transport = tcp" but that didn't 
seem to work. We are using chan_sip.

 

Thanks very much for any advice.

 

-- 

David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782

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Re: [asterisk-users] Listen on 2 of 3 IP addresses

[Thomas Ray]

Moving to chan_pjsip solves this problem.

 

From: asterisk-users  on behalf of 
David Cunningham 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 

Date: Sunday, July 17, 2022 at 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Listen on 2 of 3 IP addresses

 

Thank you for that reply, Joshua.

 

Henning, we can restrict access using the firewall, but unfortunately that 
doesn't solve the problem with the address that's put in the SDP.

 

 

On Sat, 16 Jul 2022 at 03:14, Henning Follmann  
wrote:

On Fri, Jul 15, 2022 at 08:57:46AM -0300, Joshua C. Colp wrote:
> On Fri, Jul 15, 2022 at 1:37 AM David Cunningham 
> wrote:
> 
> > Hello,
> >
> > We have an Asterisk server with 3 IP addresses, and need to listen on only
> > 2 of those. This is with chan_sip. Does anyone know if it's possible?
> >
> > If Asterisk listens on the third address then it seems to cause problems
> > with the media address put in the SDP for our use case.
> >
> 
> It's not. The chan_sip module allows you to bind to one thing, either a
> specific address or an any address.
> 

Well...
maybe chan_sip cannot, but your OS can restrict traffic on any port/iface.

-H

-- 
Henning Follmann   | hfollm...@itcfollmann.com


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Re: [asterisk-users] Listen on 2 of 3 IP addresses

[Thomas Ray]

The answer is to use chan_pjsip since chan_sip is all but dead, unsupported and 
going away in v21. Instead of making chan_sip listen on all IPs then making a 
bunch of firewall/filtering rules that block one of them from being used, use 
chan_pjsip. It is the right solution as it is the supported driver and will 
continue to see updates and bugfixes.

If you haven't started a migration from chan_sip to chan_pjsip yet, it is best 
advised to make a migration plan and do it now. If there is a case where 
something in chan_sip hasn't made it into chan_pjsip (or it did but hasn't been 
really touched/updated since) and you really need that functionality, now is 
the time to figure that out.

On 2022-07-15, 11:14 AM, "asterisk-users on behalf of Henning Follmann" 
 wrote:

On Fri, Jul 15, 2022 at 08:57:46AM -0300, Joshua C. Colp wrote:
> On Fri, Jul 15, 2022 at 1:37 AM David Cunningham 

> wrote:
> 
> > Hello,
> >
> > We have an Asterisk server with 3 IP addresses, and need to listen on 
only
> > 2 of those. This is with chan_sip. Does anyone know if it's possible?
> >
> > If Asterisk listens on the third address then it seems to cause problems
> > with the media address put in the SDP for our use case.
> >
> 
> It's not. The chan_sip module allows you to bind to one thing, either a
> specific address or an any address.
> 

Well...
maybe chan_sip cannot, but your OS can restrict traffic on any port/iface.

-H

-- 
Henning Follmann   | hfollm...@itcfollmann.com


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[asterisk-users] Antwort: Re: sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?

[Thomas Roos]

Hi,
thank you for pointing me to my problem - I used the fixed ip in the 
extensions.conf, when using peer name my problem is gone.
thanks!



Von:"Joshua C. Colp" 
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Datum:  12.11.2019 14:44
Betreff:Re: [asterisk-users] sip.conf host!=dynamic peer specific 
options (e.g. directmedia=off, transport=tcp) not working!?
Gesendet von:   "asterisk-users" 



On Tue, Nov 12, 2019 at 3:06 AM Thomas Roos  
wrote:
Hi, 
when using some non dynamic  host eg. host=192.168.111.153 in sip.conf 
asterisk is not considering specific peer options eg. directmedia=off, 
transport=tcp 
if I set host=dynamic and register the sip phone it works as expected. 
Is this a bug or feature - I wanna disable the usage of directmedia for 
some peers with fixed ip but wanna allow it in general. Same with 
transport=tcp. 

[97] 
type=peer 
host=192.168.111.153 
transport=udp 
context=extern 
dtmfmode=auto 
directmedia=off 

cheers, Thomas 

Is this an incoming or outgoing call? Are you sure that entry is being 
used? What's the console output? 

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?

[Thomas Roos]

Hi,
when using some non dynamic  host eg. host=192.168.111.153 in sip.conf
asterisk is not considering specific peer options eg. directmedia=off, 
transport=tcp 
if I set host=dynamic and register the sip phone it works as expected.
Is this a bug or feature - I wanna disable the usage of directmedia for 
some peers with fixed ip but wanna allow it in general. Same with 
transport=tcp.

[97]
type=peer
host=192.168.111.153
transport=udp
context=extern
dtmfmode=auto
directmedia=off

cheers, Thomas
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Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

[Thomas Peters]

Thanks for the reply John.

About 85-90% of what this box has to do is just handle calls, but it also has 
options to transfer calls to the main phone system, which up to now has been 
another asterisk box. For example, you can hit 6 to be transferred to the Lost 
& Found Department.

I do have allowguest set to “yes” already, but of course I also have type=peer 
and the other stuff for a sip trunk.

The Avaya engineer is telling me that I need to change my “From” header, and I 
don’t know how to do that.

I have to figure this out in the next few days, or I’m in deep doo-doo.

-T

Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

From: asterisk-users  On Behalf Of 
John Kiniston
Sent: Tuesday, February 26, 2019 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Thomas,
Does the Asterisk box need to do anything other than handle calls for this one 
specific IVR? IE does it ever originate calls?
If it's only recieving calls then I'd turn on guest access and not even bother 
with a peer.
Just set
[general]
context=transit-ivr
allowguest=yes

On Tue, Feb 26, 2019 at 3:13 PM Thomas Peters 
mailto:tpet...@mcts.org>> wrote:
Hello all, I hope someone can help me with this old Asterisk version. I have to 
run this version because of a custom IVR written on it. Porting it would take 
much too long and we’d have to hire a consultant because of all the hooks it 
has into Oracle databases and real-time information.

We have a brand-new Avaya phone system in place and we will be cutting over to 
it in late March 2019.

Presently:

  *   We have an Asterisk 13.3.2 server with no phones registered to it, acting 
as a PSTN gateway. Calls come into it and get distributed to other Asterisk 
boxes with phones.
  *   If a call comes in from the provider marked as having been dialed as 
xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
  *   The IVR box runs Asterisk 1.8.7.0 and a custom IVR.

Where we have to get to:

  *   The new Avaya Session Manager has to have a working SIP trunk to the IVR 
so it can pass calls that come into xxx-xxx-6711 to it.

What the problem is:

  *   I don’t fully understand what’s going on here, neither how it works now, 
nor what I need to do to make Avaya’s SM happy.
  *   When I do sip show peers on my IVR box, I see the Avaya session manager:

jerec*CLI> sip show peers

Name/username  HostDyn 
Forcerport ACL Port Status

sessionmgr110.10.0.17  
5060 OK (1 ms)

  *   The Avaya engineer says he is seeing “SIP/2.0 400 Bad FROM header” in his 
trace screen, and his SM status screen shows “500 NOT REACHABLE” as the status 
for our IVR.

 *   He says we are sending

“asterisk” sip:asterisk@(null):0;tag=as682f2c53

as the “From” in the SIP header.

  *   He wants us to send
10.10.0@mcts.org<mailto:10.10.0@mcts.org>
or more likely
mailto:10.10.0@mcts.org>>

instead.

  *   Pings from either end to the other work just fine.
  *   nmap doesn’t show port 5060 open. It shows only port 22/tcp open. But 
then again, my main asterisk PBX doesn’t show that port open either. So I don’t 
think that means anything.

The IVR machine (Asterisk 1.8.7.0) sip.conf file has an old section for the old 
PSTN gateway, and a new section I just added for the session manager.
Old section for existing connections to the IVR:

[general]
;context=transit-ivr
context=incoming
disallow=all
allow=ulaw
canreinvite=no

[sipivr]
host=dynamic
secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV
type=peer
;context=transit-ivr
context=incoming
dtmfmode=inband

The new section, with many failed experiments commented out, is after the 
[sipivr] section:
[sessionmgr1]
type=peer
;type=friend
port=5060
host=10.90.0.17
dtmfmode=inband
allowguest=yes
qualify=yes
realm=mcts.org<http://mcts.org>
promiscredir=yes
;Some have suggested using canreinvite=no with Avaya- didn't try that yet
;canreinvite=no
canreinvite=yes
transport=tcp
;context=incoming
context=from-internal
;username=10.90.0.103
fromdomain=mcts.org<http://mcts.org>
disallow=all
allow=ulaw
allow=alaw
tcpenable=yes
tcpbindaddr=0.0.0.0:5060<http://0.0.0.0:5060>

Nothing I tried seems to make it stop sending asterisk@(null) in the header. 
This is supposed to be a sip trunk, not an extension, so I think I should NOT 
be user a username or secret. I’m not even sure what promiscredir does, or if 
it’s helping or harming me.

There’s virtuall

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

[Thomas Peters]

All:

We have made progress.

We have the Avaya Session Manager showing a good connection now and we're 
looking at getting calls across to the Asterisk IVR machine now.

Seems we were defaulting to UDP and the Avaya SM wanted TCP. They switched the 
Avaya end to UDP and I made some other settings changes in sip.conf, and it's 
working better.

I'm using a trimmed-down sip.conf suggested by John Kiniston. Thank you very 
much indeed for your suggestions John.

My sip.conf file now consists of
[sessionmgr1]
host=10.10.0.17
type=peer
qualify=yes
fromuser=10.10.0.103

But I don't think it's the final product; Some tweaking might still be needed. 
The 10.10.0.17 device is the Avaya Session Manager, and the .103 is the 
Asterisk-based IVR.

Thanks again,
-T


Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

From: asterisk-users  On Behalf Of 
Thomas Peters
Sent: Tuesday, February 26, 2019 4:11 PM
To: Asterisk User List (asterisk-users@lists.digium.com) 

Subject: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Hello all, I hope someone can help me with this old Asterisk version. I have to 
run this version because of a custom IVR written on it. Porting it would take 
much too long and we'd have to hire a consultant because of all the hooks it 
has into Oracle databases and real-time information.

We have a brand-new Avaya phone system in place and we will be cutting over to 
it in late March 2019.
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Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

[Thomas Peters]

Thanks for the reply John.

About 85-90% of what this box has to do is just handle calls, but it also has 
options to transfer calls to the main phone system, which up to now has been 
another asterisk box. For example, you can hit 6 to be transferred to the Lost 
& Found Department. 

I do have allowguest set to “yes” already, but of course I also have type=peer 
and the other stuff for a sip trunk.

The Avaya engineer is telling me that I need to change my “From” header, and I 
don’t know how to do that. 

I tried various things in sip.conf and now I get 
> sip show peer sessionmgr1
...
Callerid : "10.90.0@mcts.org" <>
...

My sip.conf file now has:
fromdomain=mcts.org
realm=mcts.org
callerid="10.90.0@mcts.org"


I have to figure this out in the next few days, or I’m in deep doo-doo. 

-T

Thomas M. Peters | Sr. Systems Administrator |  mailto:tpet...@mcts.org  


From: asterisk-users <mailto:asterisk-users-boun...@lists.digium.com> On Behalf 
Of John Kiniston
Sent: Tuesday, February 26, 2019 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<mailto:asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Thomas,
Does the Asterisk box need to do anything other than handle calls for this one 
specific IVR? IE does it ever originate calls?
If it's only recieving calls then I'd turn on guest access and not even bother 
with a peer.
Just set
[general]
context=transit-ivr
allowguest=yes

On Tue, Feb 26, 2019 at 3:13 PM Thomas Peters <mailto:tpet...@mcts.org> wrote:
Hello all, I hope someone can help me with this old Asterisk version. I have to 
run this version because of a custom IVR written on it. Porting it would take 
much too long and we’d have to hire a consultant because of all the hooks it 
has into Oracle databases and real-time information. 
We have a brand-new Avaya phone system in place and we will be cutting over to 
it in late March 2019. 
Presently: 
• We have an Asterisk 13.3.2 server with no phones registered to it, acting as 
a PSTN gateway. Calls come into it and get distributed to other Asterisk boxes 
with phones. 
• If a call comes in from the provider marked as having been dialed as 
xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
• The IVR box runs Asterisk 1.8.7.0 and a custom IVR.
 
Where we have to get to: 
• The new Avaya Session Manager has to have a working SIP trunk to the IVR so 
it can pass calls that come into xxx-xxx-6711 to it. 
 
What the problem is: 
• I don’t fully understand what’s going on here, neither how it works now, nor 
what I need to do to make Avaya’s SM happy.
• When I do sip show peers on my IVR box, I see the Avaya session manager: 
jerec*CLI> sip show peers
Name/username  Host    Dyn 
Forcerport ACL Port Status
sessionmgr1    10.10.0.17  
5060 OK (1 ms)
• The Avaya engineer says he is seeing “SIP/2.0 400 Bad FROM header” in his 
trace screen, and his SM status screen shows “500 NOT REACHABLE” as the status 
for our IVR.
o He says we are sending 
“asterisk” sip:asterisk@(null):0;tag=as682f2c53 
as the “From” in the SIP header.
• He wants us to send 
mailto:10.10.0@mcts.org  
   or more likely 
mailto:10.10.0@mcts.org>
instead.
• Pings from either end to the other work just fine. 
• nmap doesn’t show port 5060 open. It shows only port 22/tcp open. But then 
again, my main asterisk PBX doesn’t show that port open either. So I don’t 
think that means anything. 
 
The IVR machine (Asterisk 1.8.7.0) sip.conf file has an old section for the old 
PSTN gateway, and a new section I just added for the session manager. 
Old section for existing connections to the IVR: 
 
[general]
;context=transit-ivr
context=incoming
disallow=all
allow=ulaw
canreinvite=no
 
[sipivr]
host=dynamic
secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV
type=peer
;context=transit-ivr
context=incoming
dtmfmode=inband
 
The new section, with many failed experiments commented out, is after the 
[sipivr] section: 
[sessionmgr1]
type=peer
;type=friend
port=5060
host=10.90.0.17
dtmfmode=inband
allowguest=yes
qualify=yes
realm=http://mcts.org
promiscredir=yes
;Some have suggested using canreinvite=no with Avaya- didn't try that yet
;canreinvite=no
canreinvite=yes
transport=tcp
;context=incoming
context=from-internal
;username=10.90.0.103
fromdomain=http://mcts.org
disallow=all
allow=ulaw
allow=alaw
tcpenable=yes
tcpbindaddr=http://0.0.0.0:5060
 
Nothing I tried seems to make it stop sending asterisk@(null) in the header. 
This is supposed to be a sip trunk, not an extension, so I think I should NOT 
be user a username or secret. I’m not even sure what promiscredir does, or if 
it’s helping or harming me. 
 
