Re: [asterisk-users] Asterisk server as TLS/SRTP

[Kseniya Blashchuk]

Hi!
I have used this document
https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport
You can specify transport=tls and encryption=yes for those peers which need
to use encryption.

пн, 5 мар. 2018 г. в 14:20, Antony Stone <
antony.st...@asterisk.open.source.it>:

> On Monday 05 March 2018 at 12:06:51, Atux Atux wrote:
>
> > Hi. I have an Asterisk Server (A) where it acts as the main gateway to
> > offer services.
> > There are different asterisk servers (B -D) that connect as extensions to
> > the Server A.
>
> Why not use IAX?
>
> > I would like to implement TLS and SRTP for these extensions, but have the
> > non secure as well for other extensions.
> > for example the extensions 4500-4504 be with TLS/SRTP and the rest be non
> > secure(ordinary).
> > Is there a guide on how to implement that please?
>
> How about
> https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
> or
> https://wiki.vpsget.com/index.php/Asterisk_11_with_TLS_and_SRTP_on_Centos_6
>
> > I am running asterisk 11.
>
> TLS has been available since 1.6 and SRTP since 1.8, so 11 should have no
> problems.
>
>
> Regards,
>
>
> Antony.
>
> --
> If the human brain were so simple that we could understand it,
> we'd be so simple that we couldn't.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asterisk server as TLS/SRTP

[Antony Stone]

On Monday 05 March 2018 at 12:06:51, Atux Atux wrote:

> Hi. I have an Asterisk Server (A) where it acts as the main gateway to
> offer services.
> There are different asterisk servers (B -D) that connect as extensions to
> the Server A.

Why not use IAX?

> I would like to implement TLS and SRTP for these extensions, but have the
> non secure as well for other extensions.
> for example the extensions 4500-4504 be with TLS/SRTP and the rest be non
> secure(ordinary).
> Is there a guide on how to implement that please?

How about https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial 
or https://wiki.vpsget.com/index.php/Asterisk_11_with_TLS_and_SRTP_on_Centos_6

> I am running asterisk 11.

TLS has been available since 1.6 and SRTP since 1.8, so 11 should have no 
problems.


Regards,


Antony.

-- 
If the human brain were so simple that we could understand it,
we'd be so simple that we couldn't.

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

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Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

And it is worst (and that could be the reason of the error).

127.0.0.1 is configured in 2 interfaces (lo and venet0), just with
different network masks.

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:54, andre castro  wrote:
> I am using version: 14.5.0
> No, Im not using Dundi.
> Can you a bit more informative when you say I "need to configure the IPs
> in your server"?
> thanks!
> a
> On 06/06/2017 07:47 PM, Marcelo Terres wrote:
>> I think you need to configure the IPs in your server. You just have 
>> localhost...
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 6 June 2017 at 16:27, andre castro  wrote:
>>> Thanks Anthony.
>>>
>>> I did it on the server, according to
>>> https://www.voip-info.org/wiki/view/port+forwarding
>>>
>>> However after doing it, when running Asterisk I get the following message
>>> sudo asterisk -vvr
>>> No ethernet interface found for seeding global EID. You will have to set
>>> it manually.
>>> Unable to access the running directory (No such file or directory).
>>> Changing to '/' for compatibility.
>>>
>>> How and where can it be set?
>>>
>>> My server ifconfig:
>>>
>>> loLink encap:Local Loopback
>>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>>   inet6 addr: ::1/128 Scope:Host
>>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>>
>>> venet0Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>>> Mask:255.255.255.255
>>>   inet6 addr: ::2/128 Scope:Compat
>>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>>
>>> venet0:0  Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>
>>>
>>>
>>> On 06/06/2017 05:09 PM, Antony Stone wrote:
 On Tuesday 06 June 2017 16:57:07 andre castro wrote:

> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>
>> Tell us about your networking arrangement - are both phones and the
>> Asterisk machine on the same network?
>
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk machine in another.

 Okay, that is why you have audio between the two phones, then - they can 
 see
 each other directly, on the same network, and nothing is interfering with 
 the
 traffic between them.

>> Is there a router in between any of them?
>
> Yes. In the phones network.
>
>> Is there any NAT involved?
>
> Yes in the phones' network. They both have different private IP address
> and one public IP.

 Okay, I suspect that this NATting router is not passing the UDP packets 
 from
 the server back to the phones correctly, based on the SIP connection (when 
 the
 phone makes the call).

 SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.

 If it's a Linux router, you need to make sure you are allowing FORWARDed 
 traffic
 which matches ESTABLISHED, RELATED.

 If it's not a Linux router, you need to find out how to get it to support 
 SIP
 and RTSP.


 Good luck,


 Antony.

>>>
>>> --
>>> oo.io
>>> bibliotecha.info
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at: 
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- 

Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

Well, based on the config that you sent, your server just have the
localhost IP (127.0.0.1)
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:54, andre castro  wrote:
> I am using version: 14.5.0
> No, Im not using Dundi.
> Can you a bit more informative when you say I "need to configure the IPs
> in your server"?
> thanks!
> a
> On 06/06/2017 07:47 PM, Marcelo Terres wrote:
>> I think you need to configure the IPs in your server. You just have 
>> localhost...
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 6 June 2017 at 16:27, andre castro  wrote:
>>> Thanks Anthony.
>>>
>>> I did it on the server, according to
>>> https://www.voip-info.org/wiki/view/port+forwarding
>>>
>>> However after doing it, when running Asterisk I get the following message
>>> sudo asterisk -vvr
>>> No ethernet interface found for seeding global EID. You will have to set
>>> it manually.
>>> Unable to access the running directory (No such file or directory).
>>> Changing to '/' for compatibility.
>>>
>>> How and where can it be set?
>>>
>>> My server ifconfig:
>>>
>>> loLink encap:Local Loopback
>>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>>   inet6 addr: ::1/128 Scope:Host
>>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>>
>>> venet0Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>>> Mask:255.255.255.255
>>>   inet6 addr: ::2/128 Scope:Compat
>>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>>
>>> venet0:0  Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>
>>>
>>>
>>> On 06/06/2017 05:09 PM, Antony Stone wrote:
 On Tuesday 06 June 2017 16:57:07 andre castro wrote:

> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>
>> Tell us about your networking arrangement - are both phones and the
>> Asterisk machine on the same network?
>
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk machine in another.

 Okay, that is why you have audio between the two phones, then - they can 
 see
 each other directly, on the same network, and nothing is interfering with 
 the
 traffic between them.

>> Is there a router in between any of them?
>
> Yes. In the phones network.
>
>> Is there any NAT involved?
>
> Yes in the phones' network. They both have different private IP address
> and one public IP.

 Okay, I suspect that this NATting router is not passing the UDP packets 
 from
 the server back to the phones correctly, based on the SIP connection (when 
 the
 phone makes the call).

 SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.

 If it's a Linux router, you need to make sure you are allowing FORWARDed 
 traffic
 which matches ESTABLISHED, RELATED.

 If it's not a Linux router, you need to find out how to get it to support 
 SIP
 and RTSP.


 Good luck,


 Antony.

>>>
>>> --
>>> oo.io
>>> bibliotecha.info
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at: 
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com 

Re: [asterisk-users] asterisk server - no sound

[andre castro]

I am using version: 14.5.0
No, Im not using Dundi.
Can you a bit more informative when you say I "need to configure the IPs
in your server"?
thanks!
a
On 06/06/2017 07:47 PM, Marcelo Terres wrote:
> I think you need to configure the IPs in your server. You just have 
> localhost...
> Marcelo H. Terres 
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
> 
> 
> On 6 June 2017 at 16:27, andre castro  wrote:
>> Thanks Anthony.
>>
>> I did it on the server, according to
>> https://www.voip-info.org/wiki/view/port+forwarding
>>
>> However after doing it, when running Asterisk I get the following message
>> sudo asterisk -vvr
>> No ethernet interface found for seeding global EID. You will have to set
>> it manually.
>> Unable to access the running directory (No such file or directory).
>> Changing to '/' for compatibility.
>>
>> How and where can it be set?
>>
>> My server ifconfig:
>>
>> loLink encap:Local Loopback
>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>   inet6 addr: ::1/128 Scope:Host
>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>   collisions:0 txqueuelen:0
>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>
>> venet0Link encap:UNSPEC  HWaddr
>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>> Mask:255.255.255.255
>>   inet6 addr: ::2/128 Scope:Compat
>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>   collisions:0 txqueuelen:0
>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>
>> venet0:0  Link encap:UNSPEC  HWaddr
>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>
>>
>>
>> On 06/06/2017 05:09 PM, Antony Stone wrote:
>>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>>
 On 06/06/2017 04:36 PM, Antony Stone wrote:
>
> Tell us about your networking arrangement - are both phones and the
> Asterisk machine on the same network?

 Nop. They are in 2 different networks. The phones in one and the
 Asterisk machine in another.
>>>
>>> Okay, that is why you have audio between the two phones, then - they can see
>>> each other directly, on the same network, and nothing is interfering with 
>>> the
>>> traffic between them.
>>>
> Is there a router in between any of them?

 Yes. In the phones network.

> Is there any NAT involved?

 Yes in the phones' network. They both have different private IP address
 and one public IP.
>>>
>>> Okay, I suspect that this NATting router is not passing the UDP packets from
>>> the server back to the phones correctly, based on the SIP connection (when 
>>> the
>>> phone makes the call).
>>>
>>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>>
>>> If it's a Linux router, you need to make sure you are allowing FORWARDed 
>>> traffic
>>> which matches ESTABLISHED, RELATED.
>>>
>>> If it's not a Linux router, you need to find out how to get it to support 
>>> SIP
>>> and RTSP.
>>>
>>>
>>> Good luck,
>>>
>>>
>>> Antony.
>>>
>>
>> --
>> oo.io
>> bibliotecha.info
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at: 
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
oo.io
bibliotecha.info

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
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Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

I think you need to configure the IPs in your server. You just have localhost...
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 16:27, andre castro  wrote:
> Thanks Anthony.
>
> I did it on the server, according to
> https://www.voip-info.org/wiki/view/port+forwarding
>
> However after doing it, when running Asterisk I get the following message
> sudo asterisk -vvr
> No ethernet interface found for seeding global EID. You will have to set
> it manually.
> Unable to access the running directory (No such file or directory).
> Changing to '/' for compatibility.
>
> How and where can it be set?
>
> My server ifconfig:
>
> loLink encap:Local Loopback
>   inet addr:127.0.0.1  Mask:255.0.0.0
>   inet6 addr: ::1/128 Scope:Host
>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>   collisions:0 txqueuelen:0
>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>
> venet0Link encap:UNSPEC  HWaddr
> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
> Mask:255.255.255.255
>   inet6 addr: ::2/128 Scope:Compat
>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>   collisions:0 txqueuelen:0
>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>
> venet0:0  Link encap:UNSPEC  HWaddr
> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
> Bcast:server.ip.add.r  Mask:255.255.255.255
>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>
>
>
> On 06/06/2017 05:09 PM, Antony Stone wrote:
>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>
>>> On 06/06/2017 04:36 PM, Antony Stone wrote:

 Tell us about your networking arrangement - are both phones and the
 Asterisk machine on the same network?
>>>
>>> Nop. They are in 2 different networks. The phones in one and the
>>> Asterisk machine in another.
>>
>> Okay, that is why you have audio between the two phones, then - they can see
>> each other directly, on the same network, and nothing is interfering with the
>> traffic between them.
>>
 Is there a router in between any of them?
>>>
>>> Yes. In the phones network.
>>>
 Is there any NAT involved?
>>>
>>> Yes in the phones' network. They both have different private IP address
>>> and one public IP.
>>
>> Okay, I suspect that this NATting router is not passing the UDP packets from
>> the server back to the phones correctly, based on the SIP connection (when 
>> the
>> phone makes the call).
>>
>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>
>> If it's a Linux router, you need to make sure you are allowing FORWARDed 
>> traffic
>> which matches ESTABLISHED, RELATED.
>>
>> If it's not a Linux router, you need to find out how to get it to support SIP
>> and RTSP.
>>
>>
>> Good luck,
>>
>>
>> Antony.
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

Looks like it comes com pbx_dundi.c.

Why are you using dundi?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:43, Marcelo Terres  wrote:
> Which Asterisk version are you using?
>
> Marcelo H. Terres 
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 6 June 2017 at 18:32, andre castro  wrote:
>> Any ideas.
>> After configuring  port forwarding on the server (machine making nat) to
>> forward connections originated from external clients to the machine
>> running asterisk, as explained in
>> https://www.voip-info.org/wiki/view/port+forwarding
>> My peers were unable to register.
>>
>>
>> And When running Asterisk I am getting:
>> No ethernet interface found for seeding global EID. You will have to set
>> it manually.
>> Unable to access the running directory (No such file or directory).
>> Changing to '/' for compatibility.
>>
>> Any advice what to do next?
>>
>> thanks
>> a
>>
>> On 06/06/2017 05:27 PM, andre castro wrote:
>>> Thanks Anthony.
>>>
>>> I did it on the server, according to
>>> https://www.voip-info.org/wiki/view/port+forwarding
>>>
>>> However after doing it, when running Asterisk I get the following message
>>> sudo asterisk -vvr
>>> No ethernet interface found for seeding global EID. You will have to set
>>> it manually.
>>> Unable to access the running directory (No such file or directory).
>>> Changing to '/' for compatibility.
>>>
>>> How and where can it be set?
>>>
>>> My server ifconfig:
>>>
>>> loLink encap:Local Loopback
>>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>>   inet6 addr: ::1/128 Scope:Host
>>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>>
>>> venet0Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>>> Mask:255.255.255.255
>>>   inet6 addr: ::2/128 Scope:Compat
>>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>>
>>> venet0:0  Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>
>>>
>>>
>>> On 06/06/2017 05:09 PM, Antony Stone wrote:
 On Tuesday 06 June 2017 16:57:07 andre castro wrote:

> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>
>> Tell us about your networking arrangement - are both phones and the
>> Asterisk machine on the same network?
>
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk machine in another.

 Okay, that is why you have audio between the two phones, then - they can 
 see
 each other directly, on the same network, and nothing is interfering with 
 the
 traffic between them.

>> Is there a router in between any of them?
>
> Yes. In the phones network.
>
>> Is there any NAT involved?
>
> Yes in the phones' network. They both have different private IP address
> and one public IP.

 Okay, I suspect that this NATting router is not passing the UDP packets 
 from
 the server back to the phones correctly, based on the SIP connection (when 
 the
 phone makes the call).

 SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.

 If it's a Linux router, you need to make sure you are allowing FORWARDed 
 traffic
 which matches ESTABLISHED, RELATED.

 If it's not a Linux router, you need to find out how to get it to support 
 SIP
 and RTSP.


 Good luck,


 Antony.

>>>
>>
>> --
>> oo.io
>> bibliotecha.info
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at: 
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   

Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

Which Asterisk version are you using?

