Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[John Runyon]

David, can you play around with the routing table and get the OS to handle
it for you? So long as asterisk isn’t calling bind() (or is calling with
0.0.0.0) I would imagine adding a route for the peer, with your normal
gateway, and the correct device would work.

On Thu, Oct 29, 2020 at 10:04 PM David Cunningham 
wrote:

> Hi Dovid,
>
> We can change the SDP in Kamailio, but Asterisk will still send its RTP
> from its default address. The remote end is strict about accepting RTP from
> the specified source and won't accept it. Have you any suggestions to solve
> that problem?
>
> Thank you.
>
>
> On Fri, 30 Oct 2020 at 14:49, Dovid Bender  wrote:
>
>> Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
>> it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
>>
>> On Thu, Oct 29, 2020 at 20:44 David Cunningham 
>> wrote:
>>
>>> Hello,
>>>
>>> Does anyone know a way with chan_sip to tell Asterisk to use a specific
>>> IP address for its end of the communication for a specific device?
>>> Something like:
>>>
>>> [device]
>>> type = friend
>>> host = 11.22.11.22
>>> ouraddress = 33.44.33.44
>>>
>>> This is for use on a server with multiple IP addresses. There is the
>>> "extenip" setting, but it's really designed for NAT, and can only appear in
>>> the [general] section.
>>>
>>> Any suggestions would be greatly appreciated.
>>>
>>>
>>> On Sat, 24 Oct 2020 at 09:43, David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
 OK, thank you George.


 On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:

>
>
> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi George,
>>
>> Thank you for the response. I'm a little unclear on what you mean by
>> a transport. We're using chan_sip, not pjsip.
>>
>> Do you mean a device in sip.conf, using bindaddr to set the address
>> to bind for that device? We've only used bindaddr in the [general] 
>> section
>> before, but if it will work in a device that could be the answer.
>>
>
> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it
> for chan_sip.
>
>
>
>>
>>
>> On Fri, 23 Oct 2020 at 00:13, George Joseph 
>> wrote:
>>
>>>
>>>
>>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
 Hello,

 We have an Asterisk server with two public IP addresses, let's say
 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged 
 with
 a call dialled from Asterisk to an external destination. The external
 destination sees the SIP packet as coming from 1.1.1.1 and the media
 address in the SDP is 1.1.1.1, which is great.

 However if we receive a call in to 2.2.2.2 then the call dialled
 from Asterisk to an external destination still comes from 1.1.1.1, 
 whereas
 we want it to come from 2.2.2.2. The source of any dialled call (the IP
 packet and the SDP media address) should be the same as the address the
 related inbound call was received to.

 For example:
 INVITE received to 1.1.1.1:5060 -> Asterisk dials
 destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
 termination.com
 INVITE received to 2.2.2.2:5060 -> Asterisk dials
 destinat...@pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com

 Does anyone know how this can be achieved?

>>>
>>> If termination.com is only on 1.1.1.1 and pstn.com is only on
>>> 2.2.2.2, create 2 transports, one specifically bound to 1.1.1.1,
>>> transport-1.1.1.1 for instance, and another to 2.2.2.2:
>>> transport-2.2.2.2.  The names aren't important as long as you can tell 
>>> the
>>> difference.  Then explicitly configure endpoint termination.com's
>>> "transport" parameter to "transport-1.1.1.1" and pstn.com's
>>> "transport" parameter to "transport-2.2.2.2".   In your dialplan, you 
>>> can
>>> see which endpoint the call came in on, and route it out the same 
>>> endpoint.
>>>
>>> If both providers are available from both interfaces, you can create
>>> 2 endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
>>> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with 
>>> the
>>> same transports as above.
>>>
>>>
>>>
>>>
>>>

 Thanks in advance for your help,

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

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

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[Jeff LaCoursiere]

I didn't want to post this because its kind of ugly, but we *did* 
actually do it a number of years ago to get around this issue with chan_sip.


Our original architecture was based on LXC, and we had large servers 
running hundreds of containers, each running asterisk.  The "host" ran 
asterisk too, as the gateway for all the container instances.


We once used two of those containers to run asterisk on specific host 
interfaces (one instance bridged to one nic, the other to the other).  
The host asterisk would route calls out one container or the other, with 
the effect you are looking for...


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.


On 10/29/20 9:05 PM, David Cunningham wrote:

Hi Dovid,

We can change the SDP in Kamailio, but Asterisk will still send its 
RTP from its default address. The remote end is strict about accepting 
RTP from the specified source and won't accept it. Have you any 
suggestions to solve that problem?


Thank you.