There’s virtually nothing in the logs about this connection, other than this: 
[Feb 26 16:05:42] NOTICE[32142

[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

[Thomas Peters]

Hello all, I hope someone can help me with this old Asterisk version. I have to 
run this version because of a custom IVR written on it. Porting it would take 
much too long and we'd have to hire a consultant because of all the hooks it 
has into Oracle databases and real-time information.

We have a brand-new Avaya phone system in place and we will be cutting over to 
it in late March 2019.

Presently:

  *   We have an Asterisk 13.3.2 server with no phones registered to it, acting 
as a PSTN gateway. Calls come into it and get distributed to other Asterisk 
boxes with phones.
  *   If a call comes in from the provider marked as having been dialed as 
xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
  *   The IVR box runs Asterisk 1.8.7.0 and a custom IVR.

Where we have to get to:

  *   The new Avaya Session Manager has to have a working SIP trunk to the IVR 
so it can pass calls that come into xxx-xxx-6711 to it.

What the problem is:

  *   I don't fully understand what's going on here, neither how it works now, 
nor what I need to do to make Avaya's SM happy.
  *   When I do sip show peers on my IVR box, I see the Avaya session manager:

jerec*CLI> sip show peers

Name/username  HostDyn 
Forcerport ACL Port Status

sessionmgr110.10.0.17  
5060 OK (1 ms)

  *   The Avaya engineer says he is seeing "SIP/2.0 400 Bad FROM header" in his 
trace screen, and his SM status screen shows "500 NOT REACHABLE" as the status 
for our IVR.
 *   He says we are sending

"asterisk" sip:asterisk@(null):0;tag=as682f2c53

as the "From" in the SIP header.

  *   He wants us to send
10.10.0@mcts.org<mailto:10.10.0@mcts.org>
or more likely
mailto:10.10.0@mcts.org>>

instead.

  *   Pings from either end to the other work just fine.
  *   nmap doesn't show port 5060 open. It shows only port 22/tcp open. But 
then again, my main asterisk PBX doesn't show that port open either. So I don't 
think that means anything.

The IVR machine (Asterisk 1.8.7.0) sip.conf file has an old section for the old 
PSTN gateway, and a new section I just added for the session manager.
Old section for existing connections to the IVR:

[general]
;context=transit-ivr
context=incoming
disallow=all
allow=ulaw
canreinvite=no

[sipivr]
host=dynamic
secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV
type=peer
;context=transit-ivr
context=incoming
dtmfmode=inband

The new section, with many failed experiments commented out, is after the 
[sipivr] section:
[sessionmgr1]
type=peer
;type=friend
port=5060
host=10.90.0.17
dtmfmode=inband
allowguest=yes
qualify=yes
realm=mcts.org
promiscredir=yes
;Some have suggested using canreinvite=no with Avaya- didn't try that yet
;canreinvite=no
canreinvite=yes
transport=tcp
;context=incoming
context=from-internal
;username=10.90.0.103
fromdomain=mcts.org
disallow=all
allow=ulaw
allow=alaw
tcpenable=yes
tcpbindaddr=0.0.0.0:5060

Nothing I tried seems to make it stop sending asterisk@(null) in the header. 
This is supposed to be a sip trunk, not an extension, so I think I should NOT 
be user a username or secret. I'm not even sure what promiscredir does, or if 
it's helping or harming me.

There's virtually nothing in the logs about this connection, other than this:
[Feb 26 16:05:42] NOTICE[32142] chan_sip.c: Peer 'sessionmgr1' is now 
Reachable. (1ms / 2000ms)

Can anyone help?










Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

[Thomas Peters]

Actually, I was wrong about that. We no longer use OVM. It's actually Citrix 
Xencenter 7.6

Thomas M. Peters | Sr. Systems Administrator |  tpet...@mcts.org  
Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
Milwaukee County Transit System 

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook & Twitter 

-Original Message-
From: asterisk-users  On Behalf Of 
Doug Lytle
Sent: Tuesday, January 15, 2019 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail 
- Thomas Peters

>>> Carlos and Stefan (and other who have helped):

Thomas,

You stated that your virtual environment was Oracle, would that equate to 
VirtualBox?

Doug

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Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

[Thomas Peters]

Carlos and Stefan (and other who have helped):

I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling 
Asterisk is unrealistic in my position but I wonder if I can build the one 
module. Here's what I do have: 

apbx:~ $ locate *res_timing_timerfd*
/usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts
/usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo
/usr/src/asterisk-1.8.23.1/res/res_timing_timerfd.c
/usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.makeopts
/usr/src/asterisk-1.8.7.0/res/.res_timing_timerfd.moduleinfo
/usr/src/asterisk-1.8.7.0/res/res_timing_timerfd.c

Why I have 1.8.23 and 1.8.7 I don't know. Asterisk on this system is version 
1.8.7.0.

NEXT QUESTION: There are NO timing modules listed in /etc/asterisk/modules.conf 
at all. The only ones that are explicitly loaded are format_wav format_pcm 
format_mp3 and res_musiconhold. And there are "preload" directives for 
pbx_config.so and chan_local.so.

Is res_timing_dahdi.so getting loaded somewhere else? Or is it a default of 
some kind?

SYSTEM TIME OF DAY CLOCK which someone asked about, seems accurate. I did 
watch -n1 date
and watched the time tick up, perfectly synchronized to my mobile phone. It 
might be off by a second or so, I'd have a hard time knowing for sure. NTPD is 
running, but not working for some reason. I fixed it (ownership of ntp.conf 
wrong) so now ntpq -pn returns a server ID. 



Thomas M. Peters | Sr. Systems Administrator |  tpet...@mcts.org  
Desk: 414.343.1720 | Helpdesk: x3400 or  helpd...@mcts.org
Milwaukee County Transit System 

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook & Twitter 

-Original Message-
From: asterisk-users  On Behalf Of 
Stefan Viljoen
Sent: Tuesday, January 15, 2019 12:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail 
- Thomas Peters

Here’s what I get:

apbx*CLI> module show like timing
Module Description  Use 
Count
res_timing_pthread.so  pthread Timing Interface 0
res_timing_dahdi.soDAHDI Timing Interface   4
2 modules loaded

So what would you suggest? (And thanks in advance.)

Thomas

I've had some good experience with 

res_timing_dahdi

both when we ourselves were still on 1.8 and now with us on Asterisk 13 as well.

To force usage of a certain timer, specify in your modules.conf, e. g. to force 
use of DAHDI timing only, I did the following in my modules.conf:

.
.
.
load => res_timing_dahdi.so
noload => res_timing_pthread.so
noload => res_timing_timerfd.so

That said, we have had some weird issues trying to run Asterisk in virtual 
machines - all our instances (16 of them) are physical machines.

We did a deployment at Azure in a Centos 7 "stock Azure" VM awhile ago and it 
suddenly lost the capability to encode .gsm audio files. All .gsm files the 
virtualised Asterisk 13 instances produced were all corrupt and no player would 
want to play the .gsm files. Neither could SOX convert them to anything. So we 
had to switch over to .wav, and then use a mixmonitor hook and manually convert 
the .wav files back to .gsm in SOX after each recording was written by Asterisk 
in .wav format. There were no errors logged, Asterisk just mysteriously lost 
the capacity to encode .gsm files when running on the Azure VM instance we had.

So quite probably the virtual environment / hypervisor you're using is part of 
the issue and switching timing modules around won't solve anything...

Have you checked that the system time is sane, and that one second on a stop 
watch externally to the VM instance, equates to one second inside it?

Because the symptoms described could indicate that the clock in the VM is just 
running too fast - or that some timing implementation detail inside Asterisk 
itself is running too fast.

Regards

Stefan


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Re: [asterisk-users] Various extensions ring once and go to voicemail

[Thomas Peters]

Here’s what I get:

apbx*CLI> module show like timing
Module Description  Use 
Count
res_timing_pthread.so  pthread Timing Interface 0
res_timing_dahdi.soDAHDI Timing Interface   4
2 modules loaded

So what would you suggest? (And thanks in advance.)


Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

From: asterisk-users  On Behalf Of 
Carlos Chavez
Sent: Monday, January 14, 2019 4:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail


On 1/14/19 4:02 PM, Duncan Turnbull wrote:

Sent from my iPad

On 15/01/2019, at 10:34 AM, Thomas Peters 
mailto:tpet...@mcts.org>> wrote:
Duncan:

You may have it right—I took one phone and set the ring time to 60 seconds. I 
now get about 4 rings on that one.

I wonder how I can change the timing source.

In one version (and I can’t recall which) asterisk moved to an internal timing 
system, to avoid the hardware need.

There should be quite a lot of discussion of it in the archives or perhaps 
voipinfo

I don’t know if you can slow the VM processor speed? I am guessing it is 
counting something much faster than it used to

Cheers Duncan



*CLI> module show like timing
Module Description  Use 
Count  Status  Support Level
res_timing_dahdi.soDAHDI Timing Interface   0   
   Running  core
res_timing_pthread.so  pthread Timing Interface 0   
   Running  extended
res_timing_timerfd.so  Timerfd Timing Interface 1   
   Running  core
3 modules loaded

This will show you what module Asterisk is using for timing.  You can try 
doing a noload on the two you do not need.



--

Telecomunicaciones Abiertas de México S.A. de C.V.

Carlos Chávez

+52 (55)8116-9161
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Re: [asterisk-users] Various extensions ring once and go to voicemail

[Thomas Peters]

Duncan:

You may have it right-I took one phone and set the ring time to 60 seconds. I 
now get about 4 rings on that one.

I wonder how I can change the timing source.

Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

From: asterisk-users  On Behalf Of 
Duncan
Sent: Monday, January 14, 2019 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Various extensions ring once and go to voicemail



On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters 
mailto:tpet...@mcts.org>> wrote:

We have an old Asterisk 1.8.7.0 system desperately need to keep alive for 
another 6 months or so. We had all kinds of hardware problems, so we 
virtualized it.

Thats a while back, I think it tended to use zaptel or dahdi hardware as a 
timer, you may need to find a timing source as perhaps the clock in the VM is 
just going too fast



Now, random extensions ring once and go straight to voicemail.

I went to one of the affected extensions and changed the ring time from the 
default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. 
Other extensions ring once or twice.  After some time has gone by since this 
was first reported, all phones in my random sample ring only twice.

As I trace a call to that extension through the log, I notice it setting the 
ring timer properly (I think) and then I see
app_dial.c - SIP/1234- is ringing
Then eventually
app_dial.c: -- Nobody picked up in 3 ms

But according to the timestamps, the time from the one event to the other is 
ten seconds!

Here's another example- call starts:
[2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1265", "0?Set(__RINGTIMER=0)") 
in new stack
. . .
[2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- SIP/3327-1266 is 
ringing
. . .
[2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked up in 
2 ms
So again, the elapsed time is nowhere near 20 seconds.

Another: This time the ring time has been set to 30 seconds (and I still get 
only 2 rings)
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1304", "1?Set(__RINGTIMER=30)") 
in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[s@macro-exten-vm:5] Set("SIP/4704-1304", "RT=30") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[s@macro-dial-one:30] Set("SIP/4704-1304", "D_OPTIONS=trWw") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: -- 
SIP/3327-1305 is ringing
. . .
[2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: -- Nobody 
picked up in 3 ms

So, after 9 seconds, it says "Nobody picked up after 3 ms"???

Is this some weirdness of Oracle VMs? Anybody have any advice for me?


Additional information:
FreePBX version 2.9.0.7
PBX in a Flash Version 1.2 Daemon Status

* Asterisk  * ONLINE  * Dahdi * ONLINE  * MySQL  * ONLINE  *
* SSH   * ONLINE  * Apache* ONLINE  * Iptables   * OFFLINE *
* Fail2ban  * OFFLINE * IP Connect* ONLINE  * Ip6tables  * OFFLINE *
* BlueTooth * ONLINE  * Hidd  * ONLINE  * NTPD   * ONLINE  *
* Sendmail  * ONLINE  * Samba * OFFLINE * Webmin * LOADING *
* Ethernet0 * ONLINE  * Ethernet1 * ONLINE  * Wlan0  *   N/A   *

* Running Asterisk Version : Asterisk 1.8.7.0
* Asterisk Source Version  : 1.8.7.0
* Dahdi Source Version : 2.5.0.1+2.5.0.1
* Libpri Source Version: 1.4.12
* Addons Source Version: 1.4.7
****
Voipserver on 10.10.141.251 - eth0
Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel: 
2.6.18-92.1.6.el5



Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

-- 
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-- Bandwi

[asterisk-users] Various extensions ring once and go to voicemail

[Thomas Peters]

We have an old Asterisk 1.8.7.0 system desperately need to keep alive for 
another 6 months or so. We had all kinds of hardware problems, so we 
virtualized it.

Now, random extensions ring once and go straight to voicemail.

I went to one of the affected extensions and changed the ring time from the 
default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. 
Other extensions ring once or twice.  After some time has gone by since this 
was first reported, all phones in my random sample ring only twice.

As I trace a call to that extension through the log, I notice it setting the 
ring timer properly (I think) and then I see
app_dial.c - SIP/1234- is ringing
Then eventually
app_dial.c: -- Nobody picked up in 3 ms

But according to the timestamps, the time from the one event to the other is 
ten seconds!

Here's another example- call starts:
[2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1265", "0?Set(__RINGTIMER=0)") 
in new stack
. . .
[2019-01-14 08:17:33] VERBOSE[13311] app_dial.c: -- SIP/3327-1266 is 
ringing
. . .
[2019-01-14 08:17:41] VERBOSE[13311] app_dial.c: -- Nobody picked up in 
2 ms
So again, the elapsed time is nowhere near 20 seconds.

