[asterisk-users] tcptls ssl connection error

[Chandrakant Solanki]

Hello All,

I am using asterisk 1.8.13.0 and which is running on TLS port and my
request forwarded from opensips which is also run tls port.

On both end my certificate is same.

During search about this error, I found below blog and apply patch, then
also found below error.

https://issues.asterisk.org/jira/browse/ASTERISK-18345
https://issues.asterisk.org/jira/browse/ASTERISK-20559
Also applied r375023

[Nov  8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown()
failed: 5
[Nov  8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown()
failed: 5
[Nov  8 21:57:37]   == Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
[Nov  8 21:57:37] WARNING[19274]: tcptls.c:251 handle_tcptls_connection:
FILE * open failed!
[Nov  8 21:57:39]   == Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
[Nov  8 21:57:39] WARNING[19356]: tcptls.c:251 handle_tcptls_connection:
FILE * open failed!
[Nov  8 21:57:49]   == Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
[Nov  8 21:57:49] WARNING[19357]: tcptls.c:251 handle_tcptls_connection:
FILE * open failed!


-- 
Regards,

Chandrakant Solanki
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[asterisk-users] 回覆︰ problem on LDAP (Invalid credential)

[kingman chui]

I use res_ldap.conf to connect lync server by ldap ebfore.
For auth in lync server .
The username in res_lap.conf cannot use format "cn=xx,dc=xxx" .
It should use format username=@x and passwd.
Then lync server can auth you .
I try before and it work in asterisk 1.8.11
 
Hope it can help you ..
 
Regard/chui king man

寄件人︰ Samira Hosseini 
>收件人︰ "asterisk-users@lists.digium.com"  
>傳送日期︰ 2012年11月9日 (週五) 1:39 AM
>主題︰ [asterisk-users] problem on LDAP (Invalid credential)
>
>
>
>
>
>>
>>
>>Hello all,
>>
>>
>>I am going to register asterisk sip users through active directory accounts 
>>LDAP (that is a separated server with ip : 192.168.11.17)
>>So I have followed the below link as well:
>>
>>
>>https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
>>
>>http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
>>
>>http://ensiwiki.ensimag.fr/index.php/Asterisk's_external_configuration_(LDAP)
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>Server:192.168.14.90  => asterisk
>>server:192.168.11.17 =>  ActiveDirectory
>>Finally, this is my configuration file :
>>
>>
>>[root@PBX ~]# telnet 192.168.11.17 389
>>Trying 192.168.11.17...
>>Connected to 192.168.11.17 (192.168.11.17).
>>Escape character is '^]'.
>>
>>
>>[_general]
>>host=192.168.11.17    ; LDAP host
>>port=389
>>protocol=3           ; Version of the LDAP protocol to use; default is 3.
>>url=ldap://192.168.11.17:389
>>basedn=dc=example,dc=com
>>;User=cn=,dc=example,dc=com
>>;User=cn=join_lan,dc=example,dc=com
>>;User=cn=sa_hosseini,dc=rasana,dc=ir
>>User=cn=lan,cn=technical,cn=xyz,cn=join_lan,dc=example,dc=com
>>Pass=123456
>>
>>---
>>vim /etc/asterisk/extconfig.conf
>>
>>sipusers => ldap,"dc=example,dc=com",sip
>>
>>
>>
>>vim /etc/asterisk/sip.conf
>>[general]
>>callevents=yes
>>rtcachefriends=yes
>>
>>
>>
>>
>>but i got the follwoing error :
>>
>>
>>
>>
>>
>>
>>
>>PBX*CLI> module reload res_config_ldap.so
>>    -- Reloading module 'res_config_ldap.so' (LDAP realtime interface)
>>  == Parsing '/etc/asterisk/res_ldap.conf':   == Found
>>[Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1750 ldap_reconnect: bind 
>>failed: Invalid credentials
>>[Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1598 reload: Couldn't 
>>establish connection to your directory server. Check debug.
>>  == LDAP RealTime driver reloaded.
>>
>>
>>Then i have registered with user:join_lan;pass:123456 domain:192.168.14.90
>>and get the following error on CLI:
>>Verbosity is at least 15
>>[Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register: 
>>Registration from '"join_lan"' failed for 
>>'192.168.19.21:38968' - No matching peer found
>>[Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register: 
>>Registration from '"join_lan"' failed for 
>>'192.168.19.21:38968' - No matching peer found
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
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Re: [asterisk-users] Managing complex setups with Asterisk

[Paul Belanger]

On 12-11-08 01:41 AM, martin f krafft wrote:

also sprach Paul Belanger  [2012.11.07.2340 
+0100]:

What is your point of pain? Right now we do most of the
configuration, provisioning, and system management outside of
asterisk.