On Fri, 30 Oct 2020 at 14:49, Dovid Bender > wrote:


Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1
you pass it along as is. Where you want 2.2.2.2 change the sdp in
opensips/kamailio

On Thu, Oct 29, 2020 at 20:44 David Cunningham
mailto:dcunning...@voisonics.com>> wrote:

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a
specific IP address for its end of the communication for a
specific device? Something like:

[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There
is the "extenip" setting, but it's really designed for NAT,
and can only appear in the [general] section.

Any suggestions would be greatly appreciated.



-- 
_
-- 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] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Hi Dovid,

We can change the SDP in Kamailio, but Asterisk will still send its RTP
from its default address. The remote end is strict about accepting RTP from
the specified source and won't accept it. Have you any suggestions to solve
that problem?

Thank you.


On Fri, 30 Oct 2020 at 14:49, Dovid Bender  wrote:

> Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
> it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
>
> On Thu, Oct 29, 2020 at 20:44 David Cunningham 
> wrote:
>
>> Hello,
>>
>> Does anyone know a way with chan_sip to tell Asterisk to use a specific
>> IP address for its end of the communication for a specific device?
>> Something like:
>>
>> [device]
>> type = friend
>> host = 11.22.11.22
>> ouraddress = 33.44.33.44
>>
>> This is for use on a server with multiple IP addresses. There is the
>> "extenip" setting, but it's really designed for NAT, and can only appear in
>> the [general] section.
>>
>> Any suggestions would be greatly appreciated.
>>
>>
>> On Sat, 24 Oct 2020 at 09:43, David Cunningham 
>> wrote:
>>
>>> OK, thank you George.
>>>
>>>
>>> On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:
>>>


 On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
 dcunning...@voisonics.com> wrote:

> Hi George,
>
> Thank you for the response. I'm a little unclear on what you mean by a
> transport. We're using chan_sip, not pjsip.
>
> Do you mean a device in sip.conf, using bindaddr to set the address to
> bind for that device? We've only used bindaddr in the [general] section
> before, but if it will work in a device that could be the answer.
>

 Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it
 for chan_sip.



>
>
> On Fri, 23 Oct 2020 at 00:13, George Joseph 
> wrote:
>
>>
>>
>> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hello,
>>>
>>> We have an Asterisk server with two public IP addresses, let's say
>>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged 
>>> with
>>> a call dialled from Asterisk to an external destination. The external
>>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>>> address in the SDP is 1.1.1.1, which is great.
>>>
>>> However if we receive a call in to 2.2.2.2 then the call dialled
>>> from Asterisk to an external destination still comes from 1.1.1.1, 
>>> whereas
>>> we want it to come from 2.2.2.2. The source of any dialled call (the IP
>>> packet and the SDP media address) should be the same as the address the
>>> related inbound call was received to.
>>>
>>> For example:
>>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>>> termination.com
>>> INVITE received to 2.2.2.2:5060 -> Asterisk dials
>>> destinat...@pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>>
>>> Does anyone know how this can be achieved?
>>>
>>
>> If termination.com is only on 1.1.1.1 and pstn.com is only on
>> 2.2.2.2, create 2 transports, one specifically bound to 1.1.1.1,
>> transport-1.1.1.1 for instance, and another to 2.2.2.2:
>> transport-2.2.2.2.  The names aren't important as long as you can tell 
>> the
>> difference.  Then explicitly configure endpoint termination.com's
>> "transport" parameter to "transport-1.1.1.1" and pstn.com's
>> "transport" parameter to "transport-2.2.2.2".   In your dialplan, you can
>> see which endpoint the call came in on, and route it out the same 
>> endpoint.
>>
>> If both providers are available from both interfaces, you can create
>> 2 endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
>> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with 
>> the
>> same transports as above.
>>
>>
>>
>>
>>
>>>
>>> Thanks in advance for your help,
>>>
>>> --
>>> David Cunningham, Voisonics Limited
>>> http://voisonics.com/
>>> USA: +1 213 221 1092
>>> New Zealand: +64 (0)28 2558 3782
>>> --
>>> _
>>> -- 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
>>
>>
>>
>> --
>> George Joseph
>> Asterisk Software Developer
>> direct/fax +1 256 428 6012
>> 

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[Dovid Bender]

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio

On Thu, Oct 29, 2020 at 20:44 David Cunningham 
wrote:

> Hello,
>
> Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
> address for its end of the communication for a specific device? Something
> like:
>
> [device]
> type = friend
> host = 11.22.11.22
> ouraddress = 33.44.33.44
>
> This is for use on a server with multiple IP addresses. There is the
> "extenip" setting, but it's really designed for NAT, and can only appear in
> the [general] section.
>
> Any suggestions would be greatly appreciated.
>
>
> On Sat, 24 Oct 2020 at 09:43, David Cunningham 
> wrote:
>
>> OK, thank you George.
>>
>>
>> On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:
>>
>>>
>>>
>>> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
>>> dcunning...@voisonics.com> wrote:
>>>
 Hi George,

 Thank you for the response. I'm a little unclear on what you mean by a
 transport. We're using chan_sip, not pjsip.