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 18:32, andre castro  wrote:
> Any ideas.
> After configuring  port forwarding on the server (machine making nat) to
> forward connections originated from external clients to the machine
> running asterisk, as explained in
> https://www.voip-info.org/wiki/view/port+forwarding
> My peers were unable to register.
>
>
> And When running Asterisk I am getting:
> No ethernet interface found for seeding global EID. You will have to set
> it manually.
> Unable to access the running directory (No such file or directory).
> Changing to '/' for compatibility.
>
> Any advice what to do next?
>
> thanks
> a
>
> On 06/06/2017 05:27 PM, andre castro wrote:
>> Thanks Anthony.
>>
>> I did it on the server, according to
>> https://www.voip-info.org/wiki/view/port+forwarding
>>
>> However after doing it, when running Asterisk I get the following message
>> sudo asterisk -vvr
>> No ethernet interface found for seeding global EID. You will have to set
>> it manually.
>> Unable to access the running directory (No such file or directory).
>> Changing to '/' for compatibility.
>>
>> How and where can it be set?
>>
>> My server ifconfig:
>>
>> loLink encap:Local Loopback
>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>   inet6 addr: ::1/128 Scope:Host
>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>   collisions:0 txqueuelen:0
>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>
>> venet0Link encap:UNSPEC  HWaddr
>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>> Mask:255.255.255.255
>>   inet6 addr: ::2/128 Scope:Compat
>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>   collisions:0 txqueuelen:0
>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>
>> venet0:0  Link encap:UNSPEC  HWaddr
>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>
>>
>>
>> On 06/06/2017 05:09 PM, Antony Stone wrote:
>>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>>
 On 06/06/2017 04:36 PM, Antony Stone wrote:
>
> Tell us about your networking arrangement - are both phones and the
> Asterisk machine on the same network?

 Nop. They are in 2 different networks. The phones in one and the
 Asterisk machine in another.
>>>
>>> Okay, that is why you have audio between the two phones, then - they can see
>>> each other directly, on the same network, and nothing is interfering with 
>>> the
>>> traffic between them.
>>>
> Is there a router in between any of them?

 Yes. In the phones network.

> Is there any NAT involved?

 Yes in the phones' network. They both have different private IP address
 and one public IP.
>>>
>>> Okay, I suspect that this NATting router is not passing the UDP packets from
>>> the server back to the phones correctly, based on the SIP connection (when 
>>> the
>>> phone makes the call).
>>>
>>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>>
>>> If it's a Linux router, you need to make sure you are allowing FORWARDed 
>>> traffic
>>> which matches ESTABLISHED, RELATED.
>>>
>>> If it's not a Linux router, you need to find out how to get it to support 
>>> SIP
>>> and RTSP.
>>>
>>>
>>> Good luck,
>>>
>>>
>>> Antony.
>>>
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

asterisk-users 

Re: [asterisk-users] asterisk server - no sound

[andre castro]

Any ideas.
After configuring  port forwarding on the server (machine making nat) to
forward connections originated from external clients to the machine
running asterisk, as explained in
https://www.voip-info.org/wiki/view/port+forwarding
My peers were unable to register.


And When running Asterisk I am getting:
No ethernet interface found for seeding global EID. You will have to set
it manually.
Unable to access the running directory (No such file or directory).
Changing to '/' for compatibility.

Any advice what to do next?

thanks
a

On 06/06/2017 05:27 PM, andre castro wrote:
> Thanks Anthony.
> 
> I did it on the server, according to
> https://www.voip-info.org/wiki/view/port+forwarding
> 
> However after doing it, when running Asterisk I get the following message
> sudo asterisk -vvr
> No ethernet interface found for seeding global EID. You will have to set
> it manually.
> Unable to access the running directory (No such file or directory).
> Changing to '/' for compatibility.
> 
> How and where can it be set?
> 
> My server ifconfig:
> 
> loLink encap:Local Loopback
>   inet addr:127.0.0.1  Mask:255.0.0.0
>   inet6 addr: ::1/128 Scope:Host
>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>   collisions:0 txqueuelen:0
>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
> 
> venet0Link encap:UNSPEC  HWaddr
> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
> Mask:255.255.255.255
>   inet6 addr: ::2/128 Scope:Compat
>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>   collisions:0 txqueuelen:0
>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
> 
> venet0:0  Link encap:UNSPEC  HWaddr
> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
> Bcast:server.ip.add.r  Mask:255.255.255.255
>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
> 
> 
> 
> On 06/06/2017 05:09 PM, Antony Stone wrote:
>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>
>>> On 06/06/2017 04:36 PM, Antony Stone wrote:

 Tell us about your networking arrangement - are both phones and the
 Asterisk machine on the same network?
>>>
>>> Nop. They are in 2 different networks. The phones in one and the
>>> Asterisk machine in another.
>>
>> Okay, that is why you have audio between the two phones, then - they can see 
>> each other directly, on the same network, and nothing is interfering with 
>> the 
>> traffic between them.
>>
 Is there a router in between any of them?
>>>
>>> Yes. In the phones network.
>>>
 Is there any NAT involved?
>>>
>>> Yes in the phones' network. They both have different private IP address
>>> and one public IP.
>>
>> Okay, I suspect that this NATting router is not passing the UDP packets from 
>> the server back to the phones correctly, based on the SIP connection (when 
>> the 
>> phone makes the call).
>>
>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>
>> If it's a Linux router, you need to make sure you are allowing FORWARDed 
>> traffic 
>> which matches ESTABLISHED, RELATED.
>>
>> If it's not a Linux router, you need to find out how to get it to support 
>> SIP 
>> and RTSP.
>>
>>
>> Good luck,
>>
>>
>> Antony.
>>
> 

-- 
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Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

Try to use the echo app. If you can listen your echo, so it is
something in the network.

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 14:18, andre castro  wrote:
> hello folks,
> this might be a simple question...
>
> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.
> If I have one of my registered peers call and extension (102) that plays
> back audio (extension.conf and sip.conf coffee-pasted below), Asterisk
> answers and prints no errors.
> Its `sip show channels` prints:
>
> PeerUser/ANRCall IDFormatHoldLast MessageExpiry
>Peer
> peer.ip1001 1...-5060   (ulaw)  No Rx: ACK
>1001
>
> But I hear nothing at the peer's end.
>
> When one peer calls another, sound comes through just fine.
> So my hunch is that is something to do with the audio supplied by the
> server.
> Do I need to have alsa installed??
> Any hint?
>
> sip.conf:
>
> [general]
> context = unauthenticated
> bindport = 5060
> bindaddr = 0.0.0.0
> tcpbindaddr = 0.0.0.0
> tcpenable = yes
> videosupport = no
> textsupport=yes
> alwaysauthreject=yes
> allowguest=no
>
> [1001] ; grandstream 1
> context = home
> type = friend
> callerid = One <1001>
> secret = XYZ
> host = dynamic
> mailbox = 1001
> disallow = all
> allow = ulaw
> transport = udp
> dtmfmode=auto   ; accept touch-tones from the devices, negotiated
> automatically
> nat=force_rport
>
> [1005] ; mobile
> context = home
> type = friend
> callerid = Five <1005>
> secret = XYZ
> host = dynamic
> mailbox = 1005
> disallow = all
> allow = ulaw
> transport = udp
> dtmfmode=auto   ; accept touch-tones from the devices, negotiated
> automatically
> nat=force_rport
>
>
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same =  n,Wait(1)
> same =  n,Playback(beep)
> same =  n,Wait(1)
> same =  n,Playback(im-sorry)
> same =  n,Wait(1)
> same =  n,Playback(number-not-answering)
> same =  n,Wait(1)
> same =  n,Hangup()
>
> exten => 1001,1,Dial(SIP/1001) ; grandstream phone
> exten => 1005,1,Dial(SIP/1005) ; mobile
>
>
>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] asterisk server - no sound

[andre castro]

Thanks Anthony.

I did it on the server, according to
https://www.voip-info.org/wiki/view/port+forwarding

However after doing it, when running Asterisk I get the following message
sudo asterisk -vvr
No ethernet interface found for seeding global EID. You will have to set
it manually.
Unable to access the running directory (No such file or directory).
Changing to '/' for compatibility.

How and where can it be set?

My server ifconfig:

loLink encap:Local Loopback
  inet addr:127.0.0.1  Mask:255.0.0.0
  inet6 addr: ::1/128 Scope:Host
  UP LOOPBACK RUNNING  MTU:65536  Metric:1
  RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
  TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
  collisions:0 txqueuelen:0
  RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)

venet0Link encap:UNSPEC  HWaddr
00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
  inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
Mask:255.255.255.255
  inet6 addr: ::2/128 Scope:Compat
  inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
  UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
  RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
  TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
  collisions:0 txqueuelen:0
  RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)

venet0:0  Link encap:UNSPEC  HWaddr
00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
  inet addr:server.ip.add.r  P-t-P:server.ip.add.r
Bcast:server.ip.add.r  Mask:255.255.255.255
  UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1



On 06/06/2017 05:09 PM, Antony Stone wrote:
> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
> 
>> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>>
>>> Tell us about your networking arrangement - are both phones and the
>>> Asterisk machine on the same network?
>>
>> Nop. They are in 2 different networks. The phones in one and the
>> Asterisk machine in another.
> 
> Okay, that is why you have audio between the two phones, then - they can see 
> each other directly, on the same network, and nothing is interfering with the 
> traffic between them.
> 
>>> Is there a router in between any of them?
>>
>> Yes. In the phones network.
>>
>>> Is there any NAT involved?
>>
>> Yes in the phones' network. They both have different private IP address
>> and one public IP.
> 
> Okay, I suspect that this NATting router is not passing the UDP packets from 
> the server back to the phones correctly, based on the SIP connection (when 
> the 
> phone makes the call).
> 
> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
> 
> If it's a Linux router, you need to make sure you are allowing FORWARDed 
> traffic 
> which matches ESTABLISHED, RELATED.
> 
> If it's not a Linux router, you need to find out how to get it to support SIP 
> and RTSP.
> 
> 
> Good luck,
> 
> 
> Antony.
> 

-- 
oo.io
bibliotecha.info

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Re: [asterisk-users] asterisk server - no sound

[Antony Stone]

On Tuesday 06 June 2017 16:57:07 andre castro wrote:

> On 06/06/2017 04:36 PM, Antony Stone wrote:
> > 
> > Tell us about your networking arrangement - are both phones and the
> > Asterisk machine on the same network?
> 
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk machine in another.

Okay, that is why you have audio between the two phones, then - they can see 
each other directly, on the same network, and nothing is interfering with the 
traffic between them.

> > Is there a router in between any of them?
> 
> Yes. In the phones network.
> 
> > Is there any NAT involved?
> 
> Yes in the phones' network. They both have different private IP address
> and one public IP.

Okay, I suspect that this NATting router is not passing the UDP packets from 
the server back to the phones correctly, based on the SIP connection (when the 
phone makes the call).

SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.

If it's a Linux router, you need to make sure you are allowing FORWARDed 
traffic 
which matches ESTABLISHED, RELATED.

If it's not a Linux router, you need to find out how to get it to support SIP 
and RTSP.


Good luck,


Antony.

-- 
There's a good theatrical performance about puns on in the West End.  It's a 
play on words.

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


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Re: [asterisk-users] asterisk server - no sound

[andre castro]

Thank you Daniel for pointing out the errors and debug option. Both
fixed and on.
It made no difference. There are no errors printed and still no sound on
ppers

Now to Antony questions:

On 06/06/2017 04:36 PM, Antony Stone wrote:
> On Tuesday 06 June 2017 15:18:32 andre castro wrote:
> 
>> I just installed asterisk in a debian server.
>> All seems to be running fine, but the audio sent by the server.
> 
>> But I hear nothing at the peer's end.
>>
>> When one peer calls another, sound comes through just fine.
> 
> Tell us about your networking arrangement - are both phones and the Asterisk 
> machine on the same network?

Nop. They are in 2 different networks. The phones in one and the
Asterisk machine in another.
> 
> Is there a router in between any of them?
Yes. In the phones network.
> 
> Is there any NAT involved?
Yes in the phones' network. They both have different private IP address
and one public IP.
> 
>> Do I need to have alsa installed??
> 
> No.
So I thought.

Thanks guys!!
> 
> 
> Antony.
> 

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Re: [asterisk-users] asterisk server - no sound

[Administrator TOOTAI]

Le 06/06/2017 à 16:25, Daniel Tryba a écrit :

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:

extensions.conf:
[home]
exten = 102,1,Answer()
same =  n,Wait(1)


If this is copy and paste, then your dialplan is broken (= should be =>)


Well, no. = or => are the same.

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Re: [asterisk-users] asterisk server - no sound

[Antony Stone]

On Tuesday 06 June 2017 15:18:32 andre castro wrote:

> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.

> But I hear nothing at the peer's end.
> 
> When one peer calls another, sound comes through just fine.

Tell us about your networking arrangement - are both phones and the Asterisk 
machine on the same network?

Is there a router in between any of them?

Is there any NAT involved?

> Do I need to have alsa installed??

No.


Antony.

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rather when there is nothing left to take away.

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


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Re: [asterisk-users] asterisk server - no sound

[Daniel Tryba]

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same =  n,Wait(1)

If this is copy and paste, then your dialplan is broken (= should be =>)

But to debug, enable logging (core set verbose 5), when needed debugging
(core set debug 5) and sip logging (sip set debug on / pjsip set logger
on).

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Re: [asterisk-users] asterisk server stress test

[Sevana Oy]

Hi,



Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score
the quality? Using voice files for tests has more representation to my
opinion.



Thanks,
vallu

On Thu, Aug 20, 2015 at 4:11 AM, Pete Mundy p...@fiberphone.co.nz wrote:

 Markus

 That's a fascinating concept!

 Can you share any more about how you appraised the data and determined
 your results?

 ie once you had the recordings on the second host what did you do do
 computationally score them? Do you look at the decoded (1khz?) waveform or
 do you appraise in another way?

 Pete

 On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org
 wrote:

 Am 19.08.2015 um 19:07 schrieb Steve Edwards:

 Please don't top post.

 On Wed, 19 Aug 2015, James Cass wrote:

 Steve, would you be willing to share that quick bash script?


 There's no magic in the script, but here it is, embarrassing myself:

cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10

 Here's what's wrong with this snippet:

 1) I don't know why I chmod the 'template.' No idea whatsoever. Alcohol
 may have been involved.

 2) I hate single character variable names. I love alcohol.

 3) cp is ill advised. For a testing script, it was easy. For a production
 application, use mv.

 In use, I would execute it specifying how many call files to create, like
 50. Then, take a look at top, iftop, and vmstat. Lather, rinse, repeat to
 get to your goal.


 We started the 500 calls and used milliwatt app on the first and record on
 the second host to check the quality. Alternatively just start 500+ calls
 and call yourself on top. So you can get a good idea how the quality is.

 Call-Files are explained on
 http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out

 Markus

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Re: [asterisk-users] asterisk server stress test

[Markus Weiler]

Am 20.08.2015 um 03:16 schrieb Pete Mundy:


Ah cr@p, sorry Steve, didn't mean to top-post there.


On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org 
mailto:markus_wei...@mailworks.org wrote:
We started the 500 calls and used milliwatt app on the first and 
record on the second host to check the quality. Alternatively just 
start 500+ calls and call yourself on top. So you can get a good 
idea how the quality is.


Markus

That's a fascinating concept!

Can you share any more about how you appraised the data and determined 
your results?


ie once you had the recordings on the second host what did you do do 
computationally score them? Do you look at the decoded (1khz?) 
waveform or do you appraise in another way?


Pete







Hi Pete,

we used different approaches.

Just to test the maximum channels a gateway can process the two Methods 
are enough, you can either listen to the Recordings or look at the waveform.
The easiest approach is to call a colleague and gradually increase the 
calls on the machine.