My systems are already managed automatically, thankfully no longer
with Puppet. ;)

I am only talking about configuration of Asterisk, whether in
/etc/asterisk or some sensible external data source. My point of
pain is the complexity due to a couple of special cases, e.g.

   - Roaming users, i.e. no 1:n relation between sites and users;
   - Multiple devices per user (some want them all to ring, some want
 individual extensions but shared voicemail, …)
   - Keeping track of the mappings between incoming calls (from SIP
 providers) and extensions to ring (using incoming contexts and
 extension groups for that)
   - Keeping track of which extension uses which outgoing trunk
   - …

With a logical naming scheme, a policy and include files, this is
all working. But it's very error-prone and there is a bit of
redundancy in the information, so I was wondering if there wasn't
a better way.


Either way, don't manually build your 6th machine.  Start from
fresh using some sort of automated tool (chef / puppet).  This
will help you get on the right path.


The new machine for the 6th site is up and running (provisioning
(not image-based) took less than half an hour). What now? ;)


Then you are on the right path.

Either way, it sounds like you need to store your data some place and 
start building it out.  I don't know of any existing tools to do that, 
and I'm in the same boat.  I have everything I want / need managed by 
puppet, but more dynamic data needs to be moved out into something else.


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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Re: [asterisk-users] How to propagate NOANSWER up through a Local channel?

[Tony Mountifield]

In article <20121108092952.78cb6...@ws78.int.tlc>,
Chad Wallace  wrote:
> On Thu, 8 Nov 2012 16:44:32 + (UTC)
> t...@softins.co.uk (Tony Mountifield) wrote:
> 
> > Here is a simplified example:
> > 
> > [test]
> > exten => _X.,1,Dial(Local/${EXTEN}@outbound)
> > exten => _X.,n,NoOp(${CONTEXT}:DIALSTATUS=${DIALSTATUS})
> > 
> > [outbound]
> > exten => _X.,1,Dial(SIP/ext${EXTEN},30)
> > exten => _X.,n,NoOp(${CONTEXT}:DIALSTATUS=${DIALSTATUS})
> > 
> > So if I don't answer within 30 sec, I see outbound:DIALSTATUS=NOANSWER
> > but test:DIALSTATUS=CHANUNAVAIL
> > 
> > If instead I put the timeout on the outer Dial instead, I see
> > test:DIALSTATUS=NOANSWER and outbound:DIALSTATUS=CANCEL, because the
> > Local channel hung up the inner Dial while it was still ringing.
> > 
> > So I understand the reasons for the above behaviours, but my question
> > is: How can I propagate the NOANSWER status upwards from the inner
> > Dial, so that the Local channel also returns NOANSWER?
> 
> Try Hangup(NO_ANSWER) if DIALSTATUS is NOANSWER after the Dial in the
> Local channel.

Thanks! Just need to get my platform updated from 1.2 :(

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] problem on LDAP (Invalid credential)

[Christopher Harrington]

Posting your email again after 12 hours is not going to make anyone more
likely to help you. Please don't do that.


On Thu, Nov 8, 2012 at 11:39 AM, Samira Hosseini
wrote:

>
>
>
> Hello all,
>
> I am going to register asterisk sip users through active directory
> accounts LDAP (that is a separated server with ip : 192.168.11.17)
> So I have followed the below link as well:
>
> https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
>
> http://ensiwiki.ensimag.fr/index.php/Asterisk's_external_configuration_(LDAP)
>
>
>
>
>
> Server:192.168.14.90  => asterisk
> server:192.168.11.17 =>  ActiveDirectory
> Finally, this is my configuration file :
>
> [root@PBX ~]# telnet 192.168.11.17 389
> Trying 192.168.11.17...
> Connected to 192.168.11.17 (192.168.11.17).
> Escape character is '^]'.
>
> [_general]
> host=192.168.11.17; LDAP host
> port=389
> protocol=3   ; Version of the LDAP protocol to use; default is 3.
> url=ldap://192.168.11.17:389
> basedn=dc=example,dc=com
> ;User=cn=,dc=example,dc=com
> ;User=cn=join_lan,dc=example,dc=com
> ;User=cn=sa_hosseini,dc=rasana,dc=ir
> User=cn=lan,cn=technical,cn=xyz,cn=join_lan,dc=example,dc=com
> Pass=123456
> ---
> vim /etc/asterisk/extconfig.conf
> sipusers => ldap,"dc=example,dc=com",sip
> 
> vim /etc/asterisk/sip.conf
> [general]
> callevents=yes
> rtcachefriends=yes
>
>
> but i got the follwoing error :
>
>
>
> PBX*CLI> module reload res_config_ldap.so
> -- Reloading module 'res_config_ldap.so' (LDAP realtime interface)
>   == Parsing '/etc/asterisk/res_ldap.conf':   == Found
> [Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1750 ldap_reconnect:
> bind failed: Invalid credentials
> [Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1598 reload: Couldn't
> establish connection to your directory server. Check debug.
>   == LDAP RealTime driver reloaded.
>
> Then i have registered with user:join_lan;pass:123456 domain:192.168.14.90
> and get the following error on CLI:
> Verbosity is at least 15
> [Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register:
> Registration from '"join_lan"' failed for '
> 192.168.19.21:38968' - No matching peer found
> [Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register:
> Registration from '"join_lan"' failed for '
> 192.168.19.21:38968' - No matching peer found
>
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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[asterisk-users] problem on LDAP (Invalid credential)

[Samira Hosseini]

>  
>
>Hello all,
>
>
>I am going to register asterisk sip users through active directory accounts 
>LDAP (that is a separated server with ip : 192.168.11.17)
>So I have followed the below link as well:
>
>
>https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
>
>http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
>
>http://ensiwiki.ensimag.fr/index.php/Asterisk's_external_configuration_(LDAP)
>
>
>
>
>
>
>
>
>
>
>
>Server:192.168.14.90  => asterisk
>server:192.168.11.17 =>  ActiveDirectory
>Finally, this is my configuration file :
>
>
>[root@PBX ~]# telnet 192.168.11.17 389
>Trying 192.168.11.17...
>Connected to 192.168.11.17 (192.168.11.17).
>Escape character is '^]'.
>
>
>[_general]
>host=192.168.11.17    ; LDAP host
>port=389
>protocol=3           ; Version of the LDAP protocol to use; default is 3.
>url=ldap://192.168.11.17:389
>basedn=dc=example,dc=com
>;User=cn=,dc=example,dc=com
>;User=cn=join_lan,dc=example,dc=com
>;User=cn=sa_hosseini,dc=rasana,dc=ir
>User=cn=lan,cn=technical,cn=xyz,cn=join_lan,dc=example,dc=com
>Pass=123456
>
>---
>vim /etc/asterisk/extconfig.conf
>
>sipusers => ldap,"dc=example,dc=com",sip
>
>
>
>vim /etc/asterisk/sip.conf
>[general]
>callevents=yes
>rtcachefriends=yes
>
>
>
>
>but i got the follwoing error :
>
>
>
>
>
>
>
>PBX*CLI> module reload res_config_ldap.so
>    -- Reloading module 'res_config_ldap.so' (LDAP realtime interface)
>  == Parsing '/etc/asterisk/res_ldap.conf':   == Found
>[Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1750 ldap_reconnect: bind 
>failed: Invalid credentials
>[Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1598 reload: Couldn't 
>establish connection to your directory server. Check debug.
>  == LDAP RealTime driver reloaded.
>
>
>Then i have registered with user:join_lan;pass:123456 domain:192.168.14.90
>and get the following error on CLI:
>Verbosity is at least 15
>[Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register: 
>Registration from '"join_lan"' failed for 
>'192.168.19.21:38968' - No matching peer found
>[Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register: 
>Registration from '"join_lan"' failed for 
>'192.168.19.21:38968' - No matching peer found
>
>
>
>
>
>
>
>
>
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Re: [asterisk-users] DAHDI 1.4 on Kernel 3.0

[Shaun Ruffell]

On Tue, Nov 06, 2012 at 06:49:09PM -0600, Alyed wrote:
> Hello listers,
> 
> I'm trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system,
> but have faced lots of problems mainly because it has lots of functions
> looking for the PCI.
> 
> Have seen so many problems, I'm in fact thinking it cannot be possibly done
> (at least not in a couple of weeks, by one only man). Has anyone out there
> had any experience on something like this? or can someone shed some light
> on how to overcome this issues?
> 
> Any ideas are very welcome

There isn't a 1.4 version of DAHDI. However version v2.6.0 will not
build any PCI drivers if the Kernel does not have the PCI bus
configured.