 Do you mean a device in sip.conf, using bindaddr to set the address to
 bind for that device? We've only used bindaddr in the [general] section
 before, but if it will work in a device that could be the answer.

>>>
>>> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it
>>> for chan_sip.
>>>
>>>
>>>


 On Fri, 23 Oct 2020 at 00:13, George Joseph  wrote:

>
>
> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hello,
>>
>> We have an Asterisk server with two public IP addresses, let's say
>> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged 
>> with
>> a call dialled from Asterisk to an external destination. The external
>> destination sees the SIP packet as coming from 1.1.1.1 and the media
>> address in the SDP is 1.1.1.1, which is great.
>>
>> However if we receive a call in to 2.2.2.2 then the call dialled from
>> Asterisk to an external destination still comes from 1.1.1.1, whereas we
>> want it to come from 2.2.2.2. The source of any dialled call (the IP 
>> packet
>> and the SDP media address) should be the same as the address the related
>> inbound call was received to.
>>
>> For example:
>> INVITE received to 1.1.1.1:5060 -> Asterisk dials
>> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
>> termination.com
>> INVITE received to 2.2.2.2:5060 -> Asterisk dials
>> destinat...@pstn.com -> INVITE sent from 2.2.2.2:5060 to pstn.com
>>
>> Does anyone know how this can be achieved?
>>
>
> If termination.com is only on 1.1.1.1 and pstn.com is only on
> 2.2.2.2, create 2 transports, one specifically bound to 1.1.1.1,
> transport-1.1.1.1 for instance, and another to 2.2.2.2:
> transport-2.2.2.2.  The names aren't important as long as you can tell the
> difference.  Then explicitly configure endpoint termination.com's
> "transport" parameter to "transport-1.1.1.1" and pstn.com's
> "transport" parameter to "transport-2.2.2.2".   In your dialplan, you can
> see which endpoint the call came in on, and route it out the same 
> endpoint.
>
> If both providers are available from both interfaces, you can create 2
> endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
> termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with 
> the
> same transports as above.
>
>
>
>
>
>>
>> Thanks in advance for your help,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>> --
>> _
>> -- 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
>
>
>
> --
> George Joseph
> Asterisk Software Developer
> direct/fax +1 256 428 6012
> Check us out at www.sangoma.com and www.asterisk.org
> [image: image.png]
> --
> _
> -- 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 

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[David Cunningham]

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
address for its end of the communication for a specific device? Something
like:

[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There is the
"extenip" setting, but it's really designed for NAT, and can only appear in
the [general] section.

Any suggestions would be greatly appreciated.


On Sat, 24 Oct 2020 at 09:43, David Cunningham 
wrote:

> OK, thank you George.
>
>
> On Sat, 24 Oct 2020 at 03:16, George Joseph  wrote:
>
>>
>>
>> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hi George,
>>>
>>> Thank you for the response. I'm a little unclear on what you mean by a
>>> transport. We're using chan_sip, not pjsip.
>>>
>>> Do you mean a device in sip.conf, using bindaddr to set the address to
>>> bind for that device? We've only used bindaddr in the [general] section
>>> before, but if it will work in a device that could be the answer.
>>>
>>
>> Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it
>> for chan_sip.
>>
>>
>>
>>>
>>>
>>> On Fri, 23 Oct 2020 at 00:13, George Joseph  wrote:
>>>


 On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
 dcunning...@voisonics.com> wrote:

> Hello,
>
> We have an Asterisk server with two public IP addresses, let's say
> 1.1.1.1 and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged 
> with
> a call dialled from Asterisk to an external destination. The external
> destination sees the SIP packet as coming from 1.1.1.1 and the media
> address in the SDP is 1.1.1.1, which is great.
>
> However if we receive a call in to 2.2.2.2 then the call dialled from
> Asterisk to an external destination still comes from 1.1.1.1, whereas we
> want it to come from 2.2.2.2. The source of any dialled call (the IP 
> packet
> and the SDP media address) should be the same as the address the related
> inbound call was received to.
>
> For example:
> INVITE received to 1.1.1.1:5060 -> Asterisk dials
> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
> termination.com
> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com
> -> INVITE sent from 2.2.2.2:5060 to pstn.com
>
> Does anyone know how this can be achieved?
>

 If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
 create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
 for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names
 aren't important as long as you can tell the difference.  Then explicitly
 configure endpoint termination.com's "transport" parameter to
 "transport-1.1.1.1" and pstn.com's "transport" parameter to
 "transport-2.2.2.2".   In your dialplan, you can see which endpoint the
 call came in on, and route it out the same endpoint.

 If both providers are available from both interfaces, you can create 2
 endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
 termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
 same transports as above.