Another: This time the ring time has been set to 30 seconds (and I still get 
only 2 rings)
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[3327@cc-long-distance:1] ExecIf("SIP/4704-1304", "1?Set(__RINGTIMER=30)") 
in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[s@macro-exten-vm:5] Set("SIP/4704-1304", "RT=30") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] pbx.c: -- Executing 
[s@macro-dial-one:30] Set("SIP/4704-1304", "D_OPTIONS=trWw") in new stack
. . .
[2019-01-14 08:41:54] VERBOSE[16008] app_dial.c: -- 
SIP/3327-1305 is ringing
. . .
[2019-01-14 08:42:05] VERBOSE[16008] app_dial.c: -- Nobody 
picked up in 3 ms

So, after 9 seconds, it says "Nobody picked up after 3 ms"???

Is this some weirdness of Oracle VMs? Anybody have any advice for me?


Additional information:
FreePBX version 2.9.0.7
PBX in a Flash Version 1.2 Daemon Status

* Asterisk  * ONLINE  * Dahdi * ONLINE  * MySQL  * ONLINE  *
* SSH   * ONLINE  * Apache* ONLINE  * Iptables   * OFFLINE *
* Fail2ban  * OFFLINE * IP Connect* ONLINE  * Ip6tables  * OFFLINE *
* BlueTooth * ONLINE  * Hidd  * ONLINE  * NTPD   * ONLINE  *
* Sendmail  * ONLINE  * Samba * OFFLINE * Webmin * LOADING *
* Ethernet0 * ONLINE  * Ethernet1 * ONLINE  * Wlan0  *   N/A   *

* Running Asterisk Version : Asterisk 1.8.7.0
* Asterisk Source Version  : 1.8.7.0
* Dahdi Source Version : 2.5.0.1+2.5.0.1
* Libpri Source Version: 1.4.12
* Addons Source Version: 1.4.7

Voipserver on 10.10.141.251 - eth0
Red Hat Enterprise Linux Server release 4.5 (Tikanga) :32 Bit Kernel: 
2.6.18-92.1.6.el5



Thomas M. Peters | Sr. Systems Administrator |  
tpet...@mcts.org<mailto:tpet...@mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or  
helpd...@mcts.org<mailto:helpd...@mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>

1942 N 17th Street | Milwaukee, WI  53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter 
<https://twitter.com/RideMCTS>

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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Thomas Peters]

You did indeed warn me. I've made progress, gotten the dhcp option 242 to work, 
and finally gotten the phone to the point where it asks for a username and 
password. I defined these on the Asterisk server. I entered them on the phone. 
It says "Acquiring Service" and sits there.  At least it sets the clock on the 
phone now. 

TCPDump shows traffic going back & forth every few seconds: 
10.1.138.245.13602 > ad-apbx.mcts.org.sip
ad-apbx.mcts.org.sip > 10.1.138.245.60206
10.1.138.245.30360 > ad-apbx.mcts.org.sip:

But it sits at Acquiring Service eternally. 

The username and password I am entering are the Asterisk extension and asterisk 
extension secret. Should these be entered somewhere in the craft menu instead?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Wednesday, March 07, 2018 1:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

On Tue, Mar 06, 2018 at 05:36:04PM +, Thomas Peters wrote:
> But please don't tell me the only way to program up each phone is via 
> the craft interface?
> 
> Every other phone I've ever used requires a configuration file, which 
> has the MAC address of the phone as its name. The Avaya phones must 
> have some other method. Unless I have to embed the mac address and 
> particulars for all the phones into the 46xxsettings.txt file??

I tried to warn you, didn't I? :)

The phones themselves are nice, when used with an Avaya PBX. What I have seen 
is that these phones are really dumb themselves and need a decent PBX that does 
the smartstuff via proprietary interfaces (H.232 gatekeeper like IPOffice or 
whatever they use with the SIP based Communication Manager)

There is some intelligence in parsing the 46xxsettings file, but AFAIK you need 
the CRAFT menu to do this (apart from MAC adresses).

My advise: sell these phones, any SIP device you can buy with the proceeds is 
more intelligent when used with Asterisk.

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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Thomas Peters]

Ok, to review,  I'm trying to get Avaya 9608G to come up in a pure Asterisk 
environment-- no Avaya SBC or gateway or any other Avaya gear in sight.

I have the phone working to the point where it boots up properly, then displays 
a Username and Password prompt, and says its extension is 123 and the time is 
4:57p, which is wrong.

But please don't tell me the only way to program up each phone is via the craft 
interface?

Every other phone I've ever used requires a configuration file, which has the 
MAC address of the phone as its name. The Avaya phones must have some other 
method. Unless I have to embed the mac address and particulars for all the 
phones into the 46xxsettings.txt file??

How do I get the phone number and authentication into the phone if it isn't 
looking for (for example) 50CD22B47C00.cfg or the like?? These are all the 
files it attempt to read, from /var/log/httpd/access.log:

10.1.138.245 - - [06/Mar/2018:10:19:09 -0600] "GET /voipcfg/96x1Supgrade.txt 
HTTP/1.1" 200 941 "-" "Mozilla/4.0 (compatible; MSIE 6.0)"
10.1.138.245 - - [06/Mar/2018:10:19:09 -0600] "GET /voipcfg/46xxsettings.txt 
HTTP/1.1" 200 421835 "-" "Mozilla/4.0 (compatible; MSIE 6.0)"


By the way, I don't know if I shared this already, but here's my DHCP settings 
on my network that allowed it to find my asterisk server.

First, outside of any defined scope, declare option 242. This is about line 13 
for me, YMMV

option option-242 code 242 = string;

Next, I defined a tiny little scope for Avaya phones:

pool {
range dynamic-bootp 10.194.138.241 10.194.138.254;
deny members of "aastra";
deny members of "polycom-phones";
deny members of "cyberdata-paging";
deny members of "digium-phones";
deny members of "cisco-spa-phones";
allow members of "avaya-phones";
deny members of "sophos";
#option option-242 
"MCIPADD=10.180.202.110,HTTPSRVR=fonia.teste.com.br,L2Q=0,L2QVLAN=0";
option option-242 
"MCIPADD=10.194.141.251,HTTPSRVR=10.194.141.251,HTTPDIR=voipcfg";
  }

Having already created a /voipcfg directory on the web server, the phone now 
boots from there and reads 46xxsettings.txt.



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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Thomas Peters]

Right-- I've seen the Avaya document you cite below. It says "To administer 
DHCP option 242, make a copy of an existing option 176" but I don't have any 
example of option 176 or 242 to copy, and don't know what to do to 
/etc/dhcpd.conf to make it offer option 242. 

Then there's this long table of parameters to use with (presumably) option 242.

I was hoping someone had a working minimal example of a dhcp option 242 config 
I could copy and modify.

They only have minimal function? No speed dials, BLFs, etc? 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Thursday, March 01, 2018 3:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

On Wed, Feb 28, 2018 at 08:48:38PM +, Thomas Peters wrote:
> I'd like to start configuring my Avaya 9608G phones for use on 
> Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other 
> phones on my system without major issues.
> 
> I've read the discussion back in March, May and August of 2016, but 
> unfortunately, my difficulty is much more basic. I think it has to do 
> with DHCP, specifically, what options I'm offering the phone via DHCP.

So you might want to start without configurations from DHCP. Enter the 
file/http server in the phone manually and point it to a http server containing 
the needed firmware files and a correct 46xxsettings.txt for you asterisk.

Avaya uses other dhcp options for these phone (242):
https://downloads.avaya.com/elmodocs2/one-X_Deskphone_Edition/R1.5/output/16_300698_4/admn054.html

BTW these phones are a terrible waste when used with Asterisk. They only 
provided minimal functionality (calls and transfer).

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Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Thomas Peters]

Doug: 

Not sure these phones understand TFTP.  Docs say they don't. But I made changes 
like what you suggested. No luck; the phone still never bothers the asterisk 
server for anything. I still see only broadcasts (only ARPs.)



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Wednesday, February 28, 2018 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

>>> but unfortunately, my difficulty is much more basic. I think it has to do 
>>> with DHCP, specifically, what options I’m offering the phone via DHCP.

I'm using ISC DHCP with the following:

 option tftp-server-name "10.30.100.109";
next-server 10.30.100.109;

That provides enough information for my Polycom phones to find their server.

Doug

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[asterisk-users] Avaya 9608G and DHCP and TFTP and HTTP oh my

[Thomas Peters]

I'd like to start configuring my Avaya 9608G phones for use on Asterisk / 
FreePBX / PBX-In-a-Flash. I'm using a variety of other phones on my system 
without major issues.

I've read the discussion back in March, May and August of 2016, but 
unfortunately, my difficulty is much more basic. I think it has to do with 
DHCP, specifically, what options I'm offering the phone via DHCP.


  1.  I haven't even gotten the phone to look for any files on my asterisk 
server. I know this because the only traffic on my asterisk server going to or 
from the phone is ARP packets. Here's a tcpdump:

$ tcpdump -v -vv host 10.194.138.245
tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes

10:07:57.256579 IP (tos 0x0, ttl  64, id 0, offset 0, flags [DF], proto: ICMP 
(1), length: 48) dhcpserver.mcts.org > 10.194.138.245: ICMP echo request, id 
47332, seq 0, length 28
10:07:59.215981 arp who-has 10.194.138.245 (Broadcast) tell 0.0.0.0
10:08:00.216237 arp who-has 10.194.138.245 (Broadcast) tell 0.0.0.0
10:08:02.563742 arp who-has firewall.mcts.org (Broadcast) tell 10.194.138.245


  1.  The question of boot protocol isn't settled at all-many people say go 
ahead and use tftp (which would be nice, since all my other phones do) but 
there's this document https://downloads.avaya.com/css/P8/documents/100176078 
that says in part: (page 56)

HTTP Generic Setup
You can store the binary file, upgrade file, and settings file on an HTTP 
server. With proper
administration, the deskphone seeks out and uses that material. Some 
functionality might be
lost by a reset if the HTTP server is unavailable.

Note:
If you used TFTP to provide the binary, upgrade, and settings files to older 
Avaya
IP telephones, note that 9600 Series IP Telephones do not support TFTP; you
must use HTTP or HTTPS instead.

This shouldn't matter, as I think I have provided for both- I added a new 
directory to my apache config files on my asterisk server, which opens fine in 
a web browser: 
http://astserver.mcts.org/voipcfg/<http://ad-apbx.mcts.org/voipcfg/> And I do 
have those same files in the tftpboot directory, so if it was trying to use 
tftp, I'd see that.

/tftpboot is r-x to everyone, so are all the subdirectories, and the avaya 
files are all 744 (owned by root.)

It contains an index page that just says "hello world" and all the config and 
upgrade files I downloaded from Avaya. But the phone never looks for anything 
on this server.  It's got 46xxsettings.txt 96x1Supgrade.txt and some tar files 
and bin files. But the tcpdump tells me it never checks.


  1.  The docs I've been able to find talk about DHCP option 242 and 176, which 
I've never used before.

All I know about are options like this:

pool {
range dynamic-bootp 10.194.138.241 10.194.138.254;
allow  members of "avaya-phones";
deny members of "sophos";
  # 10.194.141.251 is asterisk server
#option tftp161  "http://10.194.141.251";;
option tftp161  "10.194.141.251";
option tftp162  "/voipcfg";
next-server 10.194.141.251;
 }


DHCP is doing some things right for this phone:

14:22:14.020926 arp who-has 10.194.138.245 tell dhcpserver
14:22:15.001014 IP (tos 0x10, ttl  16, id 0, offset 0, flags [none], proto: UDP 
(17), length: 330) dhcpserver.bootps > 10.194.138.245.bootpc: BOOTP/DHCP, 
Reply, length: 302, xid:0x2c46f14e, flags: [none]
  Your IP: 10.194.138.245
  Server IP: astserver.mcts.org
  Client Ethernet Address: 50:cd:22:b4:7c:a6 (oui Unknown)
  Vendor-rfc1048:
DHCP:OFFER
SID:dhcpserver
LT:144000
SM:255.255.0.0
DG:firewall.mcts.org
DN:"transit.mcts.org"
NS:domainc1.transit.mcts.org,domainc2.transit.mcts.org
NTP:dhcpserver
14:22:15.020927 arp who-has 10.194.138.245 tell dhcpserver
14:22:16.020925 arp who-has 10.194.138.245 tell dhcpserver
14:22:16.061936 IP (tos 0x10, ttl  16, id 0, offset 0, flags [none], proto: UDP 
(17), length: 330) dhcpserver.bootps > 10.194.138.245.bootpc: BOOTP/DHCP, 
Reply, length: 302, xid:0x2c46f14e, flags: [none]
  Your IP: 10.194.138.245
  Server IP: astserver.mcts.org
  Client Ethernet Address: 50:cd:22:b4:7c:a6 (oui Unknown)
  Vendor-rfc1048:
DHCP:ACK
SID:dhcpserver
LT:144000
SM:255.255.0.0
DG:firewall.mcts.org
DN:"transit.mcts.org"
NS:domainc1.transit.mcts.org,domainc2.transit.mcts.org
NTP:dhcpserver
14:22:16.232035 arp who-has 10.194.138.245 (Broadcast) tell 0.0.0.0
14:22:17.232318 arp who-has 10.194.138.245 (Broadcast) t

[asterisk-users] UNIQUEID not unique in different channels

[Thomas]

Hello,

since al long time I have used UNIQUEID for identify calls in my dialplan, 
statistics...

Now I have had an problem, after I have checked log file I found out following:

calls same time ( hours:seconds) came in.

CallID, DID, channel name (3cf9 to 3cfa) are different.

Only UNIQUEID is identical 

Is there any known bugs or ist UNIQUEID not unique and can not be used for 
identify calls?


[Aug 28 10:14:11] VERBOSE[6413][C-1f1e] pbx.c: -- Executing 
[XX@patton4970in:1] NoOp("SIP/outbound.patton-3cf9", "INFO: 
eingehender Anruf Context patton4970in von 
SIP/outbound.patton-3cf9 ""  zu XX") in new stack
UNIQUEID: 1503908051.54095


[Aug 28 10:14:11] VERBOSE[6418][C-1f1f] pbx.c: -- Executing 
[XX@patton4970in:1] NoOp("SIP/outbound.patton-3cfa", "INFO: 
eingehender Anruf Context patton4970in von 
SIP/outbound.patton-3cfa ""  zu XX") in new stack
UNIQUEID: 1503908051.54095


thanks..