[1] http://svnview.digium.com/svn/dahdi?view=revision&revision=10397

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] How to propagate NOANSWER up through a Local channel?

[Chad Wallace]

On Thu, 8 Nov 2012 16:44:32 + (UTC)
t...@softins.co.uk (Tony Mountifield) wrote:

> Here is a simplified example:
> 
> [test]
> exten => _X.,1,Dial(Local/${EXTEN}@outbound)
> exten => _X.,n,NoOp(${CONTEXT}:DIALSTATUS=${DIALSTATUS})
> 
> [outbound]
> exten => _X.,1,Dial(SIP/ext${EXTEN},30)
> exten => _X.,n,NoOp(${CONTEXT}:DIALSTATUS=${DIALSTATUS})
> 
> So if I don't answer within 30 sec, I see outbound:DIALSTATUS=NOANSWER
> but test:DIALSTATUS=CHANUNAVAIL
> 
> If instead I put the timeout on the outer Dial instead, I see
> test:DIALSTATUS=NOANSWER and outbound:DIALSTATUS=CANCEL, because the
> Local channel hung up the inner Dial while it was still ringing.
> 
> So I understand the reasons for the above behaviours, but my question
> is: How can I propagate the NOANSWER status upwards from the inner
> Dial, so that the Local channel also returns NOANSWER?

Try Hangup(NO_ANSWER) if DIALSTATUS is NOANSWER after the Dial in the
Local channel.

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] Static on calls - v1.8.15.0

[Carlos Alvarez]

On Thu, Nov 8, 2012 at 9:34 AM, James Lamanna  wrote:

> Hi,
> I'm testing out a server with asterisk 1.8.15.0 on it.
> I'm experiencing static occurring on almost 90% of calls on this
> particular server.
> All test phones are using SIP, and calls to/from PSTN servers are
> delivered using IAX2.
>
> I have other production servers running 1.4.x that do not have this issue
> that use the same PSTN connections.
> I haven't seen any ethernet errors or anything like that. Load is minimal
> since this is still a test server.
> The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s.
>
> I'm sort of baffled as to where to start looking for the root cause of
> this issue, but it appears to be isolated to only this machine.
>

I would recommend you try SIP and lock out all CODECs on both sides except
ulaw.  See what happens.

We've had this problem both because of CODED mismatch and just due to IAX's
many many bugs.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Static on calls - v1.8.15.0

[James Lamanna]

On Thu, Nov 8, 2012 at 8:47 AM, Richard Mudgett  wrote:

> > I'm testing out a server with asterisk 1.8.15.0 on it.
> > I'm experiencing static occurring on almost 90% of calls on this
> > particular server.
> > All test phones are using SIP, and calls to/from PSTN servers are
> > delivered using IAX2.
> >
> >
> > I have other production servers running 1.4.x that do not have this
> > issue that use the same PSTN connections.
> > I haven't seen any ethernet errors or anything like that. Load is
> > minimal since this is still a test server.
> > The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s.
> >
> >
> > I'm sort of baffled as to where to start looking for the root cause
> > of this issue, but it appears to be isolated to only this machine.
>
> You might have an A-law/u-law mismatch in the audio path.  That
> kind of mismatch sounds like static on the line.
>
>
Hmm, would  translation from ulaw -> gsm cause that as well?
I noticed in 1.8 apparently iax2  allow=ulaw is off...