>
> Thanks in advance for your help,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> --
> _
> -- 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



 --
 George Joseph
 Asterisk Software Developer
 direct/fax +1 256 428 6012
 Check us out at www.sangoma.com and www.asterisk.org
 [image: image.png]
 --
 _
 -- 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
>>>
>>>
>>>
>>> --
>>> David Cunningham, Voisonics Limited
>>> http://voisonics.com/
>>> USA: +1 213 221 1092
>>> New Zealand: +64 (0)28 2558 3782
>>> --
>>> _
>>> -- Bandwidth and 

Re: [asterisk-users] Expert to work on load issue

[Jöran Vinzens]

Hi Dovid,

unfortunately I do not. We have seen similar issues when using Perl AGI
Scripts when using Asterisk16/17. But since we moved everything to Java
FastAGI we haven't seen any load issues anymore.
Are you doing anything from fastagi that takes a lot of time? Something
like a Playback? We Don't. We tried to do everything that takes some time
in the Dialplan.

Maybe you could try to kill the Asterisk with "kill -9" to see in the
backtrace what the Asterisk is doing when it has such a high load.

In the Backtrace you should be able to have a look at all processes and
what they are doing. I guess i do not have to say "DON'T do it on the
production system" :)

BR
Jöran

On Thu, Oct 29, 2020 at 7:06 PM Dovid Bender  wrote:

> Anyone have any other ideas?
>
>
> On Tue, Oct 27, 2020 at 1:27 PM Dovid Bender  wrote:
>
>> Jon,
>>
>> We are only using FastAgi. On the second system (running Asterisk 16)
>> there are no agi's running (just some bash scripts on call hangup). I did
>> add some hackey code (netstat -nua | grep -v 'udp0  0' | grep
>> -v udp6 | grep -v ' 0 0.0.0.0' | grep udp)  to my bash script to check out
>> the packet queue (with the help of
>> https://serverfault.com/questions/623377/is-there-a-way-to-get-current-txqueue-utilization-on-e-g-eth0-or-tun-device)
>> and it  *seems there may be a corelation between packets in the queue and
>> the load on Asterisk
>>
>> [root@a30 ~]# /tmp/x.sh
>> CPU USAGE 393.8 Active calls 317
>> CPU USAGE 117.6 Active calls 318
>> CPU USAGE 143.8 Active calls 318
>> CPU USAGE 305.9 Active calls 318
>> udp0768 0.0.0.0:14510   0.0.0.0:*
>>
>> CPU USAGE 123.5 Active calls 316
>> CPU USAGE 137.5 Active calls 314
>> CPU USAGE 329.4 Active calls 314
>> CPU USAGE 312.5 Active calls 317
>> CPU USAGE 125.0 Active calls 317
>> CPU USAGE 137.5 Active calls 316
>> CPU USAGE 123.5 Active calls 316
>> CPU USAGE 117.6 Active calls 315
>> CPU USAGE 117.6 Active calls 314
>> CPU USAGE 135.3 Active calls 312
>> CPU USAGE 131.2 Active calls 309
>> CPU USAGE 131.2 Active calls 308
>> CPU USAGE 118.8 Active calls 309
>> CPU USAGE 331.2 Active calls 310
>> CPU USAGE 111.8 Active calls 309
>> udp0   1280 0.0.0.0:15598   0.0.0.0:*
>>
>> CPU USAGE 517.6 Active calls 309
>> CPU USAGE 137.5 Active calls 308
>> CPU USAGE 143.8 Active calls 308
>> CPU USAGE 137.5 Active calls 307
>> CPU USAGE 311.8 Active calls 308
>> CPU USAGE 137.5 Active calls 308
>> CPU USAGE 125.0 Active calls 307
>> CPU USAGE 156.2 Active calls 307
>> udp0   1280 0.0.0.0:19148   0.0.0.0:*
>>
>> CPU USAGE 135.3 Active calls 305
>> CPU USAGE 305.9 Active calls 307
>> CPU USAGE 156.2 Active calls 309
>> CPU USAGE 450.0 Active calls 311
>> CPU USAGE 137.5 Active calls 310
>> CPU USAGE 137.5 Active calls 310
>> CPU USAGE 256.2 Active calls 311
>> CPU USAGE 150.0 Active calls 308
>> CPU USAGE 281.2 Active calls 308
>> CPU USAGE 281.2 Active calls 307
>> CPU USAGE 137.5 Active calls 304
>> udp0   1280 0.0.0.0:11284   0.0.0.0:*
>>
>> CPU USAGE 135.3 Active calls 306
>> CPU USAGE 150.0 Active calls 309
>> CPU USAGE 331.2 Active calls 310
>> CPU USAGE 125.0 Active calls 308
>> CPU USAGE 117.6 Active calls 307
>> CPU USAGE 135.3 Active calls 306
>> udp0   1280 0.0.0.0:18494   0.0.0.0:*
>>
>> CPU USAGE 137.5 Active calls 307
>> CPU USAGE 141.2 Active calls 306
>> CPU USAGE 137.5 Active calls 307
>> CPU USAGE 562.5 Active calls 308
>> CPU USAGE 862.5 Active calls 311
>> CPU USAGE 164.7 Active calls 310
>> udp0768 0.0.0.0:17890   0.0.0.0:*
>>
>> udp0   1280 0.0.0.0:11504   0.0.0.0:*
>>
>> [root@a30 ~]#
>>
>>
>>
>> On Tue, Oct 27, 2020 at 1:21 PM Jon Bonilla (Manwe) 
>> wrote:
>>
>>> El Tue, 27 Oct 2020 12:52:47 -0400
>>> Dovid Bender  escribió:
>>>
>>> > Hi,
>>> >
>>> > Sorry in advance that I am emailing the users list and not the biz
>>> list I
>>> > think I will find my target audience here. We are looking to hire a
>>> > consultant to help us figure out an issue. We are having what seems are
>>> > "random load" issues with bare metal boxes that are dedicated to
>>> Asterisk
>>> > and a few Perl AGI's. We went after all the usual suspects (CPU IDLE,
>>> > memory usage etc.). I wrote the simple bash script below to show me
>>> total
>>> > calls and CPU usage of Asterisk
>>> > #! /bin/bash
>>> >
>>> >
>>>
>>> Did you try FastAGI? Asterisk doesn't handle spawning many processes
>>> very well.
>>> It's a bottleneck.
>>>
>>>
>>>
>>>
>>> --
>>> PekePBX, the multitenant PBX solution
>>> https://pekepbx.com
>>>
>> --
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Re: [asterisk-users] astdbdir in asterisk.conf has no effect