For systematic, continuous analysis Voipmonitor is a very useful tool.
We directed the traffic to a mirroring port on the Switch to which we 
connected a Server running Voipmon. (http://www.voipmonitor.org/)
Voipmon records the call and rates its quality. You can check the 
results either using the commercial Web Interface (test for free) or 
query the mysql DB.
Unfortunately Voipmon tends to crash on a regular basis (at least when 
we used it), but it's an awesome tool.
The underlying tool pcapsipdump is running a lot more stable, but you 
need to put a lot more work into it to get started.


hope i could help

Markus





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Re: [asterisk-users] asterisk server stress test

[Steven Howes]

On 20 Aug 2015, at 11:12, Sevana Oy sa...@sevana.fi wrote:
 Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score 
 the quality? Using voice files for tests has more representation to my 
 opinion.


Spot the salesman? ;)

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Re: [asterisk-users] asterisk server stress test

[Dominique Haeber]

Hi Barry Flanagan,

Dominique Haeber dominique.hae...@xig.ch schrieb am Mit, 19. Aug 15:13:
 Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06:
  SIPP is probably what you seek. http://sipp.sourceforge.net/
  
  Hope this helps.
 
 That looks pretty like what I'm looking for! Many thanks!

The control file needs some training but I was successful to the
goal, thanks again.

Sincerely,
Dominique Haeber

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Re: [asterisk-users] asterisk server stress test

[Dominique Haeber]

Hi Barry Flanagan,

Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06:
 SIPP is probably what you seek. http://sipp.sourceforge.net/
 
 Hope this helps.

That looks pretty like what I'm looking for! Many thanks!


Sincerely,
Dominique Haeber


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Re: [asterisk-users] asterisk server stress test

[Steve Edwards]

On Wed, 19 Aug 2015, Dominique Haeber wrote:


Hi Barry Flanagan,

Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06:

SIPP is probably what you seek. http://sipp.sourceforge.net/

Hope this helps.


That looks pretty like what I'm looking for! Many thanks!


Another approach is to use another Asterisk system.

Recently, a customer wanted to confirm his platform would support 500 
simultaneous calls.


I wrote a quick bash script to dump 500 call files (at a leisurely pace) 
into another host that originated calls to the target host.


--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST

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Re: [asterisk-users] asterisk server stress test

[Steve Edwards]

Please don't top post.

On Wed, 19 Aug 2015, James Cass wrote:


Steve, would you be willing to share that quick bash script?


There's no magic in the script, but here it is, embarrassing myself:

cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10

Here's what's wrong with this snippet:

1) I don't know why I chmod the 'template.' No idea whatsoever. Alcohol 
may have been involved.


2) I hate single character variable names. I love alcohol.

3) cp is ill advised. For a testing script, it was easy. For a production 
application, use mv.


In use, I would execute it specifying how many call files to create, like 
50. Then, take a look at top, iftop, and vmstat. Lather, rinse, repeat to 
get to your goal.


--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST

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Re: [asterisk-users] asterisk server stress test

[James Cass]

Steve, would you be willing to share that quick bash script?

James Cass http://goog_987864563
jcas...@gmail.com


On Wed, Aug 19, 2015 at 12:11 PM, Steve Edwards asterisk@sedwards.com
wrote:

 On Wed, 19 Aug 2015, Dominique Haeber wrote:

 Hi Barry Flanagan,

 Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06:

 SIPP is probably what you seek. http://sipp.sourceforge.net/

 Hope this helps.


 That looks pretty like what I'm looking for! Many thanks!


 Another approach is to use another Asterisk system.

 Recently, a customer wanted to confirm his platform would support 500
 simultaneous calls.

 I wrote a quick bash script to dump 500 call files (at a leisurely pace)
 into another host that originated calls to the target host.

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


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Re: [asterisk-users] asterisk server stress test

[Markus Weiler]

Am 19.08.2015 um 19:07 schrieb Steve Edwards:

Please don't top post.

On Wed, 19 Aug 2015, James Cass wrote:


Steve, would you be willing to share that quick bash script?


There's no magic in the script, but here it is, embarrassing myself:

cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10

Here's what's wrong with this snippet:

1) I don't know why I chmod the 'template.' No idea whatsoever. 
Alcohol may have been involved.


2) I hate single character variable names. I love alcohol.

3) cp is ill advised. For a testing script, it was easy. For a 
production application, use mv.


In use, I would execute it specifying how many call files to create, 
like 50. Then, take a look at top, iftop, and vmstat. Lather, rinse, 
repeat to get to your goal.




We started the 500 calls and used milliwatt app on the first and record 
on the second host to check the quality. Alternatively just start 500+ 
calls and call yourself on top. So you can get a good idea how the 
quality is.


Call-Files are explained on 
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out


Markus

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Re: [asterisk-users] asterisk server stress test

[Pete Mundy]

Ah cr@p, sorry Steve, didn't mean to top-post there.


 On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org wrote:
 We started the 500 calls and used milliwatt app on the first and record on 
 the second host to check the quality. Alternatively just start 500+ calls 
 and call yourself on top. So you can get a good idea how the quality is.

Markus

That's a fascinating concept!

Can you share any more about how you appraised the data and determined your 
results?

ie once you had the recordings on the second host what did you do do 
computationally score them? Do you look at the decoded (1khz?) waveform or do 
you appraise in another way?

Pete





smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] asterisk server stress test

[Pete Mundy]

Markus

That's a fascinating concept!

Can you share any more about how you appraised the data and determined your 
results?

ie once you had the recordings on the second host what did you do do 
computationally score them? Do you look at the decoded (1khz?) waveform or do 
you appraise in another way?

Pete


On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org wrote:

 Am 19.08.2015 um 19:07 schrieb Steve Edwards:
 Please don't top post.
 
 On Wed, 19 Aug 2015, James Cass wrote:
 
 Steve, would you be willing to share that quick bash script?
 
 There's no magic in the script, but here it is, embarrassing myself:
 
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10
 
 Here's what's wrong with this snippet:
 
 1) I don't know why I chmod the 'template.' No idea whatsoever. Alcohol may 
 have been involved.
 
 2) I hate single character variable names. I love alcohol.
 
 3) cp is ill advised. For a testing script, it was easy. For a production 
 application, use mv.
 
 In use, I would execute it specifying how many call files to create, like 
 50. Then, take a look at top, iftop, and vmstat. Lather, rinse, repeat to 
 get to your goal.
 
 
 We started the 500 calls and used milliwatt app on the first and record on 
 the second host to check the quality. Alternatively just start 500+ calls and 
 call yourself on top. So you can get a good idea how the quality is.
 
 Call-Files are explained on 
 http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
 
 Markus
 
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Re: [asterisk-users] asterisk server stress test

[Barry Flanagan]

On 19 August 2015 at 08:01, Dominique Haeber dominique.hae...@xig.ch
wrote:


 Hi all,

 i need to test how many calls can withstand an Asterisk server.

 Do you know any good tools to strain the server?

 At best, there are scripts that I can run on a Linux server.



SIPP is probably what you seek. http://sipp.sourceforge.net/

Hope this helps.

-Barry Flanagan
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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Leif Madsen]

On 12/09/11 02:21 PM, linux guy wrote:

I'm about to start building my asterisk server and I can't seem to find
anything that discusses the pros and cons of installing the OS (Fedora
15) as console only or GUI, ie install KDE as well.


http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id291070

--
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http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Robert Huddleston]

I personally would never install a GUI o/s. By doing so you always open
yourself up to more security concerns.. Packages / ports / etc.

 

Course one might argue - it's behind a firewall..

 

In my professional experience with running numerous ISP and VoITSPs the rule
has always been install the minimum needed software to accomplish the goal.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy
Sent: Monday, September 12, 2011 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3
or init level 5 ?

 

I'm about to start building my asterisk server and I can't seem to find
anything that discusses the pros and cons of installing the OS (Fedora 15)
as console only or GUI, ie install KDE as well.

So, other than a bit of disk space, is there any reason why I shouldn't
install KDE when I set it up ?

Is there any great disadvantage to running the server in init level 5 (ie
KDE, xorg, etc) running in the background, but not being logged in, versus
init level 3 ? (Or whatever they call these things these days..., ie F15
uses systemd...)

FWIW, my server hardware will sit on a server rack in the utility room.  I
might drag a display and keyboard down there once in a while to troubleshoot
and/or do maintenance, but mostly I'd ssh in and probably use a remote
desktop app to work on it.

FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have
graphical tools.

I look forward to your input.

Thanks

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Doug Lytle]

linux guy wrote:
So, other than a bit of disk space, is there any reason why I 
shouldn't install KDE when I set it up ?



KDE has other associated background services that may slow a machine 
down.  If you're looking for a DE, I'd go with something light weight.  
LXDE is my preferred choice.


But mostly I run the graphical tools for Mandriva/Mageia over SSH.

Doug


--

Ben Franklin quote:

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


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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[linux guy]

On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston
rhuddles...@gmail.comwrote:

 I personally would never install a GUI o/s… By doing so you always open
 yourself up to more security concerns.. Packages / ports / etc.

 ** **

 Course one might argue – “it’s behind a firewall”….

 ** **

 In my professional experience with running numerous ISP and VoITSPs the
 rule has always been install the minimum needed software to accomplish the
 goal.


Thanks for the reply.  I was worried the list would find it a trite and
irritating question.

I was expecting someone to tell me that even with the GUI component running
in the background, the graphical processes have the potential to mess up the
streams.  I guess I should confess that I'm always a bit surprised to
remember that asterisk doesn't require a real time OS !

Have you really exposed much more if you install the GUI components and
normally run at init 3 ?

Thanks again.
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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Robert Huddleston]

Well you are correct - I did not include a discussion on performance impacts
including disk I/O etc.

 

It is true that by installing a GUI o/s additional init.d (startup) services
will fire.. Additional libraries will be inclusive etc.

 

This is why I say minimal is always better.

 

Also take for example risk mitigation with security aspects. If you minimize
the number of libraries (think windows DLL's) you have installed you also
thus minimize your potential exposure.

 

Again - this is just my recommendation and experience. Firewalls are great
at blocking things and in theory - sure you could nmap your box and look for
open ports and conceal them.

 

I remember a Solaris engineer we had once - he bragged and bragged about his
qualifications on Sun Solaris. Just to find out that he installed a bunch of
GUI tools just so that he could install Oracle drivers. Further he didn't
remove or lock down that exposure.

 

Start minimal and work your way up. Now for my poke / razz - GUI's in server
grade operating systems have made people a little to reliant on them.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy
Sent: Monday, September 12, 2011 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init
level 3 or init level 5 ?

 

 

On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston rhuddles...@gmail.com
wrote:

I personally would never install a GUI o/s. By doing so you always open
yourself up to more security concerns.. Packages / ports / etc.

 

Course one might argue - it's behind a firewall..

 

In my professional experience with running numerous ISP and VoITSPs the rule
has always been install the minimum needed software to accomplish the goal.


Thanks for the reply.  I was worried the list would find it a trite and
irritating question.

I was expecting someone to tell me that even with the GUI component running
in the background, the graphical processes have the potential to mess up the
streams.  I guess I should confess that I'm always a bit surprised to
remember that asterisk doesn't require a real time OS !

Have you really exposed much more if you install the GUI components and
normally run at init 3 ?

Thanks again. 

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Steve Edwards]

On Mon, 12 Sep 2011, linux guy wrote:

I'm about to start building my asterisk server and I can't seem to find 
anything that discusses the pros and cons of installing the OS (Fedora 
15) as console only or GUI, ie install KDE as well.


Parts left out don't get broke.

Install the absolute minimum OS (deselect everything) and 'yum in' the 
packages you actually need.


When you configure Asterisk, set 'autoload = no' and explicitly load the 
modules you actually use.


Is there any great disadvantage to running the server in init level 5 
(ie KDE, xorg, etc) running in the background, but not being logged in, 
versus init level 3 ? (Or whatever they call these things these days..., 
ie F15 uses systemd...)


The 'run level' you configure Asterisk to start at is not dependent on the 
interface. You can chkconfig Asterisk to run at levels 2345 regardless of 
the interface installed.


FWIW, I'm OK doing things via the CLI, but sometimes its really nice to 
have graphical tools.


None of the servers I manage have a GUI installed. All are administrated 
over ssh. The only situation where having a GUI installed would be 
convenient would be if I were local to the console and wanted to run 
wireshark.


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

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Roger Burton West]

On Mon, Sep 12, 2011 at 12:21:06PM -0600, linux guy wrote:

FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have
graphical tools.

To add to what everyone else has said: if you _really_ need to run a
graphical tool on the server, you can always ssh -X into it without
having to have a full desktop installed there.

(As for wireshark: tcpdump on site, then bring the capture file home to
analyse with wireshark. Works for me...)

Roger

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Steve Totaro]

See comments inline.

On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote:

 I'm about to start building my asterisk server and I can't seem to find
 anything that discusses the pros and cons of installing the OS (Fedora 15)
 as console only or GUI, ie install KDE as well.


If you want an OS that is going to be supported a year from now, don't use
Fedora.

Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much
beta RHEL.  It's EOL is one year from my understanding.

You want to install the very minimum as most people would agree, why do you
think you need a GUI.

Best practice is to only install the bare minimum on a server.


 So, other than a bit of disk space, is there any reason why I shouldn't
 install KDE when I set it up ?


It has and will cause issues.  I have installed KDE or whatever but booted
to init 3 for a couple of machines.  I could go to init 5 if I had to, but I
never did had to.  I don't see a single pro, but there are many cons.

What benefit do you get from KDE?  Why do you want it.  Is this just going
to be an asterisk server or a desktop?



 Is there any great disadvantage to running the server in init level 5 (ie
 KDE, xorg, etc) running in the background, but not being logged in, versus
 init level 3 ? (Or whatever they call these things these days..., ie F15
 uses systemd...)

 FWIW, my server hardware will sit on a server rack in the utility room.  I
 might drag a display and keyboard down there once in a while to troubleshoot
 and/or do maintenance, but mostly I'd ssh in and probably use a remote
 desktop app to work on it.


How does remote desktop help you over an SSH CLI?


 FWIW, I'm OK doing things via the CLI, but sometimes its really nice to
 have graphical tools.


Ok, I can understand, I used to be like this for a while.  I am a huge fan
of Webmin for a GUI.  It allows for almost everything and for me, it is
better than KDE or anything else.  It is just a webpage with tools
attached.  No big potential problem there.


 I look forward to your input.

 Thanks

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Steve Totaro]

On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:

 See comments inline.

 On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote:

 I'm about to start building my asterisk server and I can't seem to find
 anything that discusses the pros and cons of installing the OS (Fedora 15)
 as console only or GUI, ie install KDE as well.


 If you want an OS that is going to be supported a year from now, don't use
 Fedora.

 Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty
 much beta RHEL.  It's EOL is one year from my understanding.

 You want to install the very minimum as most people would agree, why do you
 think you need a GUI.

 Best practice is to only install the bare minimum on a server.


 So, other than a bit of disk space, is there any reason why I shouldn't
 install KDE when I set it up ?


 It has and will cause issues.  I have installed KDE or whatever but booted
 to init 3 for a couple of machines.  I could go to init 5 if I had to, but I
 never did had to.  I don't see a single pro, but there are many cons.

 What benefit do you get from KDE?  Why do you want it.  Is this just going
 to be an asterisk server or a desktop?



 Is there any great disadvantage to running the server in init level 5 (ie
 KDE, xorg, etc) running in the background, but not being logged in, versus
 init level 3 ? (Or whatever they call these things these days..., ie F15
 uses systemd...)