-- James
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Re: [asterisk-users] TE820 hardware detection

[Shaun Ruffell]

On Thu, Nov 08, 2012 at 09:02:43AM -0800, Justin Killen wrote:
> I checked the version and it looks like it is up to date
> 
> [root@dozer2 dahdi]# asterisk -rx 'dahdi show version'
> DAHDI Version: 2.6.1 Echo Canceller: HWEC
> [root@dozer2 dahdi]# modinfo -F version dahdi
> 2.6.1
> [root@dozer2 dahdi]# cat /sys/module/dahdi/version
> 2.6.1
> 
> On a whim, I went ahead and downloaded and built/installed dahdi 2.6.1
> After that, I ran dahdi_genconf and it made entries for the card in 
> /etc/dahdi/system.conf
>
> I think perhaps it didn't have the proper firmware downloaded in
> the distribution, or it tried to download it during the install
> and failed (we're behind a web content filtering system, so
> sometimes things get blocked)
> 
> But still when I run dahdi_hardware, I still only see the TDM800P card

Hmm... it looks like commit 10682 "dahdi_hardware: Detect the Digium
TE820 card" [1] is on the 2.6 branch in subversion but is not in the
2.6.1 release. If you need dahdi_hardware to work, you can install
DAHDI-Tools from trunk or you can wait until the next release of
DAHDI.

[1] http://svnview.digium.com/svn/dahdi?view=revision&revision=10682

Cheers,
Shaun

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Re: [asterisk-users] TE820 hardware detection

[Justin Killen]

I checked the version and it looks like it is up to date

[root@dozer2 dahdi]# asterisk -rx 'dahdi show version'
DAHDI Version: 2.6.1 Echo Canceller: HWEC
[root@dozer2 dahdi]# modinfo -F version dahdi
2.6.1
[root@dozer2 dahdi]# cat /sys/module/dahdi/version
2.6.1

On a whim, I went ahead and downloaded and built/installed dahdi 2.6.1
After that, I ran dahdi_genconf and it made entries for the card in 
/etc/dahdi/system.conf
I think perhaps it didn't have the proper firmware downloaded in the 
distribution, or it tried to download it during the install and failed (we're 
behind a web content filtering system, so sometimes things get blocked)

But still when I run dahdi_hardware, I still only see the TDM800P card

-Justin 
-Original Message-
From: Shaun Ruffell [mailto:sruff...@digium.com] 
Sent: Thursday, November 08, 2012 7:24 AM
To: Justin Killen
Subject: Re: [asterisk-users] TE820 hardware detection

On Thu, Nov 08, 2012 at 06:58:57AM -0800, Justin Killen wrote:
> Two follow up questions:
> 
> 1) is there an easy way to tell what dahdi version is installed?
> I tried looking in the man pages for something like a -version
> option, but I couldn't find one.

If you have Asterisk running you can get the version like:

  # asterisk -rx 'dahdi show version'
  DAHDI Version: v2.6.1 Echo Canceller: HWEC

You can see which version is installed in the modules (sitting on
the disk, not necessarily loaded into the running kernel) directory
like:

  # modinfo -F version dahdi
  v2.6.1

You can see which version is actually loaded in the kernel like:

  # cat /sys/module/dahdi/version 
  v2.6.1

> 2) is there an easy way to upgrade the dahdi package within the
> freepbx distro?  (i realize that this is more suited to the
> freepbx forums, but I'm hoping somebody here will know).

And I'm with you on this one...

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Re: [asterisk-users] Static on calls - v1.8.15.0

[Richard Mudgett]

> I'm testing out a server with asterisk 1.8.15.0 on it.
> I'm experiencing static occurring on almost 90% of calls on this
> particular server.
> All test phones are using SIP, and calls to/from PSTN servers are
> delivered using IAX2.
> 
> 
> I have other production servers running 1.4.x that do not have this
> issue that use the same PSTN connections.
> I haven't seen any ethernet errors or anything like that. Load is
> minimal since this is still a test server.
> The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s.
> 
> 
> I'm sort of baffled as to where to start looking for the root cause
> of this issue, but it appears to be isolated to only this machine.

You might have an A-law/u-law mismatch in the audio path.  That
kind of mismatch sounds like static on the line.

Richard

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[asterisk-users] How to propagate NOANSWER up through a Local channel?

[Tony Mountifield]

I am using a Local channel in order to provide a layer of processing
when dialling out. I have put a timeout on the inner dial, which returns
a DIALSTATUS of NOANSWER if the call times out while ringing.

However, this then causes the Local channel to return CHANUNAVAIL
instead of NOANSWER, because effectively the proxy channel has just gone
away.