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[asterisk-users] ConfBridge increase talking volume as standard

[Thomas]

Hello,

is it possible to increase talking volume for caller in ConfBridge as standard 
without need to press buttons after joining an conference room.

best regards 
Thomas 

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[asterisk-users] Conference bridge profile does not exist

[Thomas]

Hello,

I have ab profile in /etc/asterisk/confbridge.conf but in my dialplan this 
profile is not found
I tried a lot, but did no solution.
What can be wrong?


[out_bridge]
type=bridge


exten => ,n,ConfBridge(${conf_room},out_bridge)

[Jun  5 19:27:09] WARNING[11008][C-0468]: app_confbridge.c:1612 
confbridge_exec: Conference bridge profile out_bridge does not exist

The Dialplan is in the same directory and changes will be working, so the 
directory for confbridge.conf should be OK.

version is Asterisk 11.13.1~dfsg-2+deb8u1 built by pbuilder @ compile on a 
x86_64 running Linux on 2016-10-24 19:32:53 UTC

best regards
Thomas

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[asterisk-users] Confbridge or Bridge

[Thomas]

Hi,

an agent should wait in an confroom and hear some music or tones.
On an website he should put in an tel number for calling to somebody.
This is working wih phpagi very well.

I have problems with
Agent should hear ringing when callee is called, busy for may be 5 s that 
callee is busy or congestion

I have problem when agent is the leaving before callee, that hangup should 
send to callee channel.

Also habe somebody an source for an soundfile with every 5s an beep that agent 
is hearing when he is the only person in confroom.

thanks 
Thomas



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[asterisk-users] AMI Originate not working

[Thomas]

Hello,

I want to call an phone and if phone picked up I want to ring another phone.
Or I want to connect to an running channel and then call another phone or move 
to an ConfBridge

Iam using PHP 
$channel = 'IAX2/556-1696';
or $channel = 'SIP/0019736363636@outbound.patton';
$exten = '';
$context = 'test_callout';
$priority = '1';


$parameters = array(
'Channel' => $channel, 
'Exten' => $exten,
'Context' => $context,
'Priority' => $priority,
);
self::manager_com('Originate', $parameters);


I get only this message, but no action or other information
  == Manager 'vserver_webastmanager' logged on from 127.0.0.1
  == Manager 'vserver_webastmanager' logged off from 127.0.0.1


The AMI access in general should work, because I use it for another commands 
for example QueueAdd

best regards

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[asterisk-users] softphone instead of desktop phones

[Thomas]

Hello,
Iam lookong for an Softphone for iPhor oder Android smartphone using togehter 
with an headset.
I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP 
phone.

Is there an better softphone?

Or are there softphone solutions for PC desktop MAC or Android with an 
headset?
I want to save cost for desktop phones.

thanks Thomas


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Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

[James Thomas]

All I can tell you is where -3 comes from.
>From http://www.voip-info.org/wiki/view/Asterisk+Expressions :
REMAINDER(x,y) computes the remainder of dividing x by y. The return value
is x - n*y, where n is the value x/y, rounded to the nearest integer. If
this quotient is 1/2, it is rounded to the nearest even number.

-3 comes from:
n = x/y = 957/60 = 15.95 which rounds to 16
n*y = 16*60 = 960
x - 960 = 957-960 = -3

I'm not mathematically gifted either but I think the n is the problem. it
shouldn't be the rounded result it should be the integer part of x/y (n=15)

Can you just use modulo instead: ${MATH(${myNum}%60,int)}
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[asterisk-users] nagios asterisk check SIP

[Thomas]

Hi,

Iam loocking for an programm to check the SIP port of an Asterisk asterisk.

Ome time ago I have used 
#/usr/bin/sipsak 
but it seemed that it is not working anymore?

Any ideas what I can use instead?


best regards
Thomas


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[asterisk-users] looking for soft phone can be manged like Snom phones

[Thomas]

Hi,

its easy to control Snom phones from outside by an http request
http://username:password@192.168.0.1/command.htm?key=KEYEVENT

http://wiki.snom.com/FAQ/Can_I_control_my_snom_phone_remotely

also its easy to configure that an Snom phone send me an http request
for example for an incomming call
http://my.serverIP/pbx/snomcom.php?mac=$mac&actionid=1401


I would need an softphone for Linux and/or Mac with same or similar 
functionality.


best regads
Thomas

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Re: [asterisk-users] Queue logfile txt format in mySQL needed

[Thomas]

Am Donnerstag, 21. Januar 2016, 09:52:53 schrieben Sie:
> > From: Thomas 
> > To: asterisk-users@lists.digium.com,
> > Date: 01/21/2016 04:17 AM
> > Subject: [asterisk-users] Queue logfile txt format in mySQL needed
> > Sent by: asterisk-users-boun...@lists.digium.com
> > 

> > database instead of using var/log/asterisk/queue_log?
> 
> I haven't done this myself, but it looks like you just need to set up the
> appropriate database connections. See here for a semi-recent example:
> http://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql

Hi,
thanks...


in /etc/asterisk/res_config_mysql.c
for example define in [mydatabase] acsess to your database

in /etc/asterisk/extconfig.conf

queue_log => mysql,mydatabase,queue_log


CREATE TABLE IF NOT EXISTS `queue_log` (
`id` int(10) unsigned NOT NULL auto_increment,
`time` varchar(40) default NULL,
`callid` varchar(80) NOT NULL default '',
`queuename` varchar(80) NOT NULL default '',
`agent` varchar(80) NOT NULL default '',
`event` varchar(32) NOT NULL default '',
`data` varchar(255) NOT NULL default '',
`data1` varchar(255) NOT NULL default '',
`data2` varchar(255) NOT NULL default '',
`data3` varchar(255) NOT NULL default '',
`data4` varchar(255) NOT NULL default '',
`data5` varchar(255) NOT NULL default '',
PRIMARY KEY (`id`)
);


after make an reload of asterisk info will be written in this table and not 
any more in the txt file



I found some explantion for the data written in the fields:
https://wiki.asterisk.org/wiki/display/AST/Queue+Logs




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[asterisk-users] Queue logfile txt format in mySQL needed

[Thomas]

Hello,

Iam using queues and agents, thats OK.

I have interesting information form Asterisk in txt file format
var/log/asterisk/queue_log

Today Iam reading these txt files and wrote them in an mySQL databases.

I would need this information more realtime. Some information I do writing in 
the dialplan direct in an mySQL database.

Is there any way that Asterisk write this information direct in an mySQL 
database instead of using var/log/asterisk/queue_log?

best regards
Thomas





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[asterisk-users] AMI StarAstAPI.php not working for commands like SIPpeers with more output

[Thomas]

Hi,
Iam using StarAstAPI.php.
If Iam sending Action commands like 'reload' everythink works fine.

If I send an Action like 'SIPpeers' I get:

  ["Response:"]=>
  string(8) " Success"
  ["ActionID:"]=>
  string(2) " 2"
  ["EventList:"]=>
  string(6) " start"
  ["Message:"]=>
  string(29) " Peer status list will follow"


But how I receive the peer list?

best regards
Thomas





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[asterisk-users] asterisk segfault debian jessie asterisk 11.13

[Thomas]

Hi,
every two weeks the asterisk process has a segfault. Any idea whats reason or 
what I can do...
thanks

pc kernel: [1780743.239296] asterisk[11362]: segfault at 0 ip   (null) 
sp 7f1e396b04a8 error 14

version is debian jessie
Asterisk 11.13.1~dfsg-2+b1 built by buildd @ brahms on a x86_64 running Linux




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Re: [asterisk-users] Phones don't stop ringing when queue is answered

[James Thomas]

A few things I would try-
Change WaitExten to Wait(2)
Change Queue(queue_level_1,rtnC,18) to Queue(queue_level_1,rtnC,,,18)
Add an Answer() after the first Wait(2)
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Re: [asterisk-users] Phones don't stop ringing when queue is answered

[James Thomas]

What purpose do the WaitExten()s serve here? Are you really allowing the
caller to connect to different extensions in the test-queue context? Have
you tried without the WaitExten()s?
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Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[Thomas Stein]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Am 21.04.15 um 13:38 schrieb akhilesh chand:
> Hi Guenther,
> 
> When  I executed nmap -p5060  I got below output.
> 
> [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx
> 
> Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap
> scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com 
> (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE
> SERVICE 5060/tcp filtered sip

Maybe your softphone is trying UDP?

cheers
t.

> Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds
> 
> On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter
>  wrote:
> 
> On 04/21/2015 04:58 PM, akhilesh chand wrote:
>> Hi Guenther,
> 
>> What did you recommend to me, I did accordingly but there is no 
>> log showing in asterisk CLI. I'm getting same problem.
> 
> 
> 
>> Regards Akhilesh
> 
> Hi Akhilesh,
> 
> looks like your firewall is blocking it.
> 
> Have you tried 'nmap -p5060 ' or something
> similar?
> 
> Regards Guenther
> 
> 
> 
> 
>> On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter 
>> mailto:gboel...@gmail.com>> wrote:
> 
>> On 04/20/2015 12:31 PM, akhilesh chand wrote:
>>> Hi Folks,
> 
>>> I'm trying to register softphone(X-lite) but I'm not able to 
>>> register softphone whenever I'm trying to register softphone I 
>>> got below error
> 
>>> Inline image 1
> 
>>> Is there any document/guide line where I will get process to 
>>> register softphone in asterisk(Which is installed in EC2
>>> Cloud).
> 
>> Don't make it to complicated ...
> 
>> Connect to your Asterisk via ssh and run asterisk -rvv.
> 
>> Then let your Phone try to register. Asterisk should show you 
>> what's getting wrong.
> 
>> If you can't see anything while trying to register, shutdown
>> your firewall and try it again ...
> 
>> Regards Guenther
> 
> 
> 
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>> 
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> 
> 
> 
> 
> 
>> 
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>> visit: http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
> 
> 
> 

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Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[Thomas Stein]

Am 21.04.15 um 11:00 schrieb akhilesh chand:
> Hi Greg,
> 
> I moved REJECT rule to last in the list but I'm getting same error.

Please shutdown the firewall completely and try again.

cheers
t.

> Regards
> Akhilesh
> 
> On Mon, Apr 20, 2015 at 5:57 PM, Greg Woods  wrote:
> 
>>
>> On Mon, Apr 20, 2015 at 1:58 AM, akhilesh chand > > wrote:
>>
>>> Chain INPUT (policy ACCEPT)
>>> target prot opt source   destination
>>> ACCEPT all  --  0.0.0.0/00.0.0.0/0state
>>> RELATED,ESTABLISHED
>>> ACCEPT icmp --  0.0.0.0/00.0.0.0/0
>>> ACCEPT all  --  0.0.0.0/00.0.0.0/0
>>> ACCEPT tcp  --  0.0.0.0/00.0.0.0/0state NEW
>>> tcp dpt:22
>>> REJECT all  --  0.0.0.0/00.0.0.0/0
>>>  reject-with icmp-host-prohibited
>>> ACCEPT udp  --  0.0.0.0/00.0.0.0/0udp
>>> spt:5060
>>> ACCEPT udp  --  0.0.0.0/00.0.0.0/0udp
>>> spt:5083
>>> ACCEPT udp  --  0.0.0.0/00.0.0.0/0udp
>>> spt:1
>>>
>>
>> It looks like youre REJECT rule is getting hit before the accept rules for
>> asterisk. Try moving the REJECT rule to last in the list. I think your
>> firewall is blocking asterisk.
>>
>> --Greg
>>
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> 


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Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[Thomas Stein]

Am 20.04.15 um 09:58 schrieb akhilesh chand:
> Hi Thomas,
> 
> I followed your recommended command in asterisk CLI which is mentioned in
> above chain mail but I'm not able capture any log related to softphone.

Well then i guess you have the softphone somehow misconfigured. You have
to double check your settings. IP / Port / Protokoll and so on.

best regards
t.

> Chain INPUT (policy ACCEPT)
> target prot opt source   destination
> ACCEPT all  --  0.0.0.0/00.0.0.0/0state
> RELATED,ESTABLISHED
> ACCEPT icmp --  0.0.0.0/00.0.0.0/0
> ACCEPT all  --  0.0.0.0/00.0.0.0/0
> ACCEPT tcp  --  0.0.0.0/00.0.0.0/0state NEW tcp
> dpt:22
> REJECT all  --  0.0.0.0/00.0.0.0/0reject-with
> icmp-host-prohibited
> ACCEPT udp  --  0.0.0.0/00.0.0.0/0udp spt:5060
> ACCEPT udp  --  0.0.0.0/00.0.0.0/0udp spt:5083
> ACCEPT udp  --  0.0.0.0/00.0.0.0/0udp spt:1
> 
> Chain FORWARD (policy ACCEPT)
> target prot opt source   destination
> REJECT all  --  0.0.0.0/00.0.0.0/0reject-with
> icmp-host-prohibited
> 
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
> 
> 
> On Mon, Apr 20, 2015 at 1:16 PM, Thomas Stein  wrote:
> 
>> Am 20.04.15 um 09:43 schrieb akhilesh chand:
>> > Hi Thomas,
>>
>> Hello.
>>
>> > Yes I'm able to access asterisk server but there is no logs capture into
>> > log file related to softphone.If you want more information regarding
>> > configuration means sip.conf and extension.conf  I will share.
>>
>> Could you increase the verbose level?
>>
>> # core set verbose 6
>> # sip set debug on
>>
>> Looking for blocking Firewall Rules is also a valid point.
>>
>> cheers
>> t.
>>
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> 


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Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[Thomas Stein]

Am 20.04.15 um 09:43 schrieb akhilesh chand:
> Hi Thomas,

Hello.

> Yes I'm able to access asterisk server but there is no logs capture into
> log file related to softphone.If you want more information regarding
> configuration means sip.conf and extension.conf  I will share.

Could you increase the verbose level?

# core set verbose 6
# sip set debug on

Looking for blocking Firewall Rules is also a valid point.

cheers
t.


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Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[Thomas Stein]

Am 20.04.15 um 06:31 schrieb akhilesh chand:
> Hi Folks,
> 
> I'm trying to register softphone(X-lite) but I'm not able to register
> softphone whenever I'm trying to register softphone I got below error

Do you have access to the asterisk server? If so, what's in the logs?

cheers
t.