[Jeff Pyle]

Ha!  I had not.  Thank you.


- Jeff



On Thu, Oct 29, 2020 at 2:10 PM Joshua C. Colp  wrote:

> On Thu, Oct 29, 2020 at 3:05 PM Jeff Pyle  wrote:
>
>> Hello,
>>
>> On Asterisk 13.19 I'm trying to set astdbdir in /etc/asterisk.conf, but
>> it's not having any effect.  For example:
>>
>> # grep astdbdir /etc/asterisk/asterisk.conf
>> astdbdir => /tmp/asterisk
>>
>> /tmp/asterisk exists and is owned by asterisk:asterisk, as the asterisk
>> processing is running as asterisk:asterisk with the config file forced:
>>
>> # ps aux | grep asterisk
>> asterisk  3389 13.1  3.5 4295146504 214004 ?   S> /usr/sbin/asterisk -mqC /etc/asterisk/asterisk.conf
>>
>> Yet it's still running in the old place:
>>
>> # lsof -n | grep astdb.sqlite | awk '{print $11}' | sort | uniq
>> /var/lib/asterisk/astdb.sqlite3
>>
>> I see default.h defines the path with the astdb:
>>   #define DEFAULT_DB "/var/lib/asterisk/astdb"
>> so I tried astdbdir => /tmp/asterisk/astdb, but no change.  It's still
>> running in /var/lib/asterisk.
>>
>> I feel like I'm missing something.  Any suggestions?
>>
>
> Have you removed the (!) beside [ general] in asterisk.conf?
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
>
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Re: [asterisk-users] astdbdir in asterisk.conf has no effect

[Joshua C. Colp]

On Thu, Oct 29, 2020 at 3:05 PM Jeff Pyle  wrote:

> Hello,
>
> On Asterisk 13.19 I'm trying to set astdbdir in /etc/asterisk.conf, but
> it's not having any effect.  For example:
>
> # grep astdbdir /etc/asterisk/asterisk.conf
> astdbdir => /tmp/asterisk
>
> /tmp/asterisk exists and is owned by asterisk:asterisk, as the asterisk
> processing is running as asterisk:asterisk with the config file forced:
>
> # ps aux | grep asterisk
> asterisk  3389 13.1  3.5 4295146504 214004 ?   S /usr/sbin/asterisk -mqC /etc/asterisk/asterisk.conf
>
> Yet it's still running in the old place:
>
> # lsof -n | grep astdb.sqlite | awk '{print $11}' | sort | uniq
> /var/lib/asterisk/astdb.sqlite3
>
> I see default.h defines the path with the astdb:
>   #define DEFAULT_DB "/var/lib/asterisk/astdb"
> so I tried astdbdir => /tmp/asterisk/astdb, but no change.  It's still
> running in /var/lib/asterisk.
>
> I feel like I'm missing something.  Any suggestions?
>

Have you removed the (!) beside [ general] in asterisk.conf?

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Expert to work on load issue

[Dovid Bender]

Anyone have any other ideas?