 FWIW, my server hardware will sit on a server rack in the utility room.  I
 might drag a display and keyboard down there once in a while to troubleshoot
 and/or do maintenance, but mostly I'd ssh in and probably use a remote
 desktop app to work on it.


 How does remote desktop help you over an SSH CLI?


 FWIW, I'm OK doing things via the CLI, but sometimes its really nice to
 have graphical tools.


 Ok, I can understand, I used to be like this for a while.  I am a huge fan
 of Webmin for a GUI.  It allows for almost everything and for me, it is
 better than KDE or anything else.  It is just a webpage with tools
 attached.  No big potential problem there.


 I look forward to your input.

 Thanks


I have been using Vyatta (paid for with phone support.)

It makes for the most powerful Asterisk platform you can imagine.  There is
a learning curve but I love what I have put together.  There are howtos
everywhere and if you buy licenses, you get excellent support and online
training courses.

It is a very firewall/Router.  It handles everything from OpenVPN, awesome
security features, IPS, and even QoS, wireshark.

I put webmin and NTOP on these machines as well.  Vyatta has become my new
platform for Asterisk.

Check it out http://www.vyatta.org/documentation

There is very little you cannot do, but don't have to use the features if
you don't want to.

Vyatta is also a company like Asterisk.  Vyatta is the baby of former
bigtime corporate Cisco guys.  Asterisk is the baby of former Adtran execs.

Thanks,
Steve T

Thanks,
Steve T
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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Robert-iPhone]

Asterisk is a company? This is news to me

Sent from my iPhone

On Sep 12, 2011, at 5:35 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote:

 
 
 On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.com 
 wrote:
 See comments inline.
 
 On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote:
 I'm about to start building my asterisk server and I can't seem to find 
 anything that discusses the pros and cons of installing the OS (Fedora 15) as 
 console only or GUI, ie install KDE as well.
 
 
 If you want an OS that is going to be supported a year from now, don't use 
 Fedora.
 
 Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much 
 beta RHEL.  It's EOL is one year from my understanding.
 
 You want to install the very minimum as most people would agree, why do you 
 think you need a GUI.
 
 Best practice is to only install the bare minimum on a server.
  
 So, other than a bit of disk space, is there any reason why I shouldn't 
 install KDE when I set it up ?
 
 It has and will cause issues.  I have installed KDE or whatever but booted to 
 init 3 for a couple of machines.  I could go to init 5 if I had to, but I 
 never did had to.  I don't see a single pro, but there are many cons.
 
 What benefit do you get from KDE?  Why do you want it.  Is this just going to 
 be an asterisk server or a desktop?
  
 
 Is there any great disadvantage to running the server in init level 5 (ie 
 KDE, xorg, etc) running in the background, but not being logged in, versus 
 init level 3 ? (Or whatever they call these things these days..., ie F15 uses 
 systemd...)
 
 FWIW, my server hardware will sit on a server rack in the utility room.  I 
 might drag a display and keyboard down there once in a while to troubleshoot 
 and/or do maintenance, but mostly I'd ssh in and probably use a remote 
 desktop app to work on it.   
 
 How does remote desktop help you over an SSH CLI?
  
 FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have 
 graphical tools.
 
 
 Ok, I can understand, I used to be like this for a while.  I am a huge fan of 
 Webmin for a GUI.  It allows for almost everything and for me, it is better 
 than KDE or anything else.  It is just a webpage with tools attached.  No big 
 potential problem there.
  
 I look forward to your input.
 
 Thanks
 
 
 I have been using Vyatta (paid for with phone support.)
 
 It makes for the most powerful Asterisk platform you can imagine.  There is a 
 learning curve but I love what I have put together.  There are howtos 
 everywhere and if you buy licenses, you get excellent support and online 
 training courses.
 
 It is a very firewall/Router.  It handles everything from OpenVPN, awesome 
 security features, IPS, and even QoS, wireshark.
 
 I put webmin and NTOP on these machines as well.  Vyatta has become my new 
 platform for Asterisk.
 
 Check it out http://www.vyatta.org/documentation
 
 There is very little you cannot do, but don't have to use the features if you 
 don't want to.
 
 Vyatta is also a company like Asterisk.  Vyatta is the baby of former bigtime 
 corporate Cisco guys.  Asterisk is the baby of former Adtran execs.
 
 Thanks,
 Steve T
 
 Thanks,
 Steve T
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Re: [asterisk-users] Asterisk Server with Panasonic PBX

[ABBAS SHAKEEL]

Thanks CF.

The requirements are

Panasonic TDA will have all the PSTN lines from Telco company. Asterisk Box
will get phone lines from TDA.
Now it works fine when take an Extension from TDA and Put it in Asterisk BOX
(TDM400P).

Asterisk Box recieves call exactly ok but when asterisk box need to dail out
of office as usual 9 is dailed first then have a wait for line then TDA  PBX
gives line and we can dial any number.

Now this problem is solved by introducing wait in dailing a number. ie dial
(9wwthePhoneNumberToDial);.
w is used to add wait of 0.5 seconds.

It works OK now
Thanks for help

On Fri, Oct 30, 2009 at 3:41 AM, C F shma...@gmail.com wrote:

 On Wed, Oct 28, 2009 at 10:57 PM, ABBAS SHAKEEL
 shakeel.abbas@gmail.com wrote:
  C F thankyou very much.
 
  when i make a call to Asterisk server recieves and works fine. But as to
  make external calls we have to press nine so supposed a logic to dial 9
  first then wait and then dail other number. But as i dail 9 asterisk show
  the call as connected with alot of noise. Please help in how to handle
 this

 How are you connected from astersik to the TDA?

 
 
 On a side note, may I ask why you are integrating asterisk with the
 TDA? What is the functionality you plan on gaining?
  Nothing very important logical its a client who don't want to trash its
  existing system. So we need to do that. I know Asterisk is far more
 better
  to use and handle his requirements but 

 What requirement? Asterisk is NOT the solution to everything. If fact
 for some it might create more headaches than you would wish.

 In any event what exactly is the Asterisk system adding here that
 Panasonic couldn't handle?


 
 
  On Thu, Oct 29, 2009 at 5:25 AM, C F shma...@gmail.com wrote:
 
  Any simple legacy integration will work. Search on voip-info.org
  Here are some problems that I know exist with panasonic systems on
  their SLT (analog) ports:
  1. No CPC, Asterisk if connected using station ports on the TDA to FXO
  on asterisk, will not detect hangups since the TDA will not send them.
  2. BLF and the like will not work.
  3. There are different ways of making sure that asterisk users should
  be able to use the lines on the TDA depending on how you chose to
  connect them both.
 
  On a side note, may I ask why you are integrating asterisk with the
  TDA? What is the functionality you plan on gaining?
 
  On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL
  shakeel.abbas@gmail.com wrote:
   Hello
   I have a scenerio to integrate an Existing Panasonic PBX with a new
 PBX
   that
   will be Asterisk system.
   I know that Asterisk can be integrated with existing Panasonic TDA 100
   PBX
   to recieve calls (ie PSTN lines to Panasonic PBX and out lines of
   Panasaonic
   to in of Asterisk PBX).
   --But i am in doubt if i can make Asterisk to make calls outside from
   the
   existing PBX ?(ie usually press nine and then wait for a line. In
   Asterisk
   system we will dail 9 first then wait then dail the number). Please
   share
   your ideas and experience.
   All the calls will be recieved by existing Panasonic PBX and an
 Operator
   will forward calls to Asterisk PBX ... this is requirement.
   Please also let me know which type of hardware will be required at
   Asterisk
   end to handle lines from a PBX.
  
   --
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   Shakeel Abbas
  
  
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  Shakeel Abbas
 
 
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Re: [asterisk-users] Asterisk Server with Panasonic PBX

[C F]

On Wed, Oct 28, 2009 at 10:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
 C F thankyou very much.

 when i make a call to Asterisk server recieves and works fine. But as to
 make external calls we have to press nine so supposed a logic to dial 9
 first then wait and then dail other number. But as i dail 9 asterisk show
 the call as connected with alot of noise. Please help in how to handle this

How are you connected from astersik to the TDA?



On a side note, may I ask why you are integrating asterisk with the
TDA? What is the functionality you plan on gaining?
 Nothing very important logical its a client who don't want to trash its
 existing system. So we need to do that. I know Asterisk is far more better
 to use and handle his requirements but 

What requirement? Asterisk is NOT the solution to everything. If fact
for some it might create more headaches than you would wish.

In any event what exactly is the Asterisk system adding here that
Panasonic couldn't handle?




 On Thu, Oct 29, 2009 at 5:25 AM, C F shma...@gmail.com wrote:

 Any simple legacy integration will work. Search on voip-info.org
 Here are some problems that I know exist with panasonic systems on
 their SLT (analog) ports:
 1. No CPC, Asterisk if connected using station ports on the TDA to FXO
 on asterisk, will not detect hangups since the TDA will not send them.
 2. BLF and the like will not work.
 3. There are different ways of making sure that asterisk users should
 be able to use the lines on the TDA depending on how you chose to
 connect them both.

 On a side note, may I ask why you are integrating asterisk with the
 TDA? What is the functionality you plan on gaining?

 On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL
 shakeel.abbas@gmail.com wrote:
  Hello
  I have a scenerio to integrate an Existing Panasonic PBX with a new PBX
  that
  will be Asterisk system.
  I know that Asterisk can be integrated with existing Panasonic TDA 100
  PBX
  to recieve calls (ie PSTN lines to Panasonic PBX and out lines of
  Panasaonic
  to in of Asterisk PBX).
  --But i am in doubt if i can make Asterisk to make calls outside from
  the
  existing PBX ?(ie usually press nine and then wait for a line. In
  Asterisk
  system we will dail 9 first then wait then dail the number). Please
  share
  your ideas and experience.
  All the calls will be recieved by existing Panasonic PBX and an Operator
  will forward calls to Asterisk PBX ... this is requirement.
  Please also let me know which type of hardware will be required at
  Asterisk
  end to handle lines from a PBX.
 
  --
  Best Regards
  Shakeel Abbas
 
 
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 --
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 Shakeel Abbas


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Re: [asterisk-users] Asterisk Server with Panasonic PBX

[C F]

Any simple legacy integration will work. Search on voip-info.org
Here are some problems that I know exist with panasonic systems on
their SLT (analog) ports:
1. No CPC, Asterisk if connected using station ports on the TDA to FXO
on asterisk, will not detect hangups since the TDA will not send them.
2. BLF and the like will not work.
3. There are different ways of making sure that asterisk users should
be able to use the lines on the TDA depending on how you chose to
connect them both.

On a side note, may I ask why you are integrating asterisk with the
TDA? What is the functionality you plan on gaining?

On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
 Hello
 I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that
 will be Asterisk system.
 I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX
 to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic
 to in of Asterisk PBX).
 --But i am in doubt if i can make Asterisk to make calls outside from the
 existing PBX ?(ie usually press nine and then wait for a line. In Asterisk
 system we will dail 9 first then wait then dail the number). Please share
 your ideas and experience.
 All the calls will be recieved by existing Panasonic PBX and an Operator
 will forward calls to Asterisk PBX ... this is requirement.
 Please also let me know which type of hardware will be required at Asterisk
 end to handle lines from a PBX.

 --
 Best Regards
 Shakeel Abbas


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Re: [asterisk-users] Asterisk Server with Panasonic PBX

[ABBAS SHAKEEL]

C F thankyou very much.


when i make a call to Asterisk server recieves and works fine. But as to
make external calls we have to press nine so supposed a logic to dial 9
first then wait and then dail other number. But as i dail 9 asterisk show
the call as connected with alot of noise. Please help in how to handle this



On a side note, may I ask why you are integrating asterisk with the
TDA? What is the functionality you plan on gaining?

Nothing very important logical its a client who don't want to trash its
existing system. So we need to do that. I know Asterisk is far more better
to use and handle his requirements but 



On Thu, Oct 29, 2009 at 5:25 AM, C F shma...@gmail.com wrote:

 Any simple legacy integration will work. Search on voip-info.org
 Here are some problems that I know exist with panasonic systems on
 their SLT (analog) ports:
 1. No CPC, Asterisk if connected using station ports on the TDA to FXO
 on asterisk, will not detect hangups since the TDA will not send them.
 2. BLF and the like will not work.
 3. There are different ways of making sure that asterisk users should
 be able to use the lines on the TDA depending on how you chose to
 connect them both.

 On a side note, may I ask why you are integrating asterisk with the
 TDA? What is the functionality you plan on gaining?

 On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL
 shakeel.abbas@gmail.com wrote:
  Hello
  I have a scenerio to integrate an Existing Panasonic PBX with a new PBX
 that
  will be Asterisk system.
  I know that Asterisk can be integrated with existing Panasonic TDA 100
 PBX
  to recieve calls (ie PSTN lines to Panasonic PBX and out lines of
 Panasaonic
  to in of Asterisk PBX).
  --But i am in doubt if i can make Asterisk to make calls outside from
 the
  existing PBX ?(ie usually press nine and then wait for a line. In
 Asterisk
  system we will dail 9 first then wait then dail the number). Please share
  your ideas and experience.
  All the calls will be recieved by existing Panasonic PBX and an Operator
  will forward calls to Asterisk PBX ... this is requirement.
  Please also let me know which type of hardware will be required at
 Asterisk
  end to handle lines from a PBX.
 
  --
  Best Regards
  Shakeel Abbas
 
 
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Re: [asterisk-users] Asterisk server and DSCP QOS

[Darryl Dunkin]

We're using 184 here (aka TOS 5/EF).

Not set by iptables though, instead it is set in sip.conf
(tos_sip/tos_audio) and on our Polycom/Cisco phones.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Johnson
Sent: Wednesday, December 05, 2007 12:49
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk server and DSCP QOS

Can anyone comment on the DSCP quality of service settings on your
Asterisk server?

The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.

What iptables settings are you using on the Asterisk server for DSCP?
What are your Polycom DSCP settings?  We're using the Linksys SGE2000P
POE switch which supports QOS via DSCP.

Thanks a lot for any info.
Steve

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Re: [asterisk-users] Asterisk server and DSCP QOS

[Darryl Dunkin]

Looks fine to me, you only need to specify DSCP or TOS (may need to
check the manual for which it takes first).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Johnson
Sent: Wednesday, December 05, 2007 14:02
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Asterisk server and DSCP QOS

Thanks, Darryl,

To clarify:

in /etc/asterisk/sip.conf you have the lines:

tos_sip=cs3; Sets TOS for SIP packets.
tos_audio=ef   ; Sets TOS for RTP audio packets.

and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you
have something like (this is the one I'm uncertain about):

   QOS
  Ethernet
 RTP qos.ethernet.rtp.user_priority=5/
 CallControl qos.ethernet.callControl.user_priority=5/
 Other qos.ethernet.other.user_priority=2/
  /Ethernet
  IP
 RTP qos.ip.rtp.dscp=184 qos.ip.rtp.min_delay=1
qos.ip.rtp.max_throughput=1 qos.ip.rtp.max_reliability=0
qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=5/
 CallControl qos.ip.callControl.dscp=184
qos.ip.callControl.min_delay=1 qos.ip.callControl.max_throughput=0
qos.ip.callControl.max_reliability=0 qos.ip.callControl.min_cost=0
qos.ip.callControl.precedence=5/
  /IP
   /QOS

Thanks again!
Steve


Darryl Duncan wrote:

We're using 184 here (aka TOS 5/EF).