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Re: [asterisk-users] Show Log(NOTICE) messages on the console

[James Thomas]

logger.conf... You should start by comparing that file between the two
servers. Not sure if it's still called logger.conf in asterisk 11 though.
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Re: [asterisk-users] if statement recording - after hours

[James Thomas]

GotoIfTime()
Check out-
http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime

If the time is within a certain range, execute the recording dialplan. If
it's outside the range, then skip to the dialplan after the recording stuff.
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Re: [asterisk-users] why not forwarding to this number?

[Thomas Perron]

hi
resolved.

added:  include=outgoing

cheers



On Sat, Aug 9, 2014 at 7:34 AM, Administrator TOOTAI 
wrote:

> Le 09/08/2014 12:23, Thomas Perron a écrit :
>
>  exten => s,1,Answer()
>> exten => s,n,Wait(2)
>> exten => s,n,SayAlpha(495256)
>> exten => s,n,Wait(2)
>> exten => s,n,Dial(SIP/222)
>> exten => s,n,Hangup
>>
>
> Hi,
>
> you could at least tell us why you think the number is not forwarded
> (error ?) and what you have in logs
>
> I would at first check if the peer is registered assuming you have a
> congestion or unavailable error.
>
> --
> Daniel
>
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[asterisk-users] why not forwarding to this number?

[Thomas Perron]

exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,SayAlpha(495256)
exten => s,n,Wait(2)
exten => s,n,Dial(SIP/222)
exten => s,n,Hangup
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Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

[James Thomas]

Is the quality the same incoming from mobile as outgoing to mobile?


On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles 
wrote:

> I'm having a problem with a new SIP trunk.
>
> Calls within the UK to fixed lines are fine, but calls to mobiles have
> noticeably poorer audio quality.
>
> I thought it might have been a codec issue; we have used G.726 for internal
> and external calls  (over primary ISDN and GSM).  So I tried allowing
> "alaw",
> (G.711 A-law)  which is the native codec used within the PSTN in this
> country,
> but this made no improvement.
>
> We had
>   disallow=all
>   allow=g726
>
> in the [general] section of sip.conf.  In the section for one of the
> phones, I
> added
>   allow=alaw
> and then inserted
>   Set(SIP_CODEC=alaw)
> in the relevant part of extensions.conf.  For good measure, I also added
>   NoOp(Codec was ${SIP_CODEC})
> in the "h" extension.  The messages in the Asterisk CLI appeared to show
> that
> the audio codec was correctly being set to "alaw", and on hangup I got
> "Codec
> was alaw", but there was no improvement to the sound quality.
>
> Is there something I am doing wrong, or do I need to get in touch with our
> SIP
> trunk provider?
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
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[asterisk-users] Asterisk in debian Wheezy 1.8.13.1 vs. Squeeze 1.8.23.1

[Thomas]

Hello,

in Squeeze Asterisk 1.8.23.1 is installed, in Wheezy older version 
1.8.13.1~dfsg1-3+deb7u3.

With version 1.8.13.1 I have some problems so I would like to install version 
1.8.23.1 used in Squeeze whats running fine for me.

How I can do this?

thanks for help

Thomas

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Re: [asterisk-users] new install: no re-invite and unwanted transcoding

[James Thomas]

I can't help on the first issue, but for the second have you tried doing
Set(${SIP_CODEC}=ulaw) before dialing the trunk? I'm in a similar situation
where we have g722 internally but our trunk provider only offers ulaw so I
see g722->slin->ulaw transcoding. I'm thinking of trying it here (on
1.8.14.1) to troubleshoot our occasional outbound issues.
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Re: [asterisk-users] Asterisk and OSX

[Thomas Rechberger]

Am 14.04.2014 16:19, schrieb Eric Wieling:

So few people use Asteisk on OSX that I doubt anyone will answer.


Look how many answers he got, i got none



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Re: [asterisk-users] additional range parameter for sip peer

[Thomas Rechberger]

Am 29.03.2014 11:12, schrieb Thomas Rechberger:

Many ITSP are using loadbalancers, so if somebody registers on a sip
peer with specific dns host, an incoming call may be received from a
different ip and the host value in peer section doesnt match, so it will
go to default context.

For example Telekom or 1&1, biggest providers in Germany are using too
many different addresses that its not practical to define them all (up
to 50 hosts and they still add!), as this will also generate too much
traffic (especially with qualify and multiple registrations) and they
may even lock you out as untrusted, which may even result in that they
will block asterisk permanently for everybody. Thats not really desirable.

I think its also not recommended in terms of security to use default
context with allowguest=yes and sort the incoming calls by header,
because this can be faked easily.

 From my understanding the permit/deny parameters are only used for
incoming calls if host is set to dynamic and then there will be no
outgoing registration to remote peer possible. permit/deny is used for
access, not for matching.

How about an additional parameter where an range of ip addresses can be
defined in peer section, which will be used for matching calls?

hostmatchrange=x.x.x.x/24




anyone here?
What do you think about using permit/deny for host matching?


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[asterisk-users] additional range parameter for sip peer

[Thomas Rechberger]

Many ITSP are using loadbalancers, so if somebody registers on a sip 
peer with specific dns host, an incoming call may be received from a 
different ip and the host value in peer section doesnt match, so it will 
go to default context.


For example Telekom or 1&1, biggest providers in Germany are using too 
many different addresses that its not practical to define them all (up 
to 50 hosts and they still add!), as this will also generate too much 
traffic (especially with qualify and multiple registrations) and they 
may even lock you out as untrusted, which may even result in that they 
will block asterisk permanently for everybody. Thats not really desirable.


I think its also not recommended in terms of security to use default 
context with allowguest=yes and sort the incoming calls by header, 
because this can be faked easily.


From my understanding the permit/deny parameters are only used for 
incoming calls if host is set to dynamic and then there will be no 
outgoing registration to remote peer possible. permit/deny is used for 
access, not for matching.


How about an additional parameter where an range of ip addresses can be 
defined in peer section, which will be used for matching calls?


hostmatchrange=x.x.x.x/24


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Re: [asterisk-users] fromdomain not honored on outbound INVITE request

[Thomas Rechberger]

meanwhile i got an answer from fprior who was testing patch already.

add this code to chan_sip.c :

/* Allow domain to be overridden */
if (!ast_strlen_zero(p->fromdomain))
d = p->fromdomain;
else /* Save for any further attempts */
ast_string_field_set(p, fromdomain, d);

after this one (approx. at line 13741)
/* Allow user to be overridden */
if (!ast_strlen_zero(p->fromuser))
l = p->fromuser;
else /* Save for any further attempts */
ast_string_field_set(p, fromuser, l);

on 11.x it should after fromname section, but i guess wont matter.
Will test this soon.
The code is really that simple, is it really needed to have it signed 
from the author?



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Re: [asterisk-users] fromdomain not honored on outbound INVITE request

[Thomas Rechberger]

Am 20.03.2014 16:07, schrieb Joshua Colp:


That being said... this specific issue was brought up to me by a friend
and I was actually going to look at it over an upcoming weekend when I
have time. Can I guarantee it? No, but I'm going to try. That's how much
I care. Can I do this for every issue? No. I work 8+ hours a day as it
is on Asterisk and also some on the weekends. So many issues, not enough
time. ^_^


That'll be great, i would look into this myself but i have not any clue 
about C.

But should be able to test it.


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[asterisk-users] fromdomain not honored on outbound INVITE request

[Thomas Rechberger]

https://issues.asterisk.org/jira/browse/ASTERISK-20841

The patch was already posted by someone but then was deleted because of 
guide lines. Is it really that hard to fix? Since 1.8 there is this 
problem but nobody seems to care about. Asterisk isnt only used with 
itsp who dont care about fromdomain. Or are the developers saying, we 
dont care about people who are using Asterisk in smaller installations?



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[asterisk-users] what is actually a trunk in a sip trunk?

[Thomas Rechberger]

no trunking or bonding involved, so why just everybody calls this a trunk?


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[asterisk-users] externhost and reregister

[Thomas Rechberger]

externhost is monitoring for ip changes with an interval of 
externrefresh, so far so good.


Wouldnt it be handy if asterisk would do an sip reregister if it detects 
an ip change?
My SIP provider has sometimes very high intervals of 1 hour and if ip 
changes, the registration doesnt work until it expires or asterisk is 
restarted or sip reload.


Or just everyone uses fixed ip addresses?

For now, seem to have to use a script with sip reload.


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[asterisk-users] allow sip traffic in peer settings

[Thomas Rechberger]

When deny/permit is used in sip peer, does this only make sense when 
host=dynamic is used? What happens if host=ip is set?


And if insecure=invite is set, does this override all above settings?
Whats the relation of those seettings?


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[asterisk-users] Problem with SIP 480 from ITSP

[Thomas Rechberger]

I am using voip with Vodafone as SIP peer for outbound telephony and i 
have a huge problem establishing calls to other people. It works like in 
1 of 5 tries. The peer is sending SIP 480 temporarily not available.
It took a while to identify this, because on the phone you just hear 
busy tone.

On inbound calls i have not detected problems yet.

Calling to mobile numbers works better than to home/office numbers.
If i use a different SIP Peer in same dialplan it works without any problem.
If i connect a "all in one voip router" with vodafone peer it works 
without problem too.


Are they lowering availability if they detect Asterisk? Are there any 
IDs that can be changed, so they cannot detect the PBX ?


Are there any settings in Asterisk which affect timeouts or stuff like 
that?


Here:
http://voicent.com/kb/index.php/support/autodialer/581/sip-error-480-temporarily-unavailable
they write:
"Depending on the exact cause of the error, your solution may vary. For 
example, you can try to add another VOIP account for different lines 
(from the same VOIP service or different one), or slow down the calling 
(by defining dialing intervals in Voicent gateway)"


What they mean with slow down and dial interval?

I have also the impression that the ISP resonds very quick with the 
error. Or has it even to do with the router itself? (i am using OpenWRT 
and had a problem once with NAT-T after public ip change).


If i turn on qualify, the peer refuses also to answer (SIP read 403 
forbidden). I was wondering, how Asterisk even knows that it is registered?


The register and peer status in Asterisk are both ok, i even monitor 
that in Nagios.


Its a big problem in Germany because the ITSP can force you by law to 
use their own branded router models, which of course dont use Asterisk.

So if i would call them, the first answer would be, use a different router.


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Re: [asterisk-users] Not able to get remote channel variables containing RTCP values

[Thomas Rechberger]

Hello Matt


Your best option would be to parse out the values in the various channel
variables and store the ones you want.



thx, then i am gonna go this route. I got the CDR anyway displayed via 
Perl already.



This usually works pretty well, except for CDRs, which are generally a
mess no matter what. :-)



In 1.6 the default was really a mess because "" was not consistent used, 
making parsing a nightmare.




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Re: [asterisk-users] Basic quetion on Asterisk usage

[Thomas Rechberger]

Am 02.12.2013 15:02, schrieb Raghav Goud:

Hi all,
I want  peer-peer communication between two clients (soft phones)
both are behind NAT.I have  an accounts with  SIP proxy for two clients.
To communicate these two clients do i need to use Asterisk ?
As per my understanding Asterisk acts like Gateway where
conventional phone calls data transfer into internet voice calls.
  If am wrong please correct me .
Thanks,
  Raghav.




Asterisk is not required. In your soft phone you only need to enter SIP 
Account from ITSP and it is then "converted" by Provider from SIP to 
plain telephony or routed directly to the other VoIP user which is all 
transparent for you. 2 users on the same Provider, usually can call 4 
free but this depends.


When Asterisk is installed, both phones can be registered to it and both 
phones can communicate over Asterisk directly without even needing an 
ITSP, it goes all over IP. This requires good configuration to prevent 
"hackers" from misusing the system and generating high bills.



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[asterisk-users] Not able to get remote channel variables containing RTCP values

[Thomas Rechberger]

I am not sure if its just me, but i am able to get only local channel 
variables containing RTCP QOS values.

The Version is 1.8.14.
I want to store values of bridged channel in CDR.

Phone is Cisco 7941 SIP and with sip show channelstats i see all the 
relevant information (jitter,packet loss) i want to get. It even 
calculates packet loss in %. But i am not able to store it to CDR.


Asterisk 1.4 seems to have had a function ast_rtp_get_quality but i cant 
find any information about that in sources from 1.8, only a short 
reference in 1.4.


Channel variables like CHANNEL(rtpqos,audio,rxjitter) show only 
information about the local channel. So not really usefull.
In some old version they seemed to have it changed from remote_jitter to 
rxjitter, local_jitter to txjitter and so on. Was not even documented.


The 2 variables RTPAUDIOQOSBRIDGED and RTPAUDIOQOS show exactly the 
things i want, but all information is stored in one field so its not 
really usable because it looks ugly in CDR report and doesnt show packet 
loss in %.


The following interesting variables are completely empty (show 0), here 
is how i write it to CDR in h:

exten => s,n,Set(CDR(txj)=${RTPAUDIOQOSJITTER})
exten => s,n,Set(CDR(rxj)=${RTPAUDIOQOSJITTERBRIDGED})
exten => s,n,Set(CDR(txpl)=${RTPAUDIOQOSLOSS})
exten => s,n,Set(CDR(rxpl)=${RTPAUDIOQOSLOSSBRIDGED})
exten => s,n,Set(CDR(txrtt)=${RTPAUDIOQOSRTT})
exten => s,n,Set(CDR(rxrtt)=${RTPAUDIOQOSRTTBRIDGED})

I also checked variables during call with featurecode, but also empty.

Did i oversee something? Is it the same in Version 11 ?
I dont want to mess with Voipmonitor because i only need 2 variables of 
remote channel. If sip show channelstats is showing everything 
correctly, it should be not that hard to get that information.



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Re: [asterisk-users] [SPAM] - Re: queue moh - Email found in subject

[Andrew Thomas]

Hi Ioan,

I have done that [Set(CHANNEL(musicclass)=…] but it still doesn’t work when a 
‘queue’ call is put on hold.  So, I can get it to play the correct moh when I 
use the ‘r’ option – but still get silence when I don’t include the ‘r’.

So I’ve sort of fixed my second point (with the ‘r’) but the first point 
(without the ‘r’) is still not working.

Thanks for your feedback ☺

Cheers
Andy


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias
Sent: 10 July 2013 20:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM] - Re: [asterisk-users] queue moh - Email found in subject

Hello Andy,

Have you tried using SetMusicOnHold command before Queue command?