On Tue, Oct 27, 2020 at 1:27 PM Dovid Bender  wrote:

> Jon,
>
> We are only using FastAgi. On the second system (running Asterisk 16)
> there are no agi's running (just some bash scripts on call hangup). I did
> add some hackey code (netstat -nua | grep -v 'udp0  0' | grep
> -v udp6 | grep -v ' 0 0.0.0.0' | grep udp)  to my bash script to check out
> the packet queue (with the help of
> https://serverfault.com/questions/623377/is-there-a-way-to-get-current-txqueue-utilization-on-e-g-eth0-or-tun-device)
> and it  *seems there may be a corelation between packets in the queue and
> the load on Asterisk
>
> [root@a30 ~]# /tmp/x.sh
> CPU USAGE 393.8 Active calls 317
> CPU USAGE 117.6 Active calls 318
> CPU USAGE 143.8 Active calls 318
> CPU USAGE 305.9 Active calls 318
> udp0768 0.0.0.0:14510   0.0.0.0:*
>
> CPU USAGE 123.5 Active calls 316
> CPU USAGE 137.5 Active calls 314
> CPU USAGE 329.4 Active calls 314
> CPU USAGE 312.5 Active calls 317
> CPU USAGE 125.0 Active calls 317
> CPU USAGE 137.5 Active calls 316
> CPU USAGE 123.5 Active calls 316
> CPU USAGE 117.6 Active calls 315
> CPU USAGE 117.6 Active calls 314
> CPU USAGE 135.3 Active calls 312
> CPU USAGE 131.2 Active calls 309
> CPU USAGE 131.2 Active calls 308
> CPU USAGE 118.8 Active calls 309
> CPU USAGE 331.2 Active calls 310
> CPU USAGE 111.8 Active calls 309
> udp0   1280 0.0.0.0:15598   0.0.0.0:*
>
> CPU USAGE 517.6 Active calls 309
> CPU USAGE 137.5 Active calls 308
> CPU USAGE 143.8 Active calls 308
> CPU USAGE 137.5 Active calls 307
> CPU USAGE 311.8 Active calls 308
> CPU USAGE 137.5 Active calls 308
> CPU USAGE 125.0 Active calls 307
> CPU USAGE 156.2 Active calls 307
> udp0   1280 0.0.0.0:19148   0.0.0.0:*
>
> CPU USAGE 135.3 Active calls 305
> CPU USAGE 305.9 Active calls 307
> CPU USAGE 156.2 Active calls 309
> CPU USAGE 450.0 Active calls 311
> CPU USAGE 137.5 Active calls 310
> CPU USAGE 137.5 Active calls 310
> CPU USAGE 256.2 Active calls 311
> CPU USAGE 150.0 Active calls 308
> CPU USAGE 281.2 Active calls 308
> CPU USAGE 281.2 Active calls 307
> CPU USAGE 137.5 Active calls 304
> udp0   1280 0.0.0.0:11284   0.0.0.0:*
>
> CPU USAGE 135.3 Active calls 306
> CPU USAGE 150.0 Active calls 309
> CPU USAGE 331.2 Active calls 310
> CPU USAGE 125.0 Active calls 308
> CPU USAGE 117.6 Active calls 307
> CPU USAGE 135.3 Active calls 306
> udp0   1280 0.0.0.0:18494   0.0.0.0:*
>
> CPU USAGE 137.5 Active calls 307
> CPU USAGE 141.2 Active calls 306
> CPU USAGE 137.5 Active calls 307
> CPU USAGE 562.5 Active calls 308
> CPU USAGE 862.5 Active calls 311
> CPU USAGE 164.7 Active calls 310
> udp0768 0.0.0.0:17890   0.0.0.0:*
>
> udp0   1280 0.0.0.0:11504   0.0.0.0:*
>
> [root@a30 ~]#
>
>
>
> On Tue, Oct 27, 2020 at 1:21 PM Jon Bonilla (Manwe) 
> wrote:
>
>> El Tue, 27 Oct 2020 12:52:47 -0400
>> Dovid Bender  escribió:
>>
>> > Hi,
>> >
>> > Sorry in advance that I am emailing the users list and not the biz list
>> I
>> > think I will find my target audience here. We are looking to hire a
>> > consultant to help us figure out an issue. We are having what seems are
>> > "random load" issues with bare metal boxes that are dedicated to
>> Asterisk
>> > and a few Perl AGI's. We went after all the usual suspects (CPU IDLE,
>> > memory usage etc.). I wrote the simple bash script below to show me
>> total
>> > calls and CPU usage of Asterisk
>> > #! /bin/bash
>> >
>> >
>>
>> Did you try FastAGI? Asterisk doesn't handle spawning many processes very
>> well.
>> It's a bottleneck.
>>
>>
>>
>>
>> --
>> PekePBX, the multitenant PBX solution
>> https://pekepbx.com
>>
>
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[asterisk-users] astdbdir in asterisk.conf has no effect