Not set by iptables though, instead it is set in sip.conf
(tos_sip/tos_audio) and on our Polycom/Cisco phones.

-Original Message-
Subject: [asterisk-users] Asterisk server and DSCP QOS

Can anyone comment on the DSCP quality of service settings on your
Asterisk server?

The network we're setting up has data on the default VLAN, Asterisk
server and phones on VLAN 4, and we're using Polycom phones with a PC
hooked up to the phone's pass-thru port.

What iptables settings are you using on the Asterisk server for DSCP?
What are your Polycom DSCP settings?  We're using the Linksys SGE2000P
POE switch which supports QOS via DSCP.

Thanks a lot for any info.
Steve

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Re: [asterisk-users] Asterisk server and DSCP QOS

[Igor A. Goncharovsky]

Hi!
Steve Johnson wrote:
 The network we're setting up has data on the default VLAN, Asterisk
 server and phones on VLAN 4, and we're using Polycom phones with a PC
 hooked up to the phone's pass-thru port.
   
If you are using VLAN, than you also look at new options in trunk
cos_sip and cos_audio to set 802.1p. (If you run Linux). It will help
with QoS too.


-- 
Best regards,
Igor A. Goncharovsky

ICQ: 648337
mailto: [EMAIL PROTECTED]
 


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Re: [asterisk-users] Asterisk server hangs on after only few hours again.

[Eric \ManxPower\ Wieling]

johnny_xing wrote:
hi, everyone, 
i have been sufferred for the asterisk hang on problem for so long and i

just reinstalled the whole thing yesterday, but again this morning the
server hangs on again, you could not call in through PSTN line and the ppl
also could not call out throught the server, there is simply engaged dial
tone when you try to do so. and the only thing i can do is to restart
asterisk server after some hours or one day. i am using asterisk 1.2.17 +
zaptel 1.2.16 + freepbx 2.2.1. 




Any one please give me some advice on this? thanks so much really, or how I
can monitor and debug the problems when I happened again next time.


My guess would be that you have a hardware issue.  Either a bad piece of 
hardware or a hardware compatibility issue.  Check the output of cat 
/proc/interrupts to make sure you don't have any IRQ sharing.  I have 
personally not had good luck with Digium analog cards, but most people 
seem to use then without issue.

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RE: [asterisk-users] asterisk server as a voicemail server forlegacy PBX -- FXO or FXS???

[Jeronimo Romero]

 I believe it will be hooked up to extension lines. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, February 05, 2007 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk server as a voicemail server
forlegacy PBX -- FXO or FXS???


Will the Asterisk box be hooked up to external lines on the Merlin, or
extension lines?

External - FXS

Extension - FXO

later,

PaulH

On Mon, 2007-02-05 at 20:03 -0500, Jeronimo Romero wrote:
 Hey All, 
 
  
 
 I'll be configuring an asterisk box to be the voicemail server to an
 old Merlin system which had an octel 100 voicemail server that is now
 dying. 
 
 My question is simple: do I need to stick an FXO card in the asterisk
 box? My logic is that if the Merlin Magix system is actually
 generating electrical current, then I would need to have an fxo card.
  Is this correct?
 
  
 
 
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RE: [asterisk-users] asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???

[Jeronimo Romero]

Thanks. Is there a way I can log into the Merlin Magix to determine
that? How else do I tell?

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Germann
Sent: Monday, February 05, 2007 8:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] asterisk server as a voicemail server
forlegacyPBX -- FXO or FXS???

 

FXS cards generate ring (you connect a station to it and it rings).

 

FXO cards sink ring (they take ring from the office).

 

If the Octel needs ring (which it most likely does), you would need an
FXS card to generate ring for it to answer.  An FXO would take ring from
the vmail server, which, in context, doesn't make a lot of sense (vmail
doesn't call the PBX, the PBX calls vmail).

 

EKG

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, February 05, 2007 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk server as a voicemail server for
legacyPBX -- FXO or FXS???

Hey All, 

 

I'll be configuring an asterisk box to be the voicemail server to an old
Merlin system which had an octel 100 voicemail server that is now dying.


My question is simple: do I need to stick an FXO card in the asterisk
box? My logic is that if the Merlin Magix system is actually generating
electrical current, then I would need to have an fxo card.  Is this
correct?

 

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RE: [asterisk-users] asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???

[Jeronimo Romero]

here's what I found on voip-info.org

http://www.voip-info.org/wiki/index.php?page=Avaya+or+Lucent+Magix+Voice
mail+Integration

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Germann
Sent: Monday, February 05, 2007 8:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] asterisk server as a voicemail server
forlegacyPBX -- FXO or FXS???

 

FXS cards generate ring (you connect a station to it and it rings).

 

FXO cards sink ring (they take ring from the office).

 

If the Octel needs ring (which it most likely does), you would need an
FXS card to generate ring for it to answer.  An FXO would take ring from
the vmail server, which, in context, doesn't make a lot of sense (vmail
doesn't call the PBX, the PBX calls vmail).

 

EKG

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, February 05, 2007 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk server as a voicemail server for
legacyPBX -- FXO or FXS???

Hey All, 

 

I'll be configuring an asterisk box to be the voicemail server to an old
Merlin system which had an octel 100 voicemail server that is now dying.


My question is simple: do I need to stick an FXO card in the asterisk
box? My logic is that if the Merlin Magix system is actually generating
electrical current, then I would need to have an fxo card.  Is this
correct?

 

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Re: [asterisk-users] asterisk server as a voicemail server for legacyPBX -- FXO or FXS???

[John Novack]

He had it right
He is using Asterisk to REPLACE the Octal, so it needs to be equipped 
with FXO
Since most.all VM's should have multiple ports to the PBX, you probably 
will want to equip the Asterisk box with a Sangoma card with 2 FXO 
modules, for a total of four ports.
This will allow the users to get their VM, the PBX to send calls to VM, 
and also allow Asterisk to tell the PBX to light the MW lights when a 
message is left.

VM programming to interface to the PBX can be a fun job.
You may want to look for a used Keyvoice VM system on eBay and save your 
Asterisk project to replace the PBX as well.


JMO

John Novack


Eric Germann wrote:

FXS cards generate ring (you connect a station to it and it rings).
FXO cards sink ring (they take ring from the office).
If the Octel needs ring (which it most likely does), you would need an 
FXS card to generate ring for it to answer. An FXO would take ring 
from the vmail server, which, in context, doesn't make a lot of sense 
(vmail doesn't call the PBX, the PBX calls vmail).

EKG


*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of 
*Jeronimo Romero

*Sent:* Monday, February 05, 2007 8:03 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] asterisk server as a voicemail server for 
legacyPBX -- FXO or FXS???


Hey All,

I’ll be configuring an asterisk box to be the voicemail server to an 
old Merlin system which had an octel 100 voicemail server that is now 
dying.


My question is simple: do I need to stick an FXO card in the asterisk 
box? My logic is that if the Merlin Magix system is actually 
generating electrical current, then I would need to have an fxo card. 
Is this correct?




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Re: [asterisk-users] asterisk server as a voicemail server for legacy PBX -- FXO or FXS???

[Paul Hales]

Will the Asterisk box be hooked up to external lines on the Merlin, or
extension lines?

External - FXS

Extension - FXO

later,

PaulH

On Mon, 2007-02-05 at 20:03 -0500, Jeronimo Romero wrote:
 Hey All, 
 
  
 
 I’ll be configuring an asterisk box to be the voicemail server to an
 old Merlin system which had an octel 100 voicemail server that is now
 dying. 
 
 My question is simple: do I need to stick an FXO card in the asterisk
 box? My logic is that if the Merlin Magix system is actually
 generating electrical current, then I would need to have an fxo card.
  Is this correct?
 
  
 
 
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RE: [asterisk-users] asterisk server as a voicemail server for legacyPBX -- FXO or FXS???

[Eric Germann]

FXS cards generate ring (you connect a station to it and it rings).
 
FXO cards sink ring (they take ring from the office).
 
If the Octel needs ring (which it most likely does), you would need an FXS
card to generate ring for it to answer.  An FXO would take ring from the
vmail server, which, in context, doesn't make a lot of sense (vmail doesn't
call the PBX, the PBX calls vmail).
 
EKG
 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, February 05, 2007 8:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk server as a voicemail server for
legacyPBX -- FXO or FXS???



Hey All, 

 

I'll be configuring an asterisk box to be the voicemail server to an old
Merlin system which had an octel 100 voicemail server that is now dying. 

My question is simple: do I need to stick an FXO card in the asterisk box?
My logic is that if the Merlin Magix system is actually generating
electrical current, then I would need to have an fxo card.  Is this correct?

 

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Re: [asterisk-users] asterisk server RFC conformance

[Stephen Bosch]

A S wrote:
 Hi Asterisk Gurus,
 I am new to Asterisk server. We are trying to use Asterisk for testing
 one of our new products. I was wondering if anyone can tell me if it is
 RFC compliant or how can i use Asterisk to test it for some basic RFC
 compliance.
  Thanks in Advance,

That's an awfully broad question. Which RFCs are you thinking of in
particular?

-Stephen-

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Re: [asterisk-users] Asterisk server reports

[Andrew Joakimsen]

Take a look at Asterisk-Stat
http://www.areski.net/asterisk-stat-v2/about.php

pretty close

On 11/27/06, Frederico Madeira [EMAIL PROTECTED] wrote:


Hi guys,

It's possible i scheduler in cron some kind of script or application that
read asterisk logs and send via e-mail a complete report for pbx activity in
specified period  ??

I like to see how simultanios calls was made, total time in conversation,
averege time of calls, most routes calls, etc

Thanks.

--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash

[Stuart Sheldon]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I would look at ventilation if I were you. Drive failures at the rate
you are talking about can usually be traced back to thermal failures.

Just a thought

Stu


Dushyanth wrote:
 Hey guys,
 
 Iam having a peculiar problem with my asterisk installation. The specs 
 are..
 
 [EMAIL PROTECTED] ~]# asterisk -V
 Asterisk 1.2.7.1
 
 Wildcard: Digium Wildcard TE110P T1/E1
 Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
 Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS)
 Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest 
 empty)
 
 Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have 
 about 300 active SIP accounts. 
 
 Queues, SIP extensions, Agents are in MySQL database using asterisk 
 realtime static.
 
 CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading
 RAM : 1G
 Mobo : Intel SE7501HG2
 
 The system is stable, however, the IDE disk crashes every 3/4 months. There 
 are DMA timeout errors for the IDE disk before it fails completely. The 
 same issue occured for the past three disks and I was doubting the 
 recommended hdparm setting 
 
 'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device'
 
 So, I removed this setting after the last crash and the system workd fine 
 for another 3 months. Yes'day, the disk failed again with same symptoms. 
 All the disks were seagate baraccuda IDE drives.
 
 zttool doesnt show any IRQ misses even without the above hdparm setting and
 there is no noticeable problem in asterisk with the PRI line etc. Below is 
 my /proc/interrupts as well as /dev/hda settings.
 
 [EMAIL PROTECTED] ~]# cat /proc/interrupts
CPU0   CPU1
   0:   24771857   24719039IO-APIC-edge  timer
   1:102 62IO-APIC-edge  i8042
   8:  1  0IO-APIC-edge  rtc
   9:  0  0   IO-APIC-level  acpi
  14: 134159 135915IO-APIC-edge  ide0
 185:   32988610   16463264   IO-APIC-level  wctdm
 193:   22173177   27275710   IO-APIC-level  wctdm
 201:   21737611   27711650   IO-APIC-level  wctdm24xxp
 209:   22038077   27401613   IO-APIC-level  wcte11xp
 225:   18992311  0   IO-APIC-level  eth1
 233:1171166879   IO-APIC-level  eth0
 NMI:  0  0
 LOC:   49493157   49493156
 ERR:  0
 MIS:  0
 
 [EMAIL PROTECTED] ~]# hdparm -i /dev/hda
 
 /dev/hda:
 
  Model=ST340014A, FwRev=3.06, SerialNo=5JX96VFV
  Config={ HardSect NotMFM HdSw15uSec Fixed DTR10Mbs RotSpdTol.5% }
  RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=4
  BuffType=unknown, BuffSize=2048kB, MaxMultSect=16, MultSect=16
  CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
  IORDY=on/off, tPIO={min:240,w/IORDY:120}, tDMA={min:120,rec:120}
  PIO modes:  pio0 pio1 pio2 pio3 pio4
  DMA modes:  mdma0 mdma1 mdma2
  UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
  AdvancedPM=no WriteCache=enabled
  Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2:
 
  * signifies the current active mode
 
 I looked at the mailing lists and couldnt any such issues reported. 
 
 Please advice. Should i be using SCSI disks on RAID 1 or something ? Will 
 that help ?
 
 Also, should i be looking at any other mobo then Intel SE7501HG2 ? Iam 
 planning to put in a another asterisk server as a failover and would 
 appreciate inputs abt the kind of hardware i should be using for the system 
 with the specs i mentioned.
 
 Thanks
 Dushyanth
 
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Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash

[Sam Norris]

Heat = #1 cause of disk failure. If they are roasting to the touch they will 
fail in 2-3 months.


- Original Message - 
From: Dushyanth [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: 10/05/2006 9:44 AM
Subject: [asterisk-users] Asterisk Server : IDE HDD frequent crash



Hey guys,

Iam having a peculiar problem with my asterisk installation. The specs
are..

[EMAIL PROTECTED] ~]# asterisk -V
Asterisk 1.2.7.1

Wildcard: Digium Wildcard TE110P T1/E1
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 2 FXO, 2 FXS)
Wildcard TDM: Wildcard TDM400P REV I (4 modules) ( 1 FXO, 3 FXS)
Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) (12 FXO's - rest
empty)

Total 15 FX0's, 5 FXS out of which 5 to 6 FXO/FXS are being used. We have
about 300 active SIP accounts.

Queues, SIP extensions, Agents are in MySQL database using asterisk
realtime static.

CPU : Intel(R) Xeon(TM) CPU 3.06GHz with Hyper threading
RAM : 1G
Mobo : Intel SE7501HG2

The system is stable, however, the IDE disk crashes every 3/4 months. 
There

are DMA timeout errors for the IDE disk before it fails completely. The
same issue occured for the past three disks and I was doubting the
recommended hdparm setting

'hdparm -d 1 -X udma2 -c 3 /dev/IDE Device'

So, I removed this setting after the last crash and the system workd fine
for another 3 months. Yes'day, the disk failed again with same symptoms.
All the disks were seagate baraccuda IDE drives.

zttool doesnt show any IRQ misses even without the above hdparm setting 
and

there is no noticeable problem in asterisk with the PRI line etc. Below is
my /proc/interrupts as well as /dev/hda settings.