BR,
Ioan

On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas  wrote:
Hi All,

Sorry if this has been covered already, but I don't tend to follow this
list as close as I should these days.

Problem is that if a call comes in to a queue without option 'r'
specified - moh plays as expected.  Now, when that call is answered, all
is fine. Trouble comes when that person then puts the caller on-hold.
No moh is heard by the caller (in fact, they get silence).

If I use 'r' - then ringing is heard - but the queue's
musiconhold/musicclass is ignored completely.  When the caller is put on
hold, they do hear moh but the default moh context is used - not the moh
of the queue.

What I need is for the queue's moh to be used when the caller is put on
hold (and without using the 'r' feature).  Is this possible?

* 1.8.16.0 (tried on various flavours of 1.8).
Queue static and realtime (same outcome).

Cheers
Andy












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[asterisk-users] queue moh

[Andrew Thomas]

Hi All,

Sorry if this has been covered already, but I don't tend to follow this
list as close as I should these days.

Problem is that if a call comes in to a queue without option 'r'
specified - moh plays as expected.  Now, when that call is answered, all
is fine. Trouble comes when that person then puts the caller on-hold.
No moh is heard by the caller (in fact, they get silence).

If I use 'r' - then ringing is heard - but the queue's
musiconhold/musicclass is ignored completely.  When the caller is put on
hold, they do hear moh but the default moh context is used - not the moh
of the queue.

What I need is for the queue's moh to be used when the caller is put on
hold (and without using the 'r' feature).  Is this possible?

* 1.8.16.0 (tried on various flavours of 1.8).
Queue static and realtime (same outcome).

Cheers
Andy











-- 

 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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[asterisk-users] language specific email templates

[Thomas Martin]

Hi,

I am new to Asterisk. I'm using it behind a kamailio sip-router to provide 
voicemail boxes to sip-users.
I followed these instruction: 
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x
to set everything up, using ARA with a MySQL DB.

After a few tweaks "everything" is basically working, however, a few questions 
remain that I could not find 
clear answers for - anywhere. Maybe some of the experts can help a little ...  

So far, "my" Asterisk DB features 1 table ("voicemessages") and 2 views 
("vmusers" and "sipusers").

"vmusers" shows the following columns: uniqueid, customer_id, context, mailbox, 
password, fullname, email, pager, stamp
("sipusers" shows: name, defaultuser, type, secret, host, callerid, context, 
mailbox, nat, qualify, fromuser, fromuser, authuser, fromdomain, insecure, 
canreinvite, disallow, allow, restrictcid, defaultip, ipaddr, port, regseconds)

Questions:

1) In  "vmusers", the content for column "fullname" is composed from 2 columns 
(i.e. first_name, last_name) of the kamailio subscriber table 
that do not even "natively" exist there, but were manually added by me (to 
comply with the instructions).
Q1a: Is the fullname column in the "vmusers" view mandatory for Asterisk to 
function properly? "fullname" might always stay empty...
Q1b: Is the stamp column in the "vmusers" view mandatory for Asterisk to 
function properly?

2) in "my" world, sip-users have a language. At certain points, Asterisk will 
communicate with "my" users regarding voicemail.
e.g.: a) Asterisk delivers the message left as a wav-attachment by email,  b) 
VoicePrompts guide the user through the voicemail-menu
So far, I am able to hard-code the language for the voiceprompts in 
extension.conf  with Set(CHANNEL(language)=fr). But this set the language for 
everybody.
Also, you can customise the email template ($emailbody, etc) for voicemail 
delivery in voicemail.conf. However, this is also covers one language only.
Q2a:How can configure Asterisk I pick the voicemail-prompts of the 
respective sip-users language?
Q2b:Is it useful for that purpose, to add a "language" column to the 
"vmusers" view ?
Q2c:Where can language-specific email-templates for voicemail delivery be 
supplied and how can be achieved that the correct one is used ?

3) I would also like to use Asterisk as a SIP client (since Kamailio can't do 
this). Each "vmuser" may have one or many SIP accounts with foreign registrars.
I would like Asterisk to register those accounts with their registrars and 
forward incoming calls to the kamailio (parent) account that the vmuser owns.
(The kamailio (parent) account -in turn- shall forward this call to the user's 
asterisk voice-box, if the user is busy/not-registered/etc ).
Q3a:How can I use ARA to configure Asterisk as a SIP client to act in the 
described fashion ?
Q3b:Is it useful/necessary for that purpose, to host dial-plans in the 
MySQL-DB as well ?

Q4: Can you point to some documentation that explains ARA a little more in 
depth and possibly illustrates a few examples?

For any hint - thank you very much in advance!

Best regards,

-Thomas





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[asterisk-users] (no subject)

[Thomas Perron]

Basic Dial Plan

Why is this plan not engaging the line
exten => 105,n,Dial(SIP/voipvoip.com/1703501)
and dialing the 703 number?

The logs and debug dont show any problems


[incoming]
exten => 44,1,Answer()
exten => 44,n,Wait(1)
exten => 44,n,Playback(beep)
exten => 44,n,Goto(105,105,1)
;
;
[105]
exten => 105,1,Wait(2)
exten => 105,n,Playback(hello-world)
exten => 105,n,Dial(SIP/voipvoip.com/1703501)
exten => 105,n,Hangup()
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[asterisk-users] Connect to an outbound channel and dial a phone number??

[Thomas Perron]

This seems basic but something is missing.


I dial from my cell phone to my DID and enter the context in extensions.conf
I am hoping to cascade through the plan and successfully automatically dial
the 1444 number listed.
But it fails.
And, I dpon't know why?   Should I removed the Hangup application?
Syntax issue somewhere?

I have a good SIP registration with the vendor, voipvoip.

Thanks in advance for any feedback...



[incoming]
exten => 5552530146,1,Answer()
exten => 5552530146,n,Wait(1)
exten => 5552530146,n,Playback(beep)
exten => 5552530146,n,Goto(105,105,1)
;
;
[105]
exten => 105,1,Wait(2)
exten => 105,n,Playback(hello-world)
exten => 105,n,Dial(SIP/voipvoip/1444514)
exten => 105,n,Hangup()

console output ...

-- Executing [5552530146@incoming:1]
Answer("SIP/voipvoip.com-000f", "") in new stack
-- Executing [5552530146@incoming:2] Wait("SIP/voipvoip.com-000f",
"1") in new stack
-- Executing [5552530146@incoming:3]
Playback("SIP/voipvoip.com-000f", "beep") in new stack
--  Playing 'beep.alaw' (language 'en')
-- Executing [5552530146@incoming:4] Goto("SIP/voipvoip.com-000f",
"105,105,1") in new stack
-- Goto (105,105,1)
-- Executing [105@105:1] Wait("SIP/voipvoip.com-000f", "2") in new
stack
-- Executing [105@105:2] Playback("SIP/voipvoip.com-000f",
"hello-world") in new stack
--  Playing 'hello-world.alaw' (language
'en')
-- Executing [105@105:3] Dial("SIP/voipvoip.com-000f", "SIP/
sip3.voipvoip.com/17037171624") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/sip3.voipvoip.com/1444514
[Apr  9 16:07:11] WARNING[994]: chan_sip.c:4169 retrans_pkt: Retransmission
timeout reached on transmission
4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060 for seqno 102 (Critical
Request) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr  9 16:07:11] WARNING[994]: chan_sip.c:4198 retrans_pkt: Hanging up
call 4dd167154ea52bd26d63a95a56aa9526@192.168.1.10:5060 - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- SIP/sip3.voipvoip.com-0010 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [105@105:4] Hangup("SIP/voipvoip.com-000f", "") in new
stack
  == Spawn extension (105, 105, 4) exited non-zero on
'SIP/voipvoip.com-000f'
Asterisk*CLI>
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[asterisk-users] extensions.conf / test DID

[Thomas Perron]

I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.

I have a successful SIP session registered:

Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
sip3.voipvoip.com:5060  N  1112530146 105
Registered   Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI>

Here is the dial plan:
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)


[general]
register =>1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming

; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite

Thoughts please?I think something to do w/ "incoming" is incorrect.
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Re: [asterisk-users] sip registration

[Thomas Perron]

Got it...

Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 954)
Asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
State   Reg.Time
sip3.voipvoip.com:5060  N  444222146 105
Registered  Sun, 07 Apr 2013 09:42:31
1 SIP registrations.
Asterisk*CLI>

Next hurdle is extensions.conf

I must need to establish / correlate my DID number to something.
When I dial my DID I get "you have reached a non working number"





On Sat, Apr 6, 2013 at 5:36 PM, Steve Edwards wrote:

> A better subject will yield better replies.
>
>
> On Sat, 6 Apr 2013, Thomas Perron wrote:
>
>  Shouldnt I be able to at least ping the SIP provider IP?
>>
>
> Not if they don't allow it. They don't.
>
> sip3.voipvoip.com registers fine for me with your credentials.
>
> Did you put the registration statement in the [general] section?
>
> I use the 'append' format so I can group all the cruft for a provider
> together. Like:
>
> ; voipvoip.com
> [general](+)
> register= nn:xx@sip3.**
> voipvoip.com/nn<http://nn:xxx...@sip3.voipvoip.com/nn>
> [outgoing]
> secret  = xx
> username= nn
> ...
>
>
>  I have not configured anything other then entries in the sip.conf
>>
>
> I used your credentials and successfully placed a call to all of my
> Caribbean premium numbers*.
>
> Please change your password. Maybe your issue lies elsewhere. Does
> enabling SIP debugging on the console yield any clues?
>
> *) just kidding.
>
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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[asterisk-users] sip registration

[Thomas Perron]

I have a very lite layout and attempting to get the SIP configuration set
up initially before proceeding into other areas.

VMware is running my Asterisk 11 on Ubuntu 12.

Shouldnt I be able to at least ping the SIP provider IP?
I run command "sip show registry" and do not see it set up.
I run sip show peers and I do see an entry.

I have not configured anything other then entries in the sip.conf

results are:

Name/username HostDyn
Forcerport ACL Port Status  Description
outgoing/5552530146 (your
69.90.209.57   5060 OK (85
ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0
offline]
Asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
0 SIP registrations.
Asterisk*CLI>


my config is this:

[outgoing]
username=5552530146 (your VoIP VoIP account assigned while signing up)
type=peer
qualify=yes
secret=iblockedthis (your VoIP VoIP password)
nat=auto
insecure=invite,port
host=sip3.voipvoip.com
fromuser=5552530146 (your VoIP VoIP account assigned while signing up)
fromdomain=sip3.voipvoip.com
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
;
;
;
;
;
;register => 5552530146:7036361399@69.90.209.57/5552530146
register=>5552530146:boston!@#1...@sip3.voipvoip.com/5552530146
;


Please send input or guidance...

Thanks
Thomas
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[asterisk-users] Uprading to Asterisk 11 issues

[Thomas Thomas]

Hello,

I installed Asterisk 11 via the following command
*> svn co http://svn.asterisk.org/svn/asterisk/branches/11*
(as written in asteriskdocs.org
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id240883.html
)

But it seems that I have a development version instead of a stable release:
*> core show version*
*Asterisk SVN-branch-11-r375559 built by user @ user-MS-6580 on a i686
running Linux on 2012-11-01 13:05:50 UTC*

Did I do something wrong ?

Secondly, my logs from Verbose() application are no longer shown. I know
that the logging system has been changed but what shall I do to see my logs
(through asterisk -rv) ?
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Re: [asterisk-users] Read sometimes disconnects user

[Thomas Thomas]

My last message was wrong :

If the user doesn't enter any digits within 15 seconds, the file is played
again and a second attempt is made. So the timeout parameter specified for
Read() works.

But if the user enters the numbers, he might get disconnected for some
obscure reason. If he enters "04755" and takes too long before entering the
following number (timeout digit being set to 2 seconds), then he should
anyway take "04755" as the number and continue the script. It is definitily
very strange that the user gets disconnected. And that might happen when he
entered his 5 first digits, 9 digits, ... seems random.



2012/10/30 Thomas Thomas 

> But even then, it should do another attempt, play the file again and
> accept DMTF input, as I specified that there should be 2 attempts.
>
> *same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*
> *
> *
> 2012/10/30 Steve Edwards 
>
>> On Tue, 30 Oct 2012, Thomas Thomas wrote:
>>
>>  I am asking the user to enter his mobile phone followed by "#" using
>>> Read(). From time to time the Read() application disconnects the user while
>>> he is typing his number, though there is a 15 seconds timeout, and even if
>>> I type the number very fast it still may happen to me.
>>>
>>
>> It has been my casual observation that the speed at which I enter digits
>> on my phone is unrelated to the speed at which my cell provider delivers
>> the digits to my Asterisk box.
>>
>> --
>> Thanks in advance,
>> --**--**
>> -
>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>> Newline  Fax: +1-760-731-3000
>>
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>>   
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>>
>
>
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Re: [asterisk-users] Read sometimes disconnects user

[Thomas Thomas]

But even then, it should do another attempt, play the file again and accept
DMTF input, as I specified that there should be 2 attempts.

*same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*
*
*
2012/10/30 Steve Edwards 

> On Tue, 30 Oct 2012, Thomas Thomas wrote:
>
>  I am asking the user to enter his mobile phone followed by "#" using
>> Read(). From time to time the Read() application disconnects the user while
>> he is typing his number, though there is a 15 seconds timeout, and even if
>> I type the number very fast it still may happen to me.
>>
>
> It has been my casual observation that the speed at which I enter digits
> on my phone is unrelated to the speed at which my cell provider delivers
> the digits to my Asterisk box.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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[asterisk-users] Read sometimes disconnects user

[Thomas Thomas]

Hello,

I am asking the user to enter his mobile phone followed by "#" using Read().
>From time to time the Read() application disconnects the user while he is
typing his number, though there is a 15 seconds timeout, and even if I type
the number very fast it still may happen to me.

*same => n,Read(mobileNumber,app/input-mobile,10,,2,15)*

In the logs:
When it fails:
- -  Playing 'app/input-mobile.slin' (language 'fr')
- - User disconnected

When it succeeds:
- -  Playing 'app/input-mobile.slin' (language 'fr')
- - User entered '0476123456'

The strange thing is that I cannot understand when it happens, it seems
completely random...
sometimes I type the mobile number very slowly, sometimes very fast,
and it doesn't seem to matter.