[Jeff Pyle]

Hello,

On Asterisk 13.19 I'm trying to set astdbdir in /etc/asterisk.conf, but
it's not having any effect.  For example:

# grep astdbdir /etc/asterisk/asterisk.conf
astdbdir => /tmp/asterisk

/tmp/asterisk exists and is owned by asterisk:asterisk, as the asterisk
processing is running as asterisk:asterisk with the config file forced:

# ps aux | grep asterisk
asterisk  3389 13.1  3.5 4295146504 214004 ?   S /tmp/asterisk/astdb, but no change.  It's still
running in /var/lib/asterisk.

I feel like I'm missing something.  Any suggestions?


- Jeff
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Re: [asterisk-users] Bug in Dial() string processing

[Antony Stone]

On Tuesday 27 October 2020 at 11:00:10, Antony Stone wrote:

> Hi.
> 
> I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1
> at least).

I've now confirmed that the same bug exists in 16.2.1

A Dial() command containing a SIP username/password combination which has a ! 
character in it is incorrectly parsed and fails.

> According to the documentation in channels/chan_sip.c the Dial() string
> syntax is:
> 
>  * SIP/devicename
>  * or SIP/username@domain (SIP uri)
>  * or SIP/username[:password[:md5secret[:authname[:transport@host[:port]
>  * or SIP/devicename/extension
>  * or SIP/devicename/extension/IPorHost
>  * or SIP/username@domain//IPorHost
>  *
>  * and there is an optional [!dnid] argument you can append to alter the
>  * To: header.
> 
> The bug is in the last part - the optional !dnid argument.
> 
> This should parse for a ! *following* any of the above syntaxes and assign
> whatever follows it to dnid.  If there is more than one ! in the entire
> string, dnid should follow the final one.
> 
> Unfortunately, the string parsing checks *first* for a ! and assigns
> everything following the first occurrence to dnid, and then attempts to
> parse whatever preceded it according to the above possibilities.
> 
> This means that if a SIP password, for example, contains a !, it is
> impossible to use the stated syntax
> 
> SIP/username[:password[:md5secret[:authname[:transport@host[:port]
> 
> Note that in this syntax, @host is non-optional, therefore parsing for a !
> preceding the @host part and regarding what follows the ! as dnid is
> incorrect.
> 
> 
> Regards,
> 
> 
> Antony.

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Re: [asterisk-users] PJSIP tight loop on auth failure

[Kingsley Tart - Barritel Ltd]

Nice idea but I think fail2ban is a bit too much of a blunt tool for
this. What's more, I think fail2ban works by following logs and nothing
gets logged here. You'd have to do an ngrep or tcpdump really. I
suppose fail2ban could be configured to parse the output format.

All it needs is the network blocked for about 1 second, so what I'll
probably do is have a daemon tail output from this command:

ngrep -t -d br0 -q -W single Proxy-Authorization port 5060

and if it sees multiple auth invites sent with the same Call-ID then
just knobble that route for literally a second or so. PJSIP then hangs
up that channel with a 404 pretty quickly, so I've found if I do this
(example for remote host on 123.123.123.123):

route add 123.123.123.123 lo && sleep 1 && route del 123.123.123.123 lo

Cheers,
Kingsley.

On Thu, 2020-10-29 at 08:39 +0100, Olivier wrote:
> Hi,
> What if some fail2ban magic could keep OpenSIPs response from hitting
> Asterisk after N attempts ?
> 
> Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd <
> kingsley.t...@barritel.com> a écrit :
> > Hi,
> > 
> > We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
> > 
> > I've found an issue when Asterisk tries to make a SIP call out
> > using
> > auth, but has the wrong credentials and keeps getting returned a
> > SIP
> > 407, in this example to an OpenSIPs server requiring user auth.
> > 
> > Basically this happens:
> > 
> >1. Asterisk sends plain INVITE to OpenSIPs
> >2. OpenSIPs responds with SIP 407 auth required with a Proxy-
> >   Authenticate header
> >3. Asterisk re-sends INVITE to OpenSIPs with Proxy-Authorization
> >   header, but has the wrong password
> >4. goto step 2 and repeat forever
> > 
> > So what we're seeing is Asterisk re-sending an INVITE with
> > incorrect
> > auth (which is clearly never going to work), about every 2ms.
> > 
> > The Call-ID remains the same all of the time.
> > 
> > Shouldn't PJSIP realise that this isn't going to work after a few
> > tries
> > and give up?
> > 
> > The only way I've found of stopping the seemingly infinite loop is
> > to
> > either restart Asterisk or temporarily block network traffic
> > between
> > the two machines in order to break the cycle.
> > 
> > Any idea whether this has been fixed in a later version?
> > 
> > This is basically the response coming back from OpenSIPs
> > (anonymised),
> > whether Asterisk didn't provide 
> > 
> > SIP/2.0 407 Proxy Authentication Required
> > Via: SIP/2.0/UDP
> > 100.101.102.103:5060;received=100.101.102.103;rport=5060;branch=z9h
> > G4bKPja942e87d-c501-4834-9184-f002c3fd53d2
> > From: ;tag=075f669f-9115-42a8-
> > 8c98-6170a2910e4b
> > To: ;tag=c97b4d1cb1f3d0da
> > 549e06a8d482ef63.fefa
> > Call-ID: f79caf90-5b95-4db7-966b-a42e2d372c90
> > CSeq: 34157 INVITE
> > Proxy-Authenticate: Digest realm="sip.example.com",
> > nonce="5f96c21800011caac9f7e901848de60a1e186b402bd9b710",
> > qop="auth"
> > Server: OpenSIPS (1.11.6-tls (x86_64/linux))
> > Content-Length: 0
> > 
> > The caveat is that whether what OpenSIPs is doing is correct or
> > broken,
> > our customers can edit the auth on their own SIP gateways, so our
> > system needs to be able to handle it properly.
> > 
> > Cheers,
> > Kingsley.
> > 
> > 
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[asterisk-users] Suden "ast_db_put: Couldn't execute statment" in 13.14.1 after high rate of incoming REGISTERs