[EMAIL PROTECTED] ~]# cat /proc/interrupts
  CPU0   CPU1
 0:   24771857   24719039IO-APIC-edge  timer
 1:102 62IO-APIC-edge  i8042
 8:  1  0IO-APIC-edge  rtc
 9:  0  0   IO-APIC-level  acpi
14: 134159 135915IO-APIC-edge  ide0
185:   32988610   16463264   IO-APIC-level  wctdm
193:   22173177   27275710   IO-APIC-level  wctdm
201:   21737611   27711650   IO-APIC-level  wctdm24xxp
209:   22038077   27401613   IO-APIC-level  wcte11xp
225:   18992311  0   IO-APIC-level  eth1
233:1171166879   IO-APIC-level  eth0
NMI:  0  0
LOC:   49493157   49493156
ERR:  0
MIS:  0

[EMAIL PROTECTED] ~]# hdparm -i /dev/hda

/dev/hda:

Model=ST340014A, FwRev=3.06, SerialNo=5JX96VFV
Config={ HardSect NotMFM HdSw15uSec Fixed DTR10Mbs RotSpdTol.5% }
RawCHS=16383/16/63, TrkSize=0, SectSize=0, ECCbytes=4
BuffType=unknown, BuffSize=2048kB, MaxMultSect=16, MultSect=16
CurCHS=16383/16/63, CurSects=16514064, LBA=yes, LBAsects=78165360
IORDY=on/off, tPIO={min:240,w/IORDY:120}, tDMA={min:120,rec:120}
PIO modes:  pio0 pio1 pio2 pio3 pio4
DMA modes:  mdma0 mdma1 mdma2
UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5
AdvancedPM=no WriteCache=enabled
Drive conforms to: ATA/ATAPI-6 T13 1410D revision 2:

* signifies the current active mode

I looked at the mailing lists and couldnt any such issues reported.

Please advice. Should i be using SCSI disks on RAID 1 or something ? Will
that help ?

Also, should i be looking at any other mobo then Intel SE7501HG2 ? Iam
planning to put in a another asterisk server as a failover and would
appreciate inputs abt the kind of hardware i should be using for the 
system

with the specs i mentioned.

Thanks
Dushyanth

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Re: [asterisk-users] Asterisk Server : IDE HDD frequent crash

[Jay R. Ashworth]

On Thu, Oct 05, 2006 at 11:41:32PM -0700, Sam Norris wrote:
 Heat = #1 cause of disk failure. If they are roasting to the touch they 
 will fail in 2-3 months.

One word: smartd.

I didn't know it existed, and I'm amazed I didn't.  Everyone on this
list should be running smartd, and know what it's saying.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Asterisk Server Down

[Patrick Cervicek]

[EMAIL PROTECTED] schrieb:
I rebooted the server on which the Asterisk is hosted on. The * did not 
come back up and I get this message when I attempt to use CLI
 
[EMAIL PROTECTED] ~]# asterisk -r

Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)


Did you configure your server in /etc/rc?.d/, that it sould start after 
reboot?


Tools:
rcconf (debian)
chkconfig (fedora,redhat)

What does /var/log/asterisk say?
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RE: [asterisk-users] Asterisk Server Down

[Guido Hecken]

Hi,

obviously asterisk doesn't start with the installed(?) start script.
Try to start it manually and watch the cli for informations with
asterisk -vvvc
AFAIK a make config in the asterisk source should install the start script
for your system.

Hope it helps...

Guido

Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Gesendet: Sonntag, 17. September 2006 15:27
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Asterisk Server Down

I rebooted the server on which the Asterisk is hosted on. The * did not come
back up and I get this message when I attempt to use CLI
 
[EMAIL PROTECTED] ~]# asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
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Re: [asterisk-users] Asterisk Server Down

[Tzafrir Cohen]

On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
 Hi,
 
 obviously asterisk doesn't start with the installed(?) start script.
 Try to start it manually and watch the cli for informations with
 asterisk -vvvc

One warning: if your system is normally configured to run as non-root,
this may cause it to write some fiels as root, and not start properly
next time you start it with the standard script.

With the Debian packages, use:

/etc/init.d/asterisk debug

Which is normally just a glorified:

  asterisk -U asterisk -vv

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Asterisk Server Down

[Guido Hecken]

 -Ursprüngliche Nachricht-
 Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
 Gesendet: Sonntag, 17. September 2006 15:56
 An: asterisk-users@lists.digium.com
 Betreff: Re: [asterisk-users] Asterisk Server Down
 
 On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
  Hi,
 
  obviously asterisk doesn't start with the installed(?) start script.
  Try to start it manually and watch the cli for informations with
  asterisk -vvvc
 
 One warning: if your system is normally configured to run as non-root,
 this may cause it to write some fiels as root, and not start properly
 next time you start it with the standard script.
 
 With the Debian packages, use:
 
 /etc/init.d/asterisk debug
 
 Which is normally just a glorified:
 
   asterisk -U asterisk -vv

Tzafrir,

you're right, one should proof, under which user asterisk runs...
Besides security reasons, running asterisk as root, doesn't it allow a
higher prioritization of asterisk processes?

Guido

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Re: [asterisk-users] Asterisk Server Down

[Steve Totaro]

Guido Hecken wrote:

-Ursprüngliche Nachricht-
Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 17. September 2006 15:56
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Asterisk Server Down

On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:


Hi,

obviously asterisk doesn't start with the installed(?) start script.
Try to start it manually and watch the cli for informations with
asterisk -vvvc
  

One warning: if your system is normally configured to run as non-root,
this may cause it to write some fiels as root, and not start properly
next time you start it with the standard script.

With the Debian packages, use:

/etc/init.d/asterisk debug

Which is normally just a glorified:

  asterisk -U asterisk -vv



Tzafrir,

you're right, one should proof, under which user asterisk runs...
Besides security reasons, running asterisk as root, doesn't it allow a
higher prioritization of asterisk processes?

Guido

  
I can see a problem with security issues but is it a bad thing to allow 
higher priority of the asterisk process?  Not sure that it does anyways, 
but I don't see how that is a bad thing?

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Re: [asterisk-users] Asterisk Server Down

[Tzafrir Cohen]

On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:

 you're right, one should proof, under which user asterisk runs...
 Besides security reasons, running asterisk as root, doesn't it allow a
 higher prioritization of asterisk processes?

This is why we let asterisk setuid itself to user asterisk, and don't
let the wrappr script handle that. Asterisk sets scheduling priority
before running setuid/setgid .

 I can see a problem with security issues but is it a bad thing to allow 
 higher priority of the asterisk process?  Not sure that it does anyways, 
 but I don't see how that is a bad thing?

It can help the quality of Audio. On the downside, it means that a 100%
CPU loop in asterisk is a pain to recover from. Security implications:
if someone can inject you one line to the dialpan, they can (under the
right circumstances) get your system stuck very badly . Unless you have
a manager connection availble.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Asterisk Server Down

[broadbandvoice]

Thanks everyone it is working now.

-- Original message -- From: Tzafrir Cohen [EMAIL PROTECTED]  On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:you're right, one should proof, under which user asterisk runs...   Besides security reasons, running asterisk as root, doesn't it allow a   higher prioritization of asterisk processes?   This is why we let asterisk setuid itself to user asterisk, and don't  let the wrappr script handle that. Asterisk sets scheduling priority  before running setuid/setgid .I can see a problem with security issues but is it a bad thing to allow   higher priority of the asterisk process? Not sure that it does anyways,   but I don't see how that is a bad thing?   It can help the qu
 ality 
of Audio. On the downside, it means that a 100%  CPU loop in asterisk is a pain to recover from. Security implications:  if someone can inject you one line to the dialpan, they can (under the  right circumstances) get your system stuck very badly . Unless you have  a manager connection availble.   --  Tzafrir Cohen sip:[EMAIL PROTECTED]  icq#16849755 iax:[EMAIL PROTECTED]  +972-50-7952406 jabber:[EMAIL PROTECTED]  [EMAIL PROTECTED] http://www.xorcom.com  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 

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RE: [asterisk-users] asterisk server to server using sip question

[Steven Totaro]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jerry Geis
 Sent: Thursday, September 14, 2006 8:14 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] asterisk server to server using sip question
 
 I have 2 asterisk servers. I am trying to connect them with SIP and
 getting an error.
 My first box I define sip.conf as:
 
 [devcentos64_to_bt610tMM]
 type=friend
 username=devcentos64_to_bt610tMM
 secret=password
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 host=192.168.1.159
 context=default
 
 my second box I define sip.conf as:
 [devcentos64_to_bt610tMM]
 type=friend
 username=devcentos64_to_bt610tMM
 secret=password
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 host=192.168.1.10
 context=default
 
 So Box2 points to Box1 and Box1 points to Box2 by the host= fields.
 
 I am getting the following error:
 -- Attempting call on SIP/devcentos64_to_bt610tmm/1041 for
 [EMAIL PROTECTED]:1 (Retry 1)
 Sep 14 08:10:52 WARNING[2512]: chan_sip.c:9715 handle_response_invite:
 Forbidden - wrong password on authentication for INVITE to 'Jerry
Geis
 204 sip:[EMAIL PROTECTED];tag=as07330b38'
 Channel SIP/devcentos64_to_bt610tmm-007afe00 was never
answered.
 Sep 14 08:10:52 WARNING[4639]: cdr.c:550 ast_cdr_disposition: Cause
not
 handled
 
 
 Why is that??? My passwords match. I am using asterisk.1.2.11
 Or what is the correct way to connect asterisk SIP server to asterisk
 SIP server.
 
 Jerry
 
 


Do you have other peers on the same boxes pointing to each other?

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Re: [asterisk-users] Asterisk server crashes after two years

[Tzafrir Cohen]

On Thu, Aug 31, 2006 at 09:40:50PM -0600, Michael Welter wrote:
 My Asterisk colo server has been up for almost two years.  Today it 
 crashed.  When I gave the reboot command, it crashed so hard that it had 
 to be power cycled.  I wasn't in attendance, but I can speculate that it 
 had a kernel panic during the shutdown.
 
 Yesterday I added a PHP agi script, and it had been user over 1000 times 
 before the crash.  I don't think the Linux/Asterisk crash is coincidental.
 
 Can someone give me things to look for?  I'm watching memory, and it has 
 750MB free (out of 1GB).  When I restart Asterisk, I see 19 
 processes--is this normal?  

Is this kernel 2.4? If so: do they happen to have exactly the same
memory size and the same files open? If so: this is normal: threads of
the same process.

 What else should I be doing to narrow down 
 on this problem.

One way to get a (huge) trace:

strace -f -o path/to/log/file command to start asterisk

Also try starting asterisk without -p, if you normally start it with it.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Asterisk server crashes after two years

[Nir Simionovich]

Hi Tzafrir,

  Actually, it would appear as something is wrong with the PHP script 
Michael is referring to. As far as I understand AGI, for each AGI script
that has to be run, asterisk will fork it self out, run the AGI within
the fork, then return back to asterisk once the AGI is complete. 

  Now, I've written quite a few AGI scripts, some really bad and some 
really good - so I can surely say the following: If you have 1000 asterisk
threads running and you can see your AGI script running 1000 times, then
something in your AGI it surely wrong. 

  Just as an example, I've once built an AGI that was supposed to handle 
some garbage collection at the end of a call. I never assumed that my 
system would may end up running to around 3000 concurrent calls (10 servers),
then that script would hang on all machines waiting for the database. 

  In other words, I would highly suspect the AGI script at this point as
being a little faulty, and I would sure go and examine it freshly. Although,
I wouldn't go and debounce your 2.4 theory, as that one holds water too.

Nir S

- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, September 2, 2006 2:36:23 PM GMT+0200
Subject: Re: [asterisk-users] Asterisk server crashes after two years

On Thu, Aug 31, 2006 at 09:40:50PM -0600, Michael Welter wrote:
 My Asterisk colo server has been up for almost two years.  Today it 
 crashed.  When I gave the reboot command, it crashed so hard that it had 
 to be power cycled.  I wasn't in attendance, but I can speculate that it 
 had a kernel panic during the shutdown.
 
 Yesterday I added a PHP agi script, and it had been user over 1000 times 
 before the crash.  I don't think the Linux/Asterisk crash is coincidental.
 
 Can someone give me things to look for?  I'm watching memory, and it has 
 750MB free (out of 1GB).  When I restart Asterisk, I see 19 
 processes--is this normal?  

Is this kernel 2.4? If so: do they happen to have exactly the same
memory size and the same files open? If so: this is normal: threads of
the same process.

 What else should I be doing to narrow down 
 on this problem.

One way to get a (huge) trace:

strace -f -o path/to/log/file command to start asterisk

Also try starting asterisk without -p, if you normally start it with it.

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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-- 
Kind Regards,
  Nir Simionovich
  Chief Technology Officer
  Atelis PLC

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Re: [asterisk-users] Asterisk server crashes after two years

[Ronald Wiplinger]

Michael Welter wrote:
My Asterisk colo server has been up for almost two years.  Today it 
crashed.  When I gave the reboot command, it crashed so hard that it 
had to be power cycled.  I wasn't in attendance, but I can speculate 
that it had a kernel panic during the shutdown.


Yesterday I added a PHP agi script, and it had been user over 1000 
times before the crash.  I don't think the Linux/Asterisk crash is 
coincidental.


Can someone give me things to look for?  I'm watching memory, and it 
has 750MB free (out of 1GB).  When I restart Asterisk, I see 19 
processes--is this normal?  What else should I be doing to narrow down 
on this problem.


Thanks for your help.



Have you checked the log files?
Do you use Real-time? Is your database ok?
Have you checked the hard disk space?

2 years Asterisk sounds strange, since I can remember there was a bug 
with the date a year ago. If you have not upgraded, than this bug is 
still in your code. Maybe you just meant no reboot for two years.


bye

Ronald
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Re: [asterisk-users] Asterisk server crashes after two years

[Hadley Rich]

On Friday 01 September 2006 16:32, Ronald Wiplinger wrote:
 2 years Asterisk sounds strange, since I can remember there was a bug
 with the date a year ago. If you have not upgraded, than this bug is
 still in your code. Maybe you just meant no reboot for two years.

That bug was only in one version IIRC

hads

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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Re: [Asterisk-Users] Asterisk server

[Zoa]

Its overkill, go get some more employees :)

So yes, its just fine and there's room for expansion.

Zoa

Andrew Nowrot wrote:

Hi,

I have to build Asterisk server for about 30 user (30 concurrent 
calls). I decided to buy this box:


-- motherboard Intel E7210 + Hence Rapids
-- processor P4 3.0 GHz
-- RAM 2x512 MB DDR ECC
-- network interface Intel 82541 GI

Is this configuration enough to handle 30 users at the same time. I am 
not planning to use any transcoding (everything will be alaw).


Cheers

Andrew


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Re: [Asterisk-Users] Asterisk server

[Mats Karlsson]

Andrew Nowrot wrote:

Hi,

I have to build Asterisk server for about 30 user (30 concurrent 
calls). I decided to buy this box:


-- motherboard Intel E7210 + Hence Rapids
-- processor P4 3.0 GHz
-- RAM 2x512 MB DDR ECC
-- network interface Intel 82541 GI

Is this configuration enough to handle 30 users at the same time. I am 
not planning to use any transcoding (everything will be alaw).


Cheers

Andrew


Yes.

/M
-- Those that sacrifice essential liberty to obtain a little temporary 
safety deserve neither liberty nor safety. -- Ben Franklin (1759)

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RE: [Asterisk-Users] Asterisk server

[jacobso1]

Hi,



With 30 users and NO transcoding, that is
certainly enough.

Even if you use real-time
configuration (that requires a SQL server)



Now, if you system will be accessible both
from inside (LAN) and outside (Internet), I would advice a second network card (10/100)



Regards,



T. Jacobson 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot
Sent: mercredi 14 juin 2006 11:23
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk
server





Hi,

I have to build Asterisk server for about 30 user (30 concurrent calls). I
decided to buy this box:

-- motherboard Intel E7210 + Hence Rapids
-- processor P4 3.0 GHz
-- RAM 2x512 MB DDR ECC
-- network interface Intel 82541 GI 

Is this configuration enough to handle 30 users at the same time. I am not
planning to use any transcoding (everything will be alaw).