Here is a simplified example:

[test]
exten => _X.,1,Dial(Local/${EXTEN}@outbound)
exten => _X.,n,NoOp(${CONTEXT}:DIALSTATUS=${DIALSTATUS})

[outbound]
exten => _X.,1,Dial(SIP/ext${EXTEN},30)
exten => _X.,n,NoOp(${CONTEXT}:DIALSTATUS=${DIALSTATUS})

So if I don't answer within 30 sec, I see outbound:DIALSTATUS=NOANSWER
but test:DIALSTATUS=CHANUNAVAIL

If instead I put the timeout on the outer Dial instead, I see
test:DIALSTATUS=NOANSWER and outbound:DIALSTATUS=CANCEL, because the
Local channel hung up the inner Dial while it was still ringing.

So I understand the reasons for the above behaviours, but my question
is: How can I propagate the NOANSWER status upwards from the inner Dial,
so that the Local channel also returns NOANSWER?

Cheers
Tony
-- 
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Play: t...@mountifield.org - http://tony.mountifield.org

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[asterisk-users] Static on calls - v1.8.15.0

[James Lamanna]

Hi,
I'm testing out a server with asterisk 1.8.15.0 on it.
I'm experiencing static occurring on almost 90% of calls on this particular
server.
All test phones are using SIP, and calls to/from PSTN servers are delivered
using IAX2.

I have other production servers running 1.4.x that do not have this issue
that use the same PSTN connections.
I haven't seen any ethernet errors or anything like that. Load is minimal
since this is still a test server.
The server itself has 16GB of RAM and Dual Quad Core Xeon E5345s.

I'm sort of baffled as to where to start looking for the root cause of this
issue, but it appears to be isolated to only this machine.

Thanks.
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Re: [asterisk-users] Managing complex setups with Asterisk

[Hans Witvliet]

On Thu, 2012-11-08 at 10:07 +0100, martin f krafft wrote:
> also sprach Jeff LaCoursiere  [2012.11.07.2049 +0100]:
> > Just to chime in, if you REALLY want multi-tenant, it is super
> > easy and surprisingly efficient to use kernel level virtualization
> > to run multiple instances of asterisk (and even FreePBX).  We use
> > LXC to do this.  The "host" runs an instance that has the dahdi
> > hardware, drivers, and upstream connections.  The "clients" have
> > SIP connections to the host for all inbound/outbound
> 
> Yes, separation into logical units is one way forward, but then you
> will necessarily have redundant configuration between the instances.
> It's nice to have clear separations (unless you cannot clearly
> separate), but I am not convinced that this decreases complexity.

Actually, i would suggest breaking it up and store most of your data
into mysql (realtime).

By breaking up, you can separate distinctive parts, like pstn-gateways,
GSM-gateways, internal-proxies, external-proxies, voice-mail,
conference-server, etc etc.
If you store the user-specific data into a database, it doesn't matter
on which proxy you register, the configuration is shared them all.
Same for your dialplan.

If you use LXC, the overhead will be less compared with using XEN. 
And if you keep each asterisk-container stupid, it is easier to
maintain/replace.

hw


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Re: [asterisk-users] Managing complex setups with Asterisk

[Shaun Ruffell]

On Thu, Nov 08, 2012 at 07:41:45AM +0100, martin f krafft wrote:
> 
> My systems are already managed automatically, thankfully no longer
> with Puppet. ;)

Just out of curiosity why do you say this?

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Re: [asterisk-users] Impromptu conferencing

[Aldo Bergamini]

On 8 Nov 2012, at 10:46, martin f krafft  wrote:

>> For a 3 way conference, all those days phones are able to do this.
> 
> Yeah, except I want Asterisk to handle that, not my phone (which
> might lose reception or run out of battery etc.).

Martin,

I understand that having the feature in the PBX makes it uniform to all phones, 
makes it available to all users, etc.

But it is not easy to do this inside the dialplan: the biggest problem is that 
you basically miss any kind of GUI to handle a set of steps that can each 
require a decision, to tell Asterisk what to do e.g. if the third person is 
busy and sends you to voicemail, having just a dialpad in your hands, with no 
visual feedback ...

Having complex functions in Asterisk is than linked with instruction sets for 
users (that should memorize them) in form of strange dialing sequences..

The best way to handle your idea is probably done through some kind of 
application to be used during calls..

Aldo
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Re: [asterisk-users] Impromptu conferencing

[martin f krafft]

also sprach Administrator TOOTAI  [2012.11.08.1018 +0100]:
> For a 3 way conference, all those days phones are able to do this.

Yeah, except I want Asterisk to handle that, not my phone (which
might lose reception or run out of battery etc.).