Anyone encountered such an error? Thank you for any ideas!
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Re: [asterisk-users] RTP IP re-write

[Thomas Kenyon]

Joshua Colp wrote:

Thomas Kenyon wrote:

I am having a problem trying to get a particular softphone working on my
setup.

The machine it runs on has more than one interface. When the softphone
registers, it registers fine, and asterisk is given the correct IP for
registration.

Whenever RTP is set-up however, the client gives the wrong IP to connect
to and I get the inevitable problem with one-way media.

Is there any way of forcing that SIP account to have the rtp always sent
to a particular IP. (I know that this still may not work, because the
device is probably listening on the wrong interface as well, but it's
worth a try).


It's not possible to do this as you describe but if you set "nat=yes" 
the RTP module will lock on to the source of the incoming media after a 
certain number of packets. This does require that the softphone send 
packets to Asterisk and that they make it, of course.


Cheers,


Thanks, works perfectly :-)

I should have known that.

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[asterisk-users] RTP IP re-write

[Thomas Kenyon]

I am having a problem trying to get a particular softphone working on my 
setup.


The machine it runs on has more than one interface. When the softphone 
registers, it registers fine, and asterisk is given the correct IP for 
registration.


Whenever RTP is set-up however, the client gives the wrong IP to connect 
to and I get the inevitable problem with one-way media.


Is there any way of forcing that SIP account to have the rtp always sent 
to a particular IP. (I know that this still may not work, because the 
device is probably listening on the wrong interface as well, but it's 
worth a try).


I haven't been able to get a response from the vendor of the softphone.


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Re: [asterisk-users] Peculiar problem with failover provision.

[Thomas Kenyon]

Eric Wieling wrote:

You are doing it wrong.  I know 50 bazillion Asterisk dialplan examples on the 
internet do it the same way.  It is still wrong.

When you do a Dial on the dialplan you need check the value of DIALSTATUS or 
HANGUPCAUSE before dialing again.  Both variables will give you some indication 
of why the first call ended.  Then your dialplan logic can decide how to 
proceed.


Thanks for your help.

In previous versions of asterisk it worked, and iirc after the called 
party hung up, the dialplan only progressed if there was a particular 
flag used with Dial (g?).


It's going to cause a heck of a headache but I'll look into doing this 
properly in the week.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon
Sent: Monday, September 24, 2012 7:00 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Peculiar problem with failover provision.

I have noticed a peculiar problem recently with the way that the failover 
operates in my dialplan.

I normally have:

1,Dial(SIP//extension)
n,Dial(SIP//extension)

(or something similar).

This has up until now worked flawlessly.

If there is an error with the first provider, the call is completed with the 
second one.

Now, what is happening is, if the remote party hags up first, then the call 
progresses to the next priority and re-dials them.

Is this a change in default behaviour?
Do I need to add a particular flag / config directive to my dialplan

I am running Asterisk 10.6.0.

Thanks for any help in solving this.

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[asterisk-users] Peculiar problem with failover provision.

[Thomas Kenyon]

I have noticed a peculiar problem recently with the way that the 
failover operates in my dialplan.


I normally have:

1,Dial(SIP//extension)
n,Dial(SIP//extension)

(or something similar).

This has up until now worked flawlessly.

If there is an error with the first provider, the call is completed with 
the second one.


Now, what is happening is, if the remote party hags up first, then the 
call progresses to the next priority and re-dials them.


Is this a change in default behaviour?
Do I need to add a particular flag / config directive to my dialplan

I am running Asterisk 10.6.0.

Thanks for any help in solving this.

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[asterisk-users] Rookie / sip and extensions

[Thomas Perron]

Sorry for blasting another desperate note but I am trying!   I have changed
the username and password and IP to protect my system.
But, the logic is unchanged.  It is does not work!  I simply want to dial
the telephone number provided to me for my DID which corresponds with my
SIP info.
And, then it should connect and hit the "incoming" context and simply dial
the 617 number.   I am close but no cigar.  Now I get a fast busy tone only.

What is missing or what is needed please?

extensions.conf
[globals]

;
;
[incoming]
;
;exten=> s,1,Goto(125010155_incoming)
;
;[125010155_incoming]
exten => s,1,Answer
exten => s,n,Dial(SIP/16175551212)


sip.conf
[general]
;register => 125010155:funnyti...@sip3.voipvoip.com/125010155
register => 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11
;
[incoming]
username=125010155
type=peer
secret=funnytiger
nat=auto
insecure=invite,port
host=69.90.209.11
fromdomain=69.90.209.11
dtmfmode=rfc2833
context=incoming
allow=g729
allow=ulaw
allow=alaw
allow=ilbc
srvlookup=yes
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Re: [asterisk-users] sip and extensions

[Thomas Perron]

Hi,
I changed these codes to not coincide with actual account info.
Thanks


On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson  wrote:

> - Original Message -
>
> > I am new. Here is the code that I am playing with on CentOS 6.x
>
> > register => 5552530146:funnytiger...@sip3.voipvoip.com
>
> > [outgoing]
> > username=5552530146
> > type=peer
> > qualify=yes
> > secret=funnytiger123
> > nat=auto
> > insecure=very
> > host=69.90.209.57
> > fromuser=5552530146
> > fromdomain=69.90.209.57
> > dtmfmode=rfc2833
> > allow=g729
> > allow=ilbc
> > allow=ulaw
> > allow=alaw
> > disallow=all
> > srvlookup=no
>
> > [incoming]
> > username=5552530146
> > type=user
> > secret=funnytiger123
> > nat=auto
> > insecure=very
> > host=69.90.209.57
> > fromdomain=69.90.209.57
> > dtmfmode=rfc2833
> > context=incoming
> > allow=g729
> > allow=ulaw
> > allow=alaw
> > allow=ilbc
> > disallow=all
> > srvlookup=no
>
>
> *PLEASE* if that is your real username/password with your VoIP provider
> change it immediately. Just FYI, you've broadcast it to (tens or hundreds
> of) thousands of list readers. I have to believe some are of the nefarious
> type that would love to use your account for free calling at your expense.
> :/
>
> --Tim
>
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[asterisk-users] sip and extensions

[Thomas Perron]

I am new.  Here is the code that I am playing with on CentOS 6.x

When I dial the number that corresponds w/ my SIP account I get a
recording:  "reached a non-working number"

I built Asterisk a few times last year and am now back working on a similar
project.   In my view, there is something wrong in sip.conf
I don't remember using a file that long to get a basic call set up.  The
format was provided to me by voipvoip.com (the SIP provider).

Does anyone have any comments please?  I just want a very simple config to
get my machine to recognize a call to the SIP provider.

Here is results of sip show registry:

Hostdnsmgr Username   Refresh
StateReg.Time
sip3.voipvoip.com:5060  N  5552530146 285
Registered   Thu, 05 Jul 2012 21:39:56
1 SIP registrations.

Here is sip and extensions.conf

sip.conf

[general]
register => 5552530146:funnytiger...@sip3.voipvoip.com
;

[sip3.voipvoip.com]

[outgoing]
username=5552530146
type=peer
qualify=yes
secret=funnytiger123
nat=auto
insecure=very
host=69.90.209.57
fromuser=5552530146
fromdomain=69.90.209.57
dtmfmode=rfc2833
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
disallow=all
srvlookup=no

[incoming]
username=5552530146
type=user
secret=funnytiger123
nat=auto
insecure=very
host=69.90.209.57
fromdomain=69.90.209.57
dtmfmode=rfc2833
context=incoming
allow=g729
allow=ulaw
allow=alaw
allow=ilbc
disallow=all
srvlookup=no



extensions.conf

[general]

;
;
[incoming]
;first creating extensions for your local users
exten=> s,1,Dial(SIP/1703717)
exten=> s,2,Hangup()
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[asterisk-users] basic sip quesiton

[Thomas Perron]

What am I missing please?   sip show registry shows that I am registered.

[general]
register => 5552530146:tam...@sip3.voipvoip.com
;
;
[sip3.voipvoip.com]
bindport=5060 ;you can use different port if the default is blocked
bindaddr=0.0.0.0 ;binds to all



;this is for codec negotiation between the useragent and asterisk
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm



context=incoming ;default context where incoming calls are passed. this
should be the context where your local user.s extensions reside

[outbound-trunk]
;this is the second section of you sip.conf file. Here you can create your
trunk through which you will throw your outgoing calls to axvoice.
host=sip3.voipvoip.com
type=peer
dtmfmode=rfc2833
canreinvite=yes
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Re: [asterisk-users] port 5060 is blocked by ISP

[Thomas Kenyon]

alok srivastava wrote:

dear
i have configured properly asterisk. At the one end i am using x-lite 
soft ph and another end twinkle. call is going properly from both end 
but after picking the phone not able to listen other one.
when i checked the port 5060 on the asterisk server it is always showing 
closed while i have flushed all the rules from iptables (iptables -F)


PORT STATE  SERVICE VERSION
5060/tcp closed sip

 telnet localhost 5060 (could not connect)

regards
alok


SIP is only used to setup (and stop etc.) the call. The actual audio is 
sent via rtp.


/etc/asterisk/rtp.conf

Should tell which ports asterisk is using for rtp, you will need to make 
sure that the remote host can connect to these ports.


There are lots of articles around on how to resolve this.



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Re: [asterisk-users] unable to create channel of type 'SIP'

[James Thomas]

I think you need to change:
exten => 2012,1,Macro(dialSIP,IMSI262428511722625)
exten => 2013,1,Macro(dialSIP,IMSI262422146099205)

to:
exten => 2012,1,Macro(dialGSM,IMSI262428511722625)
exten => 2013,1,Macro(dialGSM,IMSI262422146099205)

also what does sip show peers show, as opposed to sip show registry?


On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick wrote:

> I'm trying to use OpenBTS with Asterisk.
> I have two phones that are connecting to OpenBTS correctly, but on the
> Asterisk side the phones can't call each other.
>
> I followed this guide:
> http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk
> I set up two phones in sip.conf and extensions.conf.
>
> In my SIP output I see this:
> WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create
> channel of type 'SIP' (cause 20 - unknown)
>
> If I type sip show registry it says there are 0 SIP registrations.
> Should I see both the phones registered at this point?
> If that's what's wrong, what am I doing wrong that's making the phones
> not able to register?
>
> Below is my Asterisk configuration.
>
> Jacob
>
> #/etc/asterisk/sip.conf
> [general]
> context=sip-external
>
> #...
>
> [IMSI262428511722625]
> callerid=2012
> canreinvite=no
> type=friend
> context=sip-external
> allow=gsm
> host=dynamic
> dtmfmode=info
>
> [IMSI262422146099205]
> callerid=2013
> canreinvite=no
> type=friend
> context=sip-external
> allow=gsm
> host=dynamic
> dtmfmode=info
>
>
> #/etc/asterisk/extensions.conf
> [macro-dialGSM]
> exten => s,1,Dial(SIP/${ARG1})
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-CANCEL,1,Hangup
> exten => s-NOANSWER,1,Hangup
> exten => s-BUSY,1,Busy(30)
> exten => s-CONGESTION,1,Congestion(30)
> exten => s-CHANUNAVAIL,1,playback(ss-noservice)
> exten => s-CANCEL,1,Hangup
>
> [sip-external]
> exten => 2012,1,Macro(dialSIP,IMSI262428511722625)
> exten => 2013,1,Macro(dialSIP,IMSI262422146099205)
>
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[asterisk-users] Voicemail crashs asterisk

[Thomas Hoellriegel]

Hi guys,
I have the following problem:
My System: asterisk 1.8.11.0 on debian squeeze
I login to my mailbox from voicemailmain.
Once I am logged in, I get my number of messages announced.
I change the directory of old messages 2 times in a row: 2 1 2 1.
Asterisk exits completely.
Can your help please?
thanks.

---
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste für blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc
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[asterisk-users] Problem callerid ignored by using callfiles

[Thomas Hoellriegel]

hi all,
the same problem for a long time.
i upgraded now from 1.8.8.2 to 1.8.9.1.
I generate a callfile with the option:
Callerid: test Callback Service <00498925007676>

The callback is established correctly, but the variable ${CALLERID(num):} is 
empty

no  output: NoOp(Calleridnumber: ${CALLERID(num):} can be found.
I don.t find my "test Callback Service <00498925007676>" on the cli or as
DumpChan().
Under the 1.8.8.2 version still has it all works.
has something changed?

Can you help please?
thanks.




---
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste für blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc--
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Re: [asterisk-users] [SPAM] - FW by postmas...@datavox.co.uk : Delivery Status Notification (Failure) - Email found in subject

[Andrew Thomas]

Hi Michelle,

Sorry about this.  We've just moved servers and I forgot to turn NDR's
off!

Nick left us some time ago - but never deleted his subscription.  Can a
list Admin please remove him - as requested below?

BTW - it's n...@datavox.co.uk

Cheers
Andy
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Konzack
Sent: 04 February 2012 04:31
To: asterisk-users@lists.digium.com
Subject: [SPAM] - [asterisk-users] FW by postmas...@datavox.co.uk :
Delivery Status Notification (Failure) - Email found in subject

Can an ADMIN or Listmaster unsubscribe

n...@datavox.co.com

because his Mailserver is sending me huge error messages to my cellphone
du to server errors.

Thanks, Greetings and nice Day/Evening
Michelle Konzack

--
# Debian GNU/Linux Consultant ##
   Development of Intranet and Embedded Systems with Debian GNU/Linux
   Internet Service Provider, Cloud Computing


itsystems@tdnet Jabber  linux4miche...@jabber.ccc.de
Owner Michelle Konzack

Gewerbe Strasse 3   Tel office: +49-176-86004575
77694 Kehl  Tel mobil:  +49-177-9351947
Germany Tel mobil:  +33-6-61925193  (France)

USt-ID:  DE 278 049 239

Linux-User #280138 with the Linux Counter, http://counter.li.org/

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[asterisk-users] Sip.conf and extensions.conf configuration for Exchange 2010 U.M.

[James Thomas]

Hi All,

I'm using Exchange as our voicemail system. Everything works fine until the
1 week mark when Exchange changes the port number used, then Asterisk 1.8
seg faults and I have no phones (unless I restart the U.M. service before
the 1 week period is up). Since that is a hack, I'm hoping someone can post
their working configs that accomodates the port change. The documentation
I've seen is still a little unclear to me. I'm not using secured mode, so
just using ports 5065/5067.

Thanks for your help.

Jim
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[asterisk-users] Proper sip.conf and extensions.conf for Exchange 2010 U.M.