Cheers

Andrew

--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13/06/2006








--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13/06/2006
 
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Re: [Asterisk-Users] Asterisk server

[Andrew Nowrot]

Thanks for all replies Now, if you system will be accessible both
from inside (LAN) and outside (Internet), I would advice a second network card (10/100)Actually the machine has two interfaces - 1000 and 100 Mbit/s
CheersAndrew
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Re: [Asterisk-Users] Asterisk Server Hangs

[Giovanni Miano]

whats kernel version ? check in dmesg for system messagesCheers,Giovanni Miano2005/12/29, Dushyanth Harinath [EMAIL PROTECTED]
:Hey guys,Asterisk Server Specs :Asterisk version :CLI show version
Asterisk SVN-trunk-r7230 built by [EMAIL PROTECTED] on a i686 running Linuxon 2005-12-25 16:14:47 UTCSystem details :Centos 4.2 (Final)Linux ip-pbx 2.6.9-22.ELsmp #1 SMPIntel Dual Xeon 3.06Ghz
Intel SE7501CW2 MotherboardDigium cards : T110P (E1) , TDM22B, TDM31B, TDM24012BI added TDM24012B yes'day but haven't configured or used it yet. Itsjust connected to the system. The same problem used to occur before
adding TDM24012B to the mix.This setup hangs up i,e total freeze cant ssh, cant login even from thesystem console and nothing in system logs or asterisk logs point me toany obvious problem. There is no coredump in /tmp too.
Asterisk also freezes up the server when i issue a stop now command inthe CLI sometimes.The only call traffic at this moment are SIP to SIP internal calls, SIPto ZAP external calls and ZAP to SIP incoming calls. In all there must
be a total of 10 simultaneous calls.Im using queues, rxfax, txfax, voicemail, meetme (still testing).This happens three or four times in a day.I cant see any IRQ misses in zttool and zttest output is below.
Opened pseudo zap interface, measuring accuracy...99.987793% 99.987793% 99.987793% 99.987793% 100.00% 100.00%99.987793%99.987793% 100.00% 100.00% 100.00% 99.987793% 100.00%99.987793%
 100.00%Best: 100.00 -- Worst: 99.987793 -- Average: 99.992300Found the below messages in dmesg but seems informational rather than aerror.Dec 27 22:04:24 asterisk kernel: zaptel Disabled echo canceller because
of tone (tx) on channel 32Dec 29 21:02:12 asterisk kernel: zaptel Disabled echo canceller becauseof tone (rx) on channel 35I dont know what the problem could be. I followed the doc at
http://www.voip-info.org/wiki-Asterisk+debugging and started asteriskusing safe_asterisk and applied the logger related changes.Wat else i can do to debug this issue ?Dushyanth___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[hgaillac-sip]

Have you test it for  virtuals IPBX ?

--- C F [EMAIL PROTECTED] a écrit :

 The workaround for the parking limitation is
 app_valetparking.so from
 http://www.pbxfreeware.org/app_valetparking.c
 instructions on how to install is on the wiki.
 
 On 12/21/05, Olle E Johansson [EMAIL PROTECTED]
 wrote:
  [EMAIL PROTECTED] wrote:
   Hello,
  
   Is Asterisk able to provide virtuals IPBX ?
   I mean one hardware server which handle one IPBX
 per
   enterprise .
  A lot of service providers do that. One caveat is
 the parking function,
  that only supports one parking lot for all virtual
 PBXs.
 
  /O
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[hgaillac-sip]

Does these providers use blabe servers for reliability
and scalability ?

Harry
--- Olle E Johansson [EMAIL PROTECTED] a écrit :

 [EMAIL PROTECTED] wrote:
  Hello,
  
  Is Asterisk able to provide virtuals IPBX ?
  I mean one hardware server which handle one IPBX
 per
  enterprise .
 A lot of service providers do that. One caveat is
 the parking function,
 that only supports one parking lot for all virtual
 PBXs.
 
 /O
 







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RE: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Kevin Kiely]

App_valetparking is a great (and necessary) addition to asterisk. Does
app_valetparking.c work with the current release of asterisk?  I tried
to install it on Asterisk 1.0.9 and I get errors following the
instruction in the wiki?

app_valetparking.c:678: dereferencing pointer to incomplete type


-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED] 
Sent: Thursday, December 22, 2005 2:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

Christopher L. Wade wrote:
 On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote:
 
The workaround for the parking limitation is app_valetparking.so from
http://www.pbxfreeware.org/app_valetparking.c
instructions on how to install is on the wiki.

On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:

[EMAIL PROTECTED] wrote:

Hello,

Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .

A lot of service providers do that. One caveat is the parking
function,
that only supports one parking lot for all virtual PBXs.

/O
 
 
 There is also a work in progress in svn to add context support to the
 builtin asterisk parking.  I forget which developer is working on it
but
 it should be hard to find if you check the asterisk-commits archive on
 lists.digium.com.

That would be me :-)


It is in the multiparking branch.

/O
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[C F]

The on on pbxfreeware works with 1.2.1

On 12/22/05, Kevin Kiely [EMAIL PROTECTED] wrote:
 App_valetparking is a great (and necessary) addition to asterisk. Does
 app_valetparking.c work with the current release of asterisk?  I tried
 to install it on Asterisk 1.0.9 and I get errors following the
 instruction in the wiki?

 app_valetparking.c:678: dereferencing pointer to incomplete type


 -Original Message-
 From: Olle E Johansson [mailto:[EMAIL PROTECTED]
 Sent: Thursday, December 22, 2005 2:44 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

 Christopher L. Wade wrote:
  On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote:
 
 The workaround for the parking limitation is app_valetparking.so from
 http://www.pbxfreeware.org/app_valetparking.c
 instructions on how to install is on the wiki.
 
 On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
 
 [EMAIL PROTECTED] wrote:
 
 Hello,
 
 Is Asterisk able to provide virtuals IPBX ?
 I mean one hardware server which handle one IPBX per
 enterprise .
 
 A lot of service providers do that. One caveat is the parking
 function,
 that only supports one parking lot for all virtual PBXs.
 
 /O
 
 
  There is also a work in progress in svn to add context support to the
  builtin asterisk parking.  I forget which developer is working on it
 but
  it should be hard to find if you check the asterisk-commits archive on
  lists.digium.com.

 That would be me :-)


 It is in the multiparking branch.

 /O
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Rehan Ahmed]

Hello

You can try www.ip-pabx.com its a commercial centrex product, the server is also available for sale on www.supertec.com/solutions/


Rehan

On 12/23/05, C F [EMAIL PROTECTED] wrote:
The on on pbxfreeware works with 1.2.1On 12/22/05, Kevin Kiely 
[EMAIL PROTECTED] wrote: App_valetparking is a great (and necessary) addition to asterisk. Does app_valetparking.c work with the current release of asterisk?I tried to install it on Asterisk 
1.0.9 and I get errors following the instruction in the wiki? app_valetparking.c:678: dereferencing pointer to incomplete type -Original Message- From: Olle E Johansson [mailto:
[EMAIL PROTECTED]] Sent: Thursday, December 22, 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX
 Christopher L. Wade wrote:  On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote:  The workaround for the parking limitation is app_valetparking.so from 
http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki.  On 12/21/05, Olle E Johansson 
[EMAIL PROTECTED] wrote:  [EMAIL PROTECTED] wrote:  Hello,
  Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . 
 A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs.  /O
There is also a work in progress in svn to add context support to the  builtin asterisk parking.I forget which developer is working on it but  it should be hard to find if you check the asterisk-commits archive on
  lists.digium.com. That would be me :-) It is in the multiparking branch. /O ___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Rehan Ahmed AllahWala
http://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service.
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Dmitry Ivanov]

On Wednesday 21 December 2005 15:11, [EMAIL PROTECTED] wrote:
 Is Asterisk able to provide virtuals IPBX ?
 I mean one hardware server which handle one IPBX per
 enterprise .

Yes, just create separate context for each enterprise.
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Olle E Johansson]

[EMAIL PROTECTED] wrote:

Hello,

Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .

A lot of service providers do that. One caveat is the parking function,
that only supports one parking lot for all virtual PBXs.

/O
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[C F]

The workaround for the parking limitation is app_valetparking.so from
http://www.pbxfreeware.org/app_valetparking.c
instructions on how to install is on the wiki.

On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
 [EMAIL PROTECTED] wrote:
  Hello,
 
  Is Asterisk able to provide virtuals IPBX ?
  I mean one hardware server which handle one IPBX per
  enterprise .
 A lot of service providers do that. One caveat is the parking function,
 that only supports one parking lot for all virtual PBXs.

 /O
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Christopher L. Wade]

On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote:
 The workaround for the parking limitation is app_valetparking.so from
 http://www.pbxfreeware.org/app_valetparking.c
 instructions on how to install is on the wiki.
 
 On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
  [EMAIL PROTECTED] wrote:
   Hello,
  
   Is Asterisk able to provide virtuals IPBX ?
   I mean one hardware server which handle one IPBX per
   enterprise .
  A lot of service providers do that. One caveat is the parking function,
  that only supports one parking lot for all virtual PBXs.
 
  /O

There is also a work in progress in svn to add context support to the
builtin asterisk parking.  I forget which developer is working on it but
it should be hard to find if you check the asterisk-commits archive on
lists.digium.com.

--
Chris
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Christopher L. Wade]

s/should/shouldn't/

--
Chris
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Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX

[Olle E Johansson]

Christopher L. Wade wrote:

On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote:


The workaround for the parking limitation is app_valetparking.so from
http://www.pbxfreeware.org/app_valetparking.c
instructions on how to install is on the wiki.

On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:


[EMAIL PROTECTED] wrote:


Hello,

Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .


A lot of service providers do that. One caveat is the parking function,
that only supports one parking lot for all virtual PBXs.

/O



There is also a work in progress in svn to add context support to the
builtin asterisk parking.  I forget which developer is working on it but
it should be hard to find if you check the asterisk-commits archive on
lists.digium.com.


That would be me :-)


It is in the multiparking branch.

/O
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Re: [Asterisk-Users] Asterisk server behind NAT, and SIP clinet behind another NAT.

[Martinez Felix]

you need a stun server on asterisk side...I use the one that vovida.org provides...it is very easy to install and configure...On 11/23/05, 
jeffery chen [EMAIL PROTECTED] wrote:
Asterisk server behind NAT,and SIP clinet behind another NAT.SIP.conf have set NAT=yes,SIP client can register with Asterisk server, but can not hearing anything..PLS help me, how to resolve this trouble,,
As refer to the item 9http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutionsI can not register with Asterisk server too, how this happen..
_Don't just search. Find. Check out the new MSN Search!http://search.msn.click-url.com/go/onm00200636ave/direct/01/
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Marie]

I've yet to run into a Co-Lo facility that didn't offer a reboot
service. Yes, some charged me for it, but it always seemed to be an
option.

Check your machine before purchasing the remote control power strips.
I bought one for a troublesome server that was not capable of
automatically repowering itself. Made for a rather expensive D'oh.

Most of my Asterisk boxes are running full Linux installs (gui, etc --
not stripped down -- vnc, webserver, and Oracle one) and go without
trouble for months and months -- the only time I've had to reboot them
is when I do something stupid.

For what the remote power strips cost, you're better off determine
what the problem is to begin with and investing in new hardware (if
necessary). Any of these solutions worth having are not cheap. It's
not going to help you much if you buy a crap remote power solution
that ends up needing to be monitored itself.
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[beonice]

--- Tzafrir Cohen [EMAIL PROTECTED] wrote:

 Hi
 
 On Thu, Jun 23, 2005 at 05:52:44PM -0700, beonice
 wrote:
  
  Okay, so what makes more sense:
1) a remote management card that will let me
  actually log in to the machine to monitor it as
 well
  as to reboot it
  vs.
2) a remote-accessible powerstrip that will
 allow me
  to remotely reboot the server?
 
 Linux also has a software watchdog module. Maybe it
 could work here
 without extra hardware to set up.
 
  
  I'm realising that sometimes the problem may
 simply be
  processes out of control, and may be something
 that
  doesn't require killing the entire machine, but
 just
  some processes. 
  
  In my current setup (an ordinary PIII 1.someting
 GHz
  machine, not a server-class machine), when a
 process
  goes haywire, I lose remote access via SSH, so I
 drive
  to the colo, log in, sigh in frustration, and
 reboot
  because I'm already here, so why not?.
 
 Because you destroy any evidence of the problem.
 
 What processes are taking much CPU time? Are there
 any relevant log
 messages? Is this a case of over-swapping? (not 100%
 CPU usage, but
 rather large swap usage, CPU spends too much time at
 system, though 
 the latter may be probably normal for an Asterisk
 server).
 
 Could you login from the console? Did you manage to
 move between virtual
 consoles?
 
 Install the package sysstat and run sar to get some
 stats. Consider
 adding a cron job to gather more relevant stats
 every 5 minutes or every
 minute.
 
 BTW: does asterisk run with real-time priority? try
 removing it, so at
 least asterisk won't hang the whole system. Though I
 doubt it if this
 would help.
 
  
  Some of the problems were caused by my old router
 ...
  since I replaced it, the need to drive the 40
 miles
  each way has gone down significantly ... in fact,
 to
  pretty much zero. So I have time to contemplate my
  options here. :)
 
 You have an extra router there? I recall that there
 was a kernel patch
 to reboot the system upon recieving a specific ICMP
 packet.
 
 -- 
 Tzafrir Cohen | [EMAIL PROTECTED] |
 VIM is
 http://tzafrir.org.il |   |
 a Mutt's  
 [EMAIL PROTECTED] |   | 
 best
 ICQ# 16849755 |   |
 friend
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Thanks, everyone, for all the responses and
suggestions. I already do have remote access via SSH.
The problem is that, occasionally, the box simply
won't let me SSH in, so I have to drive over and
reboot.

Several of those times, it turned out to be simply the
router acting up, so power-cycling the router would
bring things back to life, including SSH. As I
mentioned, the router has now been replaced and the
new one seems a lot more stable.

However, on a _few_ occasions, when I went into the
colo, it turned out that the box itself was not
responding to input, even on the console. At those
times, the ONLY thing I could do was a hard reboot ...
and yes, I'm aware of the potential hazards involved
in a hard reboot. :)

The frustrating thing is that since I upgraded the
router, the box hasn't crashed at all ... now I'm
thinking back and wondering if I imagined those
occasions. :)

Based on the suggestions provided by all of you, I
think I'm definitely going to try to get better stats
on what exactly is going on (thanks for the tip that
there are production Asterisk servers with
months/years uptime ... that was an eye-opener!)
before I invest in any new hardware.

Thanks again, everyone! 

Cheers,
Maya




 
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Max Clark]

Maya,

Where are you colocated? Remote reboot is something that we offer our 
customers standard with the rack space.


I have found the Baytech products to be fantastic for remote 
reboot/remote serial access. You might want to look for something like 
this: http://www.baytech.net/products/showprod.php?prod=DS2-RPC 
(DS2-RPC) that offers both power switching and serial control in a 1U 
form factor. It's a must have for remote systems.


-Max

beonice wrote:

Hello, all.

I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.) 