-- 
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Re: [asterisk-users] Managing complex setups with Asterisk

[Julian Lyndon-Smith]

A newbie question : Can each LXC "client" have their own ip address ?

Thanks

Julian

On 8 November 2012 07:17, Olivier  wrote:
>
>
> 2012/11/7 Jeff LaCoursiere 
>>>
>>>
>> Just to chime in, if you REALLY want multi-tenant, it is super easy and
>> surprisingly efficient to use kernel level virtualization to run multiple
>> instances of asterisk (and even FreePBX).  We use LXC to do this.  The
>> "host" runs an instance that has the dahdi hardware, drivers, and upstream
>> connections.  The "clients" have SIP connections to the host for all
>> inbound/outbound, so you have a central place to collect/process CDR records
>> for billing.  Getting your phones to connect to each instance is an exercise
>> for the network admin ;)
>>
>> Much simpler than working out multiple contexts, extension overlaps, etc.,
>> IMO.
>
>
> Yes, it's a very interesting idea !!
>
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Re: [asterisk-users] Impromptu conferencing

[Administrator TOOTAI]

Le 08/11/2012 10:01, martin f krafft a écrit :

also sprach Administrator TOOTAI  [2012.11.08.0954 +0100]:

Does anyone have a working example they would be willing to
share?

As said by James, you just have to transfer all parties in
a conference room and then you call this conference.

The scenario is usually that we are in a discussion and need a third
party. I suppose I can tell the initial correspondent "I will now
transfer you to a conference room, enter this PIN when asked", then
hang up, dial the next, and do the same.

What I would like to do is to convert the current channel into
a conference room, go on hold and dial a third party, and when
I come back to the conference room, I bring along the third party.

Put differently: I don't really want my correspondents to have to do
anything, just wait and listen to MOH.


For a 3 way conference, all those days phones are able to do this.

Anyway, the conference trick is as simply as the 3-way phone conference: 
no need of PIN conference, just a room with MOH when a user is sole. 
That's it, no manipulation for your correspondents.


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Re: [asterisk-users] Managing complex setups with Asterisk

[martin f krafft]

also sprach Jeff LaCoursiere  [2012.11.07.2049 +0100]:
> Just to chime in, if you REALLY want multi-tenant, it is super
> easy and surprisingly efficient to use kernel level virtualization
> to run multiple instances of asterisk (and even FreePBX).  We use
> LXC to do this.  The "host" runs an instance that has the dahdi
> hardware, drivers, and upstream connections.  The "clients" have
> SIP connections to the host for all inbound/outbound

Yes, separation into logical units is one way forward, but then you
will necessarily have redundant configuration between the instances.
It's nice to have clear separations (unless you cannot clearly
separate), but I am not convinced that this decreases complexity.

-- 
martin | http://madduck.net/ | http://two.sentenc.es/
 
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 takt heißt, sie nicht bemerken."
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Re: [asterisk-users] Impromptu conferencing

[martin f krafft]

also sprach Administrator TOOTAI  [2012.11.08.0954 +0100]:
> >Does anyone have a working example they would be willing to
> >share?
> 
> As said by James, you just have to transfer all parties in
> a conference room and then you call this conference.

The scenario is usually that we are in a discussion and need a third
party. I suppose I can tell the initial correspondent "I will now
transfer you to a conference room, enter this PIN when asked", then
hang up, dial the next, and do the same.

What I would like to do is to convert the current channel into
a conference room, go on hold and dial a third party, and when
I come back to the conference room, I bring along the third party.

Put differently: I don't really want my correspondents to have to do
anything, just wait and listen to MOH.

-- 
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 -- seneca
 
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Re: [asterisk-users] Impromptu conferencing

[Administrator TOOTAI]

Le 07/11/2012 20:01, martin f krafft a écrit :

Dear list,

we would really like to be able to "invite a third and fourth party"
to our current one-on-one call. At the moment, we have to agree to
dial into MeetMe 10 minutes later, then make calls to the third
parties, and hope it all works out.

I have found a couple of examples on the Internet for converting
channels into conferences, but I could not get any of them working.

Does anyone have a working example they would be willing to share?


As said by James, you just have to transfer all parties in a conference 
room and then you call this conference.


--
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Re: [asterisk-users] Can you help me to use SIPML5 with Asterisk ?