[James Thomas]

Hi All,

I'm using Exchange as our voicemail system. Everything works fine until the
1 week mark when Exchange changes the port number used, then Asterisk 1.8
seg faults and I have no phones (unless I restart the U.M. service before
the 1 week period is up). Since that is a hack, I'm hoping someone can post
their working configs that accomodates the port change. The documentation
I've seen is still a little unclear to me. I'm not using secured mode, so
just using ports 5065/5067.

TIA
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Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card

[Andrew Thomas]

The Debian command I use is:

apt-get install linux-headers-`uname -r`

That will get the bits you need and place them in /usr/src/.




 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)

[Andrew Thomas]

This is a brilliant idea.  How do I contribute my attackers to this
list?  

Cheers
Andy
 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: 22 September 2011 16:11
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP
Abuse)



Sounds like a great idea.. Hopefully the page/account never gets hacked
and bad IP's published.. I could see a great hack of 

127.0.0.1  

192.168.0.0/16 

10.0.0.0/8 

getting up there somehow and next thing you know - BAM!

 

But I haven't RTFM - I'm guessing there is probably a white list that
supersedes the naughty list.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, September 22, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP
Abuse)

 

very cool!

On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo 
wrote:


Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.

For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.

So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.

http://twitter.com/#!/voipabuse

Now, you can concoct a quick script off of it, e.g.:

links -dump "http://twitter.com/voipabuse";|awk '/attacker/{print
"iptables -A INPUT -s "$2" -j DROP"| "sort -u"}'

Will get a quickie soon from my Acme's, nCites, etc. when I have time.

For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)



--

=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM

"It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently." - Warren Buffett

42B0 5A53 6505 6638 44BB  3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x2BF7D83F210A95AF


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 If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments. 

Registered in England. No. 27459085.



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Re: [asterisk-users] SNMP problem

[Robert Thomas]

Did you copy the asterisk-mib.txt and digium-mib.txt to the proper folder on
your distro?

I see people forgetting about that step.

On Wed, Sep 14, 2011 at 2:18 AM, Ishfaq Malik  wrote:

> Hi
>
> I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as
> my resource
>
> http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html
>
> when I execute the following command
>
> snmpwalk -On -v2c -c public 127.0.0.1 .1.3.6.1.4.1.22736
>
> I get the following response
>
> .1.3.6.1.4.1.22736 = No Such Object available on this agent at this OID
>
> The contents of my /etc/snmp/snmpd.conf is exactly like the book
> instructs
>
> com2sec notConfigUser  default   public
>
> group   notConfigGroup v1   notConfigUser
> group   notConfigGroup v2c   notConfigUser
>
> view allincluded  .1
> view system included  .iso.org.dod.internet.mgmt.mib-2.system
>
> access  notConfigGroup ""  any   noauthexact  allnone none
>
> master agentx
> agentXSocket /var/agentx/master
> agentXPerms 0660 0775 nobody root
>
> sysObjectID .1.3.6.1.4.1.22736.1
>
> Does anyone have any pointers as to where I'm going wrong?
>
> Thanks in advancde
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office:   0161 660 3062
>
>
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[asterisk-users] Sip profiles per customer, behind a SIP proxy. How?

[Robert Thomas]

Hello List,

I have been trying to configure  a sip profile ( peer / friend ) for each of
my customers behind a sip proxy for some time, but I have had no success, so
I would appreciate your help.

Customer  -> OpenSIPS -> Asterisk -> PSTN

The opensips is working as a sip proxy with record route, for billing, load
balancing and authentication purposes.

I would like to be able to define a particular context, or settings per
customer. Since the customer is behind the SIP proxy, all I see if the
packet comming from the SIP proxy. So I have created a peer profile with the
IP Address of my proxy. Problem been any setting I enable affect all
traffic coming throught the SIP proxy.

I was reading that Asterisk checks the SIP From: address username and
matches against names of devices with type=user-

However I have some problems with Asterisk 1.6.2, taking the caller id
either from the RPID so I manually parse the PAI header.

I was thinking about replacing the From with a customer ID, and for those
customers that use the FROM to signal caller id, to copy it over the PAI
header at the SIP proxy.

I don't know if overwriting the FROM would cause any problem with the SIP
clients behind the proxy.

Is there any better or different way to acomplish this?

I would say we should have some flexibility although we have a SIP proxy as
the source IP for all of our traffic.


Robert
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Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer

[Mark G Thomas]

Hi,

This is still broken in 1.6.2.20. Please see below.

On Fri, May 06, 2011 at 04:27:42PM +, Watkins, Bradley wrote:
> >From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> >boun...@lists.digium.com] On Behalf Of Mark G Thomas
> >Sent: Friday, May 06, 2011 12:02 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U()
> >option in 1.6.2.17.2 and newer
> >
> >Hi,
> >
> >On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote:
> >> On 11-05-05 05:14 PM, Mark G Thomas wrote:
> >> >Hi,
> >> >
> >> >I think this must be a bug introduced with 1.6.2.17.something.
> >> >
> >> >When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or
> >> >1.6.2.18, my AEL Dial() commands with the "U" options fail with the
> >following error:
> >> >
> >> >[May  3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-
> >existent
> >> > destination for gosub: (Context:screen, Extension:s, Priority:1)
> >> >
> >> You might want to have a look at:
> >> https://issues.asterisk.org/view.php?id=18910
> >
> >Thanks. This is it.
> >
> >If I'm reading this right, it describes the change which broke things for me,
> >but no solution applicable to my Dial() command U flag, which is invoking my
> >AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it
> >either.
> >
> >It sure seems to me this change to AEL has had unexpected consequences in
> >terms of breaking things in dialplans.
> >
> 
> I was under the impression that this had been fixed, although perhaps it's 
> not yet in a release.  Is there a chance you try with the latest 1.6.2 branch 
> from SVN?
> 
> - Brad

In AEL, Dial() with the U flag is still broken. 

Reverting to a pre-1.6.2.17.2 pval.c fixes the problem.

[Aug  8 13:36:01] ERROR[24608]: app_stack.c:402 gosub_exec: Attempt to reach a 
non-existent destination for gosub: (Context:screen, Extension:s, Priority:1)

My AEL dialplan segment:
--
context internals {
102 => {
Dial(${MARKCELL},30,tgU(screen));
jump s@home-menu;
};
};

macro screen() {
Wait(0.5);
Read(ACCEPT,followme/options,1,,1,20);
if( "${ACCEPT}" = "1" ) {
Background(connecting);
} else {
Set(GOSUB_RESULT=CONTINUE);
};
return;
};

Here's the dialplan it created under 1.6.2.20, but U(screen)
in Dial() tries to call screen,s which doesn't exist.
---
[ Context 'internals' created by 'pbx_ael' ]
  '102' =>  1. Dial(${MARKCELL},30,tgU(screen))   [pbx_ael]
2. Goto(home-menu,s,1)[pbx_ael]

[ Context 'screen' created by 'pbx_ael' ]
  '~~s~~' =>1. Wait(0.5)  [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20)  [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_25)  [pbx_ael]
8. Return()   [pbx_ael]


This works, from an earlier version, before the pval.c change:

[ Context 'screen' created by 'pbx_ael' ]
  's' =>1. Wait(0.5)  [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20)  [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_25)  [pbx_ael]
8. Return()   [pbx_ael]



-- 
Mark G. Thomas (m...@misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112

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Re: [asterisk-users] Queue Issue : Duration between 2 agents call

[Florent THOMAS]

Hy,

I checked so many parameters that I can't see anymore a solution to this.
The duration is always of 5sec.
Is there a config on hardphones that could freeze the process,
Does the log could help you to identify the origin of this?

Regards

Le 11/07/2011 09:38, Ishfaq Malik a écrit :

What have you set the retry parameter for this queue?

On Sun, 2011-07-10 at 13:04 +0200, Florent THOMAS wrote:

Hy,

I'm currently working with one queue and whatever I change in the
config, it stills a gap of 6 seconds during which no agents are
ringing for this queue.
Is ther any parameter to configure there?

regards
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Re: [asterisk-users] Queue Issue : Duration between 2 agents call

[Florent THOMAS]

;Local/5034@from-queue-0c79;2", "0?nodial") 
in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:9] GotoIf("Local/5034@from-queue-0c79;2", 
"0?continue") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:10] Set("Local/5034@from-queue-0c79;2", 
"EXTHASCW=ENABLED") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:11] GotoIf("Local/5034@from-queue-0c79;2", 
"0?next1:cwinusebusy") in new stack

[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (macro-dial-one,s,23)
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:23] GotoIf("Local/5034@from-queue-0c79;2", 
"1?next3:continue") in new stack

[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Goto (macro-dial-one,s,24)
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:24] ExecIf("Local/5034@from-queue-0c79;2", 
"0?Set(DIALSTATUS_CW=BUSY)") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:25] GotoIf("Local/5034@from-queue-0c79;2", "0?nodial") 
in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[s@macro-dial-one:26] GosubIf("Local/5034@from-queue-0c79;2", 
"1?dstring,1:dlocal,1") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:1] Set("Local/5034@from-queue-0c79;2", 
"DSTRING=") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:2] Set("Local/5034@from-queue-0c79;2", 
"DEVICES=5034") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:3] ExecIf("Local/5034@from-queue-0c79;2", 
"0?Return()") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:4] ExecIf("Local/5034@from-queue-0c79;2", 
"0?Set(DEVICES=034)") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:5] Set("Local/5034@from-queue-0c79;2", 
"LOOPCNT=1") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:6] Set("Local/5034@from-queue-0c79;2", "ITER=1") 
in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:7] Set("Local/5034@from-queue-0c79;2", 
"THISDIAL=SIP/5034") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[dstring@macro-dial-one:8] GosubIf("Local/5034@from-queue-0c79;2", 
"1?zap2dahdi,1") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[zap2dahdi@macro-dial-one:1] ExecIf("Local/5034@from-queue-0c79;2", 
"0?Return()") in new stack
[Jul 15 14:30:26] VERBOSE[25233] pbx.c: -- Executing 
[zap2dahdi@macro-dial-one:2] Set("Local/5034@from-queue-0c79;2", 
"NEWDIAL=") in new stack/



I stil have a delay between 2 agents.

Hope this log could give some ideas to anybody

Regards
Le 13/07/2011 12:41, Florent THOMAS a écrit :

Hy,

I still struggle with this issue, does anybody can help me?

Regards

Le 10/07/2011 13:04, Florent THOMAS a écrit :

Hy,

I'm currently working with one queue and whatever I change in the 
config, it stills a gap of 6 seconds during which no agents are 
ringing for this queue.

Is ther any parameter to configure there?

regards


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Re: [asterisk-users] Queue Issue : Duration between 2 agents call

[Florent THOMAS]

Hy,

I still struggle with this issue, does anybody can help me?

Regards

Le 10/07/2011 13:04, Florent THOMAS a écrit :

Hy,

I'm currently working with one queue and whatever I change in the 
config, it stills a gap of 6 seconds during which no agents are 
ringing for this queue.

Is ther any parameter to configure there?

regards


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Re: [asterisk-users] Queue Issue : Duration between 2 agents call

[Florent THOMAS]

Of course I would like to say that the problem is not solved.

regards

Le 11/07/2011 13:11, Florent THOMAS a écrit :

Satish and IchFaq, thank yor for answering so fast.

I checked the queue conf and the retry is put on zero :
/[0860]
announce-frequency=30
announce-holdtime=no
announce-position=yes
autofill=no
eventmemberstatus=no
eventwhencalled=no
joinempty=yes
leavewhenempty=no
maxlen=0
memberdelay=0
music=default
penaltymemberslimit=0
periodic-announce-frequency=0
queue-callswaiting=queue-callswaiting
queue-thankyou=queue-thankyou
queue-thereare=queue-thereare
queue-youarenext=queue-youarenext
reportholdtime=no
/retry=0/
ringinuse=yes
servicelevel=60
strategy=linear
timeout=15
timeoutpriority=app
timeoutrestart=no
weight=0
wrapuptime=0
member=Local/5030@from-queue/n,0,toto
member=Local/5034@from-queue/n,0,tata
member=Local/5032@from-queue/n,0,titi
/
Le 10/07/2011 13:04, Florent THOMAS a écrit :

Hy,

I'm currently working with one queue and whatever I change in the 
config, it stills a gap of 6 seconds during which no agents are 
ringing for this queue.

Is ther any parameter to configure there?

regards


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Re: [asterisk-users] Thomson ST022 - External Call problems

[Florent THOMAS]

Dear Lali,

I'll try your proposal ASAP.

Regards

Le 10/07/2011 10:52, Florent THOMAS a écrit :

Hy all of you,

I've successfully installed a freepbx solution with 10 extensions :
- 5 on Linksys SPA922
- 1 on Linksys SPA942
- 1 on Thomson ST022

Everything seems to work fine with all the hardphones excepts last week.
The thomson has a strange behaviour. It can reach french mobile cell 
phones but when it reaches "fix" phones, the correspondant can't hear 
the caller.
What is very stange is that for incoming calls for this extension, 
everything works fine.


Thanks for your help,

Regards


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Re: [asterisk-users] Queue Issue : Duration between 2 agents call

[Florent THOMAS]

Satish and IchFaq, thank yor for answering so fast.

I checked the queue conf and the retry is put on zero :
/[0860]
announce-frequency=30
announce-holdtime=no
announce-position=yes
autofill=no
eventmemberstatus=no
eventwhencalled=no
joinempty=yes
leavewhenempty=no
maxlen=0
memberdelay=0
music=default
penaltymemberslimit=0
periodic-announce-frequency=0
queue-callswaiting=queue-callswaiting
queue-thankyou=queue-thankyou
queue-thereare=queue-thereare
queue-youarenext=queue-youarenext
reportholdtime=no
/retry=0/
ringinuse=yes
servicelevel=60
strategy=linear
timeout=15
timeoutpriority=app
timeoutrestart=no
weight=0
wrapuptime=0
member=Local/5030@from-queue/n,0,toto
member=Local/5034@from-queue/n,0,tata
member=Local/5032@from-queue/n,0,titi
/
Le 10/07/2011 13:04, Florent THOMAS a écrit :

Hy,

I'm currently working with one queue and whatever I change in the 
config, it stills a gap of 6 seconds during which no agents are 
ringing for this queue.

Is ther any parameter to configure there?

regards


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