I've been looking at the fancy xeon-based systems
listed on ebay for a couple of hundred dollars, in the
hope that some of them have remote reboot
capabilities, but most of the sellers don't mention
this ability, and by the time I send out email, the
item is already taken anyway. :)

So, to cut the long story short, has anyone used one
of these server-class machines with remote reboot
capability, and does it really help? Are there any
particular configurations to stay away from? 


The wiki doesn't talk specifically about issues
regarding dual-CPU machines, but in following the chat
here on asterisk-users, it seems there are definitely
issues there ... can anyone elaborate? I don't want to
spend money on a fancy system that turns out to be
useless for my purposes.

Thanks for any insight!

Cheers,
Maya




 
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Tzafrir Cohen]

Hi

On Thu, Jun 23, 2005 at 05:52:44PM -0700, beonice wrote:
 
 Okay, so what makes more sense:
   1) a remote management card that will let me
 actually log in to the machine to monitor it as well
 as to reboot it
 vs.
   2) a remote-accessible powerstrip that will allow me
 to remotely reboot the server?

Linux also has a software watchdog module. Maybe it could work here
without extra hardware to set up.

 
 I'm realising that sometimes the problem may simply be
 processes out of control, and may be something that
 doesn't require killing the entire machine, but just
 some processes. 
 
 In my current setup (an ordinary PIII 1.someting GHz
 machine, not a server-class machine), when a process
 goes haywire, I lose remote access via SSH, so I drive
 to the colo, log in, sigh in frustration, and reboot
 because I'm already here, so why not?.

Because you destroy any evidence of the problem.

What processes are taking much CPU time? Are there any relevant log
messages? Is this a case of over-swapping? (not 100% CPU usage, but
rather large swap usage, CPU spends too much time at system, though 
the latter may be probably normal for an Asterisk server).

Could you login from the console? Did you manage to move between virtual
consoles?

Install the package sysstat and run sar to get some stats. Consider
adding a cron job to gather more relevant stats every 5 minutes or every
minute.

BTW: does asterisk run with real-time priority? try removing it, so at
least asterisk won't hang the whole system. Though I doubt it if this
would help.

 
 Some of the problems were caused by my old router ...
 since I replaced it, the need to drive the 40 miles
 each way has gone down significantly ... in fact, to
 pretty much zero. So I have time to contemplate my
 options here. :)

You have an extra router there? I recall that there was a kernel patch
to reboot the system upon recieving a specific ICMP packet.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Rich Adamson]

 I'm tired of having to drive out to the colocation
 facility each time my dedicated asterisk server craps
 out, just to press the button to do a hard reboot.
 (I'm running 1.05 stable at present, no telephony
 hardware, as this is mainly a system that receives
 calls, no dial-out ability is needed.) 

Then fix the root-cause. Rebooting a box is not a fix. There
are plenty of uptime examples in the months/years timeframes.


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RE: [Asterisk-Users] Asterisk server with remote monitoringcapabilities

[Matt Schulte]

I would have to agree, an IAD locking up is bad either way you look at
it. Even if you're there to reboot it on demand, it takes nearly 5
minutes to come back up. What kind of servers are they? What kind of
phones? In all honesty, none of our IAD's ever lock up. And ones that
did were defective and replaced.

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: Friday, June 24, 2005 8:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk server with remote
monitoringcapabilities


 I'm tired of having to drive out to the colocation
 facility each time my dedicated asterisk server craps
 out, just to press the button to do a hard reboot.
 (I'm running 1.05 stable at present, no telephony
 hardware, as this is mainly a system that receives
 calls, no dial-out ability is needed.)

Then fix the root-cause. Rebooting a box is not a fix. There are plenty
of uptime examples in the months/years timeframes.


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Re: [Asterisk-Users] Asterisk server with remote monitoringcapabilities

[Chris Mason (Lists)]

I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.)
   

I have never had a pbx lock up. I suggest you change your hardware. I 
would think your problem is RAM, as Asterisk is not hard on the drives 
or other hardware.


--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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RE: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Max W Blackmer Jr]

  Original Message 
 Subject: Re: [Asterisk-Users] Asterisk server with remote monitoring
 capabilities
 From: beonice [EMAIL PROTECTED]
 Date: Thu, June 23, 2005 7:52 pm

 --- Michael Welter [EMAIL PROTECTED] wrote:

  William Boehlke wrote:
   Dell sells a remote management card for under $400
  that enables remote
   reboots. I know there are others out there but
  have no experience with them.
  
  
   William Boehlke
   Signate
  
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED]
  On Behalf Of beonice
  
   I'm tired of having to drive out to the colocation
  facility each time my
   dedicated asterisk server craps out, just to press
  the button to do a hard
   reboot.
   (I'm running 1.05 stable at present, no telephony
  hardware, as this is
   mainly a system that receives calls, no dial-out
  ability is needed.)
  
  APC makes a power strip with a web server.  Each
  socket has its own IP
  address.  All you have to do to power cycle is
  access the IP address
  from your web browser and give the power cycle
  command.  It is sooo cool.

 Thanks for your responses, folks.

 Okay, so what makes more sense:
   1) a remote management card that will let me
 actually log in to the machine to monitor it as well
 as to reboot it
 vs.
   2) a remote-accessible powerstrip that will allow me
 to remotely reboot the server?


A little note. make sure your server motherboard/bios supports power on
after power loss to use the remote control power strip. Secondly make
sure the power strip control uses SSH and NOT telnet to control it.
Telnet is too insecure because passwords are sent plain text.

Another possibility is to write a reboot script and set up a cron job to
automatically reboot every night until you solve the bigger problem of
why is the server having problems?

With Linux their is little need to reboot Linux. There is only one time
that you have to reboot Linux. When you upgrade the kernel or its
modules. Kernel modules do not always need a reboot. Kernel module that
do require a reboot are critical to operation of your system for example
RAID# .

The best way is to have a script that uses the init script to restart
the applications that are questionable on a cron job schedule for low
usage.  With a good script you could also check on the status of the
service and perform functional test of the service. Then the script
would perform the necessary tasks to recover from application failure. 
This wont help with a total system failure as the script will not work.
Some of the remote monitoring cards can detect a system lockup and
preform a system reboot automatically.  When all of these fail you can
use remote control power strips or a KVM (Keyboard Video Mouse) over IP
to remotely control the hardware as if you are there.  Cyclades
(www.cyclades.com) sells both KVM and Remote Power management solutions
that are secure. They even have RSA authentication tokens and a
Biometric/RSA token authentications for secure management of the remote
locations.


Cheers,

Max W. Blackmer, Jr.
Consultant, Knowledge Power IT

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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Mark Musone]

Warning - Shameless plug..

Why not just use a managed service provider (like www.shatterit.com)
that is _really_ there 24/7 and can not only reboot your box for you
at any time, but can also monitor it so that it doesnt go down in the
first place.

I apologize for the commerical nature, but this is a real solution for
this real problem...all those expensive hardware solutions is no
replacement for a human..

-Mark


On 6/24/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
   Original Message 
  Subject: Re: [Asterisk-Users] Asterisk server with remote monitoring
  capabilities
  From: beonice [EMAIL PROTECTED]
  Date: Thu, June 23, 2005 7:52 pm
 
  --- Michael Welter [EMAIL PROTECTED] wrote:
 
   William Boehlke wrote:
Dell sells a remote management card for under $400
   that enables remote
reboots. I know there are others out there but
   have no experience with them.
   
   
William Boehlke
Signate
   
   
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
   On Behalf Of beonice
   
I'm tired of having to drive out to the colocation
   facility each time my
dedicated asterisk server craps out, just to press
   the button to do a hard
reboot.
(I'm running 1.05 stable at present, no telephony
   hardware, as this is
mainly a system that receives calls, no dial-out
   ability is needed.)
   
   APC makes a power strip with a web server.  Each
   socket has its own IP
   address.  All you have to do to power cycle is
   access the IP address
   from your web browser and give the power cycle
   command.  It is sooo cool.
 
  Thanks for your responses, folks.
 
  Okay, so what makes more sense:
1) a remote management card that will let me
  actually log in to the machine to monitor it as well
  as to reboot it
  vs.
2) a remote-accessible powerstrip that will allow me
  to remotely reboot the server?
 
 
 A little note. make sure your server motherboard/bios supports power on
 after power loss to use the remote control power strip. Secondly make
 sure the power strip control uses SSH and NOT telnet to control it.
 Telnet is too insecure because passwords are sent plain text.
 
 Another possibility is to write a reboot script and set up a cron job to
 automatically reboot every night until you solve the bigger problem of
 why is the server having problems?
 
 With Linux their is little need to reboot Linux. There is only one time
 that you have to reboot Linux. When you upgrade the kernel or its
 modules. Kernel modules do not always need a reboot. Kernel module that
 do require a reboot are critical to operation of your system for example
 RAID# .
 
 The best way is to have a script that uses the init script to restart
 the applications that are questionable on a cron job schedule for low
 usage.  With a good script you could also check on the status of the
 service and perform functional test of the service. Then the script
 would perform the necessary tasks to recover from application failure.
 This wont help with a total system failure as the script will not work.
 Some of the remote monitoring cards can detect a system lockup and
 preform a system reboot automatically.  When all of these fail you can
 use remote control power strips or a KVM (Keyboard Video Mouse) over IP
 to remotely control the hardware as if you are there.  Cyclades
 (www.cyclades.com) sells both KVM and Remote Power management solutions
 that are secure. They even have RSA authentication tokens and a
 Biometric/RSA token authentications for secure management of the remote
 locations.
 
 
 Cheers,
 
 Max W. Blackmer, Jr.
 Consultant, Knowledge Power IT
 
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RE: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[William Boehlke]

Dell sells a remote management card for under $400 that enables remote
reboots. I know there are others out there but have no experience with them.

 
William Boehlke
Signate


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of beonice
Sent: Thursday, June 23, 2005 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk server with remote monitoring
capabilities

Hello, all.

I'm tired of having to drive out to the colocation facility each time my
dedicated asterisk server craps out, just to press the button to do a hard
reboot.
(I'm running 1.05 stable at present, no telephony hardware, as this is
mainly a system that receives calls, no dial-out ability is needed.) 

I've been looking at the fancy xeon-based systems listed on ebay for a
couple of hundred dollars, in the hope that some of them have remote reboot
capabilities, but most of the sellers don't mention this ability, and by the
time I send out email, the item is already taken anyway. :)

So, to cut the long story short, has anyone used one of these server-class
machines with remote reboot capability, and does it really help? Are there
any particular configurations to stay away from? 

The wiki doesn't talk specifically about issues regarding dual-CPU machines,
but in following the chat here on asterisk-users, it seems there are
definitely issues there ... can anyone elaborate? I don't want to spend
money on a fancy system that turns out to be useless for my purposes.

Thanks for any insight!

Cheers,
Maya





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RE: [Asterisk-Users] Asterisk server with remote monitoring capab ilities

[mattf]

We have two Baytech RPC3 remote power switches(8 outlets each), they are
great, you can telnet into them and reset ports as needed. I even setup one
of them to be controlled by an AGI script on our Asterisk servers to cycle
power over the phone. Saved countless hours of driving. APC makes them too
although they are more expensive.

MATT---

-Original Message-
From: beonice [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 23, 2005 7:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk server with remote monitoring
capabilities


Hello, all.

I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability is needed.) 

I've been looking at the fancy xeon-based systems
listed on ebay for a couple of hundred dollars, in the
hope that some of them have remote reboot
capabilities, but most of the sellers don't mention
this ability, and by the time I send out email, the
item is already taken anyway. :)

So, to cut the long story short, has anyone used one
of these server-class machines with remote reboot
capability, and does it really help? Are there any
particular configurations to stay away from? 

The wiki doesn't talk specifically about issues
regarding dual-CPU machines, but in following the chat
here on asterisk-users, it seems there are definitely
issues there ... can anyone elaborate? I don't want to
spend money on a fancy system that turns out to be
useless for my purposes.

Thanks for any insight!

Cheers,
Maya




 
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Rekindle the Rivalries. Sign up for Fantasy Football 
http://football.fantasysports.yahoo.com
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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Michael Welter]

William Boehlke wrote:

Dell sells a remote management card for under $400 that enables remote
reboots. I know there are others out there but have no experience with them.

 
William Boehlke

Signate


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of beonice
Sent: Thursday, June 23, 2005 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk server with remote monitoring
capabilities

Hello, all.

I'm tired of having to drive out to the colocation facility each time my
dedicated asterisk server craps out, just to press the button to do a hard
reboot.
(I'm running 1.05 stable at present, no telephony hardware, as this is
mainly a system that receives calls, no dial-out ability is needed.) 

APC makes a power strip with a web server.  Each socket has its own IP 
address.  All you have to do to power cycle is access the IP address 
from your web browser and give the power cycle command.  It is sooo cool.



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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[Joseph]

If you want very secure way of login-in and rebooting - install Freesco
firewall on any 486-machine with floppy drive and two network cards.
Install port knocking on a Freesco firewall floppy (if you want absolute
security)
enable ssh and you are set.  
Port knocking will enable you to open the ssh port (only the machine
that issue successful knock will be able to log-in via ssh, so I
consider it secure.
ssh to root and reboot the machine from console.

Maybe you don't need to reboot just restart the asterisk, any how you
can check the status when you log-in as root and do what is needed.

-- 
#Joseph

[snip]
 So, to cut the long story short, has anyone used one
 of these server-class machines with remote reboot
 capability, and does it really help? Are there any
 particular configurations to stay away from? 
 
 The wiki doesn't talk specifically about issues
 regarding dual-CPU machines, but in following the chat
 here on asterisk-users, it seems there are definitely
 issues there ... can anyone elaborate? I don't want to
 spend money on a fancy system that turns out to be
 useless for my purposes.
 
 Thanks for any insight!
 
 Cheers,
 Maya

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Re: [Asterisk-Users] Asterisk server with remote monitoring capabilities

[beonice]

--- Michael Welter [EMAIL PROTECTED] wrote:

 William Boehlke wrote:
  Dell sells a remote management card for under $400
 that enables remote
  reboots. I know there are others out there but
 have no experience with them.
  
   
  William Boehlke
  Signate
  
  
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]
 On Behalf Of beonice
  
  I'm tired of having to drive out to the colocation
 facility each time my
  dedicated asterisk server craps out, just to press
 the button to do a hard
  reboot.
  (I'm running 1.05 stable at present, no telephony
 hardware, as this is
  mainly a system that receives calls, no dial-out
 ability is needed.) 
  
 APC makes a power strip with a web server.  Each
 socket has its own IP 
 address.  All you have to do to power cycle is
 access the IP address 
 from your web browser and give the power cycle
 command.  It is sooo cool.

Thanks for your responses, folks.

Okay, so what makes more sense:
  1) a remote management card that will let me
actually log in to the machine to monitor it as well
as to reboot it
vs.
  2) a remote-accessible powerstrip that will allow me
to remotely reboot the server?

I'm realising that sometimes the problem may simply be
processes out of control, and may be something that
doesn't require killing the entire machine, but just
some processes. 

In my current setup (an ordinary PIII 1.someting GHz
machine, not a server-class machine), when a process
goes haywire, I lose remote access via SSH, so I drive
to the colo, log in, sigh in frustration, and reboot
because I'm already here, so why not?.

Some of the problems were caused by my old router ...
since I replaced it, the need to drive the 40 miles
each way has gone down significantly ... in fact, to
pretty much zero. So I have time to contemplate my
options here. :)

Cheers,
Maya





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