[qasimak...@gmail.com]

You can also hardcode these values in call.htm find below lines:

i_port = 4062 + (((new Date().getTime()) % 5) * 1000);^M
s_proxy = "sipml5.org";^M

and change them to

i_port = "/ws";^M
s_proxy = "ws://<* server IP>:";^M

Change  and <* server IP> with required values.

Regards,
Qasim


On Wed, Nov 7, 2012 at 7:52 PM, Joshua Colp  wrote:

> Lionel BEAUDOIN wrote:
>
>> Hello,
>>
>
> Hola,
>
>  I saw your email in a forum message, can you help me, I try to use
>> SIPML5 with an Asterisk 11 server ?
>>
>> My Asterisk server is installed on a Debian server.
>> I have download all the sources from sipml5.org
>>
>
> Please ensure you have followed the instructions at
> https://wiki.asterisk.org/**wiki/display/AST/Asterisk+**WebRTC+Supportto
>  set up the Asterisk side of things for WebSocket.
>
>  I have modifiied call.htm to target the requests on my server.
>>
>> - If I use the port 5060, I can register but I cant emet calls
>> - If I use the port 8088, I can't register.
>>
>> I think it's because I don't use the WS protocol but when I watch the
>> request on the 8088 port with tcpdump, I see that transport is UDP.
>>
>> How can I define a registring session with WS transport in the call.htm
>> file ?
>>
>
> You don't need to use your own copy of sipml5. Point a suitable browser to
> the following URL:
>
> http://sipml5.org/call.htm?**svn=9 
>
> Go into "Expert Mode" and disable Video support. Use the WebSocket Server
> URL for your server, like below:
>
> ws://:8088/ws
>
> Fill out the rest of the registration details as you normally would.
>
> Display Name: 
> Private Identity: 
> Public Identity: sip:@ Asterisk>
> Password: 
> Realm: 
>
> In the future please send emails of this type to the asterisk-users
> mailing list so that everyone can see the conversation and learn. I've
> copied my reply to it.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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Re: [asterisk-users] Managing complex setups with Asterisk

[Johan Wilfer]

2012-11-08 00:26, Jeff LaCoursiere skrev:

On 11/07/2012 05:20 PM, Jeff LaCoursiere wrote:

On 11/07/2012 02:16 PM, Johan Wilfer wrote:

2012-11-07 20:49, Jeff LaCoursiere skrev:

Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level virtualization to run
multiple instances of asterisk (and even FreePBX).  We use LXC to do
this.  The "host" runs an instance that has the dahdi hardware,
drivers,
and upstream connections.  The "clients" have SIP connections to the
host for all inbound/outbound, so you have a central place to
collect/process CDR records for billing.  Getting your phones to
connect
to each instance is an exercise for the network admin ;)

Any quirks / observations you have running LXC? We run OpenVZ now with
the same setup and it works very well. But as Debian will not support
OpenVZ in the next version we are looking for alternate solutions..

Do you run Dahdi run Dahdi for timing / meetme on both the "host" (HN)
and the "clients" (VE)?

Distribution?

Any other pitfalls or recommendations with LXC?




Since moving to Ubuntu 12.04 server, LXC mgmt has been much simpler
and stable.  Had some troubles with Ubuntu 10, though that was our
proof-of-concept, and mainly just with getting a template finalized.
Shutting down a container, back then, was a scary and often fatal
thing to do.  WIth 12.04 I have had zero LXC related issues in roughly
six months.  Have a few dozen companies running on the platform and
getting ready to white label the infrastructure to several resellers.
In our lab we have managed to get 200 instances, with FreePBX, running
simultaneously (though idle) on one host. Each (optimized!) container
seems to eat about 75M of RAM.  Our latest tweak is to make all of the
containers internally addressed on an OpenVPN-only accessible virtual
LAN, and are only distributing telephony hardware that can connect to
the platform natively (still on a search for the right ATA, though
getting by with DD-WRT router in front of Cisco ATA).

Cheers,

j




Realized I didn't answer your questions.  Yes, the host runs Dahdi for
timing, entirely for Meetme in the containers (host is just for
transport), and the driver is exposed to the containers in the LXC conf
file.

LXC does NOT have the same level of operational controls over the
containers, yet, as OpenVZ (like limiting resources).  That hasn't
really been an issue for us so far.

Cheers,

j


That's very cool.
I've also looked at Ubuntu server for their long support-time. I 
definitely have to test this setup.


Thanks for the insight!

--
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