[Olivier]

Hello,

The other day, a 13.14.1 server suddenly stopped working correctly.
First, it printed:
Oct 23 21:53:40 FOOBAR asterisk[2377]: WARNING[27942]: db.c:332 in
ast_db_put: Couldn't execute statment: SQL logic error or missing database

This occurred while this server received a lot incoming REGISTER such as:
Oct 23 21:53:40 FOOBAR asterisk[2377]: [Oct 23 21:53:40]
#033[1;33mNOTICE#033[0m[20418]:
#033[1;37mres_pjsip/pjsip_distributor.c#033[0m:#033[1;37m525#033[0m
#033[1;37mlog_failed_request#033[0m: Request 'REGISTER' from '<
sip:johndoe@100.64.0.1>' failed for '10.50.59.5:5060' (callid:
OA48732DD2johndoe42259F1C) - Failed to authenticate

I estimate incoming (invalid) REGISTER rate to be 120 per second, (which is
by the way, well above the upper limit we load tested our system before
putting it online).

What would you suggest me to do ?

Best regards
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Re: [asterisk-users] PJSIP tight loop on auth failure

[Olivier]

Hi,
What if some fail2ban magic could keep OpenSIPs response from hitting
Asterisk after N attempts ?

Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd <
kingsley.t...@barritel.com> a écrit :

> Hi,
>
> We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
>
> I've found an issue when Asterisk tries to make a SIP call out using
> auth, but has the wrong credentials and keeps getting returned a SIP
> 407, in this example to an OpenSIPs server requiring user auth.
>
> Basically this happens:
>
>1. Asterisk sends plain INVITE to OpenSIPs
>2. OpenSIPs responds with SIP 407 auth required with a Proxy-
>   Authenticate header
>3. Asterisk re-sends INVITE to OpenSIPs with Proxy-Authorization
>   header, but has the wrong password
>4. goto step 2 and repeat forever
>
> So what we're seeing is Asterisk re-sending an INVITE with incorrect
> auth (which is clearly never going to work), about every 2ms.
>
> The Call-ID remains the same all of the time.
>
> Shouldn't PJSIP realise that this isn't going to work after a few tries
> and give up?
>
> The only way I've found of stopping the seemingly infinite loop is to
> either restart Asterisk or temporarily block network traffic between
> the two machines in order to break the cycle.
>
> Any idea whether this has been fixed in a later version?
>
> This is basically the response coming back from OpenSIPs (anonymised),
> whether Asterisk didn't provide
>
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 100.101.102.103:5060
> ;received=100.101.102.103;rport=5060;branch=z9hG4bKPja942e87d-c501-4834-9184-f002c3fd53d2
> From:  >;tag=075f669f-9115-42a8-8c98-6170a2910e4b
> To:  >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.fefa
> Call-ID: f79caf90-5b95-4db7-966b-a42e2d372c90
> CSeq: 34157 INVITE
> Proxy-Authenticate: Digest realm="sip.example.com",
> nonce="5f96c21800011caac9f7e901848de60a1e186b402bd9b710", qop="auth"
> Server: OpenSIPS (1.11.6-tls (x86_64/linux))
> Content-Length: 0
>
> The caveat is that whether what OpenSIPs is doing is correct or broken,
> our customers can edit the auth on their own SIP gateways, so our
> system needs to be able to handle it properly.
>
> Cheers,
> Kingsley.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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