Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

Wait a second, I thought in your original email that you said that
Asterisk was generating reinvites.  It sounds now like you're saying
that the remote side is initiating reinvites instead.

My understanding is that the canreinvite/directmedia option only
influences Asterisk's behavior with regards to generating reinivites.
If it receives a reinvite, I don't think these options will do
anything about that.  In fact, I'd guess that not properly responding
to a received reinvite is going to potentially break things from the
SIP perspective.

Matthew Fredrickson


On Wed, Aug 10, 2016 at 4:53 PM, Tammy Firefly  wrote:
>
>
> On 8/9/16 12:40 PM, Matt Fredrickson wrote:
>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  
>> wrote:
>>> Hi All,
>>>
>>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>>> split off to where they need to go.  We are having a problem getting
>>> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
>>> sending a reinvite which their side & they do not support us sending a
>>> reinvite.  Ive tried:
>>>
>>> canreinvite=no which was supposedly replaced by:
>>>
>>> directmedia=no
>>>
>>> Can anyone shed any light on this matter?  I'd love to get this fixed.
>>>
>>
>> Those options *should* influence chan_sip's reinvite behavior - at
>> least they have from my experiences working with chan_sip.  Do you
>> know what is triggering the reinvite in the first place, or does it
>> look like a normal media reinvite?
>>
>
>
> every 15 minutes vitelity sends a re-invite to keep the call going.  I
> have a packet capture from it if you'd like it feel free to email me off
> list @ tamara.wis...@wiztech.biz
>
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Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Tammy Firefly]

On 8/9/16 12:40 PM, Matt Fredrickson wrote:
> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  wrote:
>> Hi All,
>>
>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>> split off to where they need to go.  We are having a problem getting
>> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
>> sending a reinvite which their side & they do not support us sending a
>> reinvite.  Ive tried:
>>
>> canreinvite=no which was supposedly replaced by:
>>
>> directmedia=no
>>
>> Can anyone shed any light on this matter?  I'd love to get this fixed.
>>
> 
> Those options *should* influence chan_sip's reinvite behavior - at
> least they have from my experiences working with chan_sip.  Do you
> know what is triggering the reinvite in the first place, or does it
> look like a normal media reinvite?
> 


every 15 minutes vitelity sends a re-invite to keep the call going.  I
have a packet capture from it if you'd like it feel free to email me off
list @ tamara.wis...@wiztech.biz

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Re: [asterisk-users] Original Callerid on transfer in asterisk 13

[Israel Gottlieb]

In 11 setting trustrpid sendrpid is enough that phone getting the tranfered
call shows the name and number of the caller and not the tranferer
In 13 the same shows the transferrs info

בתאריך 11 באוג׳ 2016 00:21,‏ "Matt Fredrickson"  כתב:

> How are you attempting to view the original CallerId?
>
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb  wrote:
> > Hi
> > Is there any configuration change in asterisk 13.9.1 to show original
> > callerid on a transfer
> > In asterisk 11.21 it works as expected
> >
> > Thanks
> >
> >
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Re: [asterisk-users] Original Callerid on transfer in asterisk 13

[Matt Fredrickson]

How are you attempting to view the original CallerId?

Matthew Fredrickson

On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb  wrote:
> Hi
> Is there any configuration change in asterisk 13.9.1 to show original
> callerid on a transfer
> In asterisk 11.21 it works as expected
>
> Thanks
>
>
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse.  WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens  wrote:
> Hello
>
> thank you for your answer.
>
> I don't understand how there are many tutorials and examples on the web
> where every time the outcome is a working setup. Very strange I feel now
> after my personal experience with Asterisk 11 and webRTC.
>
> You also say Asterisk 13. How about Asterisk 12 then ??
>
>
>
> Kind regards.
>
>
>
> On 10-08-16 21:53, Matt Fredrickson wrote:
>
> I don't see an ice-ufrag or ice-pwd line in the response from
> Asterisk, correlating with your suspicion that there is no ICE.  Are
> you sure that the stun server you're using (the google one) still
> works?  I haven't tried that server in a while, but I distantly seem
> to recall that maybe they shut it down.
>
> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
> feature updated in a while, and it could be that it could be a number
> of patches/fixes behind with regards to webrtc support, particularly
> with regards to interoperating with a modern browser version.
>
> Hope that helps,
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens 
> wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
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>
>
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Jonas Kellens]

Hello

thank you for your answer.

I don't understand how there are many tutorials and examples on the web 
where every time the outcome is a working setup. Very strange I feel now 
after my personal experience with Asterisk 11 and webRTC.


You also say Asterisk 13. How about Asterisk 12 then ??



Kind regards.



On 10-08-16 21:53, Matt Fredrickson wrote:

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens  wrote:

On 10-08-16 08:52, Ludovic Gasc wrote:

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/




Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.



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[asterisk-users] Original Callerid on transfer in asterisk 13

[Israel Gottlieb]

Hi
Is there any configuration change in asterisk 13.9.1 to show original
callerid on a transfer
In asterisk 11.21 it works as expected

Thanks
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens  wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _
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Re: [asterisk-users] Replacement for phpagi?

[Alex Villací­s Lasso]

El 10/08/16 a las 12:06, Carlos Chavez escribió:

Anyone know a good replacement for phpagi? Unfortunately development 
stalled long ago and it has not been updated.  What is the best solution for 
AMI and AGI on PHP? Thanks.



In the case of AMI, you could use the AMI client from the Elastix CallCenter 
dialer daemon:

https://sourceforge.net/p/elastix/code/HEAD/tree/branches/2.5.0/apps/extras/callcenter/setup/dialer_process/dialer/AMIClientConn.class.php

This class was once based on phpagi-asmanager.php but has since been completely 
rewritten to make use of an internal non-blocking I/O model. The main internal 
client is the AMIEventProcess class in the same project directory.


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[asterisk-users] Realtime warnings for database structure

[Carlos Chavez]

I keep getting messages like these in the cli:

[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'qualify_timeout' cannot be 
type 'int(10)' (need char)
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1246 require_mysql: 
Possibly unsupported column type 'enum('yes','no')' on column 
'authenticate_qualify'
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'via_port' cannot be type 
'int(11)' (need char)


Since I am using alembic with the "official" database table 
structures I simply do not understand why.  Who is wrong here, the 
alembic structure or realtime for expecting different things? Obviously 
I do not want to make changes to the database as that will break the 
updates to new versions so it probably needs to be fixed on the realtime 
side.  I just hope that, since they are warnings, they do not affect 
regular operations.


I also get this error every time an endpoint registers to my Asterisk:

[Aug 10 12:24:32] ERROR[23411]: res_pjsip_registrar.c:411 
register_aor_core: Unable to bind contact 
'sip:4...@xx.xx.xx.xx:48007;transport=UDP;rinstance=34b3595c6901f19e' to 
AOR '4001'


I do not know if this is related to the same database problems, I 
have never seen that message when using text files for configuration.  
Contacts do get created:


  Contact:
 

==

  Contact:  4001/sip:4...@xx.xx.xx.xx:48007;transport=UD 653ab7af98 
Created   0.000


ODBC for realtime is still very unstable so I hope we can get a 
little more stability from Mysql while development continues.  So far my 
Asterisk 13.10 installations work fine but the warnings and error clog 
the log files and cli.  I can ignore that but when my clients peer over 
my shoulder they freak out.  Should I open an issue on jira?


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[asterisk-users] Calls are dropped after 15 minutes

[Marlon Araujo]

What version of asterisk are you on?

Marlon Araujo

> On Aug 10, 2016, at 13:00, asterisk-users-requ...@lists.digium.com wrote:
> 
> Calls are dropped after 15 minutes

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[asterisk-users] Replacement for phpagi?

[Carlos Chavez]

Anyone know a good replacement for phpagi?  Unfortunately 
development stalled long ago and it has not been updated.  What is the 
best solution for AMI and AGI on PHP?  Thanks.



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Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

[Jacek Konieczny]

On 2016-08-10 11:53, Joshua Colp wrote:

Jacek Konieczny wrote:

On 2016-08-09 10:06, Faheem Muhammad wrote:

trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
Call setup time by configuring 'T2' upto you choice as per you telephony
network. Configure t1min, timert1 and timerb according to your network.


No, that won't work.

First – 't1min', 'timert1' and 'timerb' are chan_sip, not pjsip options.

Second – the 'timer_t1' and 'timer_b' settings available in 'pjsip.conf'
are global and not per-endpoint. I cannot change T1 for trunks, as they
might not be fast enough to respond and I cannot set it for phones only.

It seems I need to bring back the chan_sip behaviour – 'do not bother
with INVITE to Unreachable devices'.


I'd suggest filing an issue on the issue tracker[1] for this. It's
reasonable behavior.


Done:
https://issues.asterisk.org/jira/browse/ASTERISK-26281

I just wanted to make sure I am not missing something, first.

Jacek

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Jonas Kellens]

On 10-08-16 08:52, Ludovic Gasc wrote:


For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/





Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.


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Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

[Joshua Colp]

Jacek Konieczny wrote:

On 2016-08-09 10:06, Faheem Muhammad wrote:

trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
Call setup time by configuring 'T2' upto you choice as per you telephony
network. Configure t1min, timert1 and timerb according to your network.


No, that won't work.

First – 't1min', 'timert1' and 'timerb' are chan_sip, not pjsip options.

Second – the 'timer_t1' and 'timer_b' settings available in 'pjsip.conf'
are global and not per-endpoint. I cannot change T1 for trunks, as they
might not be fast enough to respond and I cannot set it for phones only.

It seems I need to bring back the chan_sip behaviour – 'do not bother
with INVITE to Unreachable devices'.


I'd suggest filing an issue on the issue tracker[1] for this. It's 
reasonable behavior.


[1] https://issues.asterisk.org/jira

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Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

[Jacek Konieczny]

On 2016-08-09 10:06, Faheem Muhammad wrote:

trip time and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
Call setup time by configuring 'T2' upto you choice as per you telephony
network. Configure t1min, timert1 and timerb according to your network.


No, that won't work.

First – 't1min', 'timert1' and 'timerb' are chan_sip, not pjsip options.

Second – the 'timer_t1' and 'timer_b' settings available in 'pjsip.conf' 
are global and not per-endpoint. I cannot change T1 for trunks, as they 
might not be fast enough to respond and I cannot set it for phones only.


It seems I need to bring back the chan_sip behaviour – 'do not bother 
with INVITE to Unreachable devices'.



Jacek


On Tue, Aug 9, 2016 at 12:03 PM, Jacek Konieczny > wrote:

Hi,

We have been migrating our PBX system from Asterisk 1.8 and chan_sip to
Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have
stumbled on a behaviour difference I don't like.

With chan_pjsip when a phone went unexpectedly offline (Ethernet cable
disconnected) Asterisk would detect this quickly (through the 'qualify'
pings), mark the phone as 'Unavailable' and fail immediately with
'CHANUNAVAIL' when dialling this phone.

With Asterisk 13 and chan_pjsip qualify still works for determining
current phone availability (endpoint shown as 'Unavailable' shortly
after disconnecting the cable), but the phone is being dialled like
nothing is wrong – Asterisk sends the INVITE and waits for the response,
until SIP timeout (a bit more than 30s total). That is much longer time
until 'CHANUNAVAIL' than I expect. It is also longer than the dial
timeout in some cases, so I would get 'NOANSWER' instead of
'CHANUNAVAIL' which breaks my dialplan logic.

Is that that the expected behaviour, a bug or a configuration problem?
Am I supposed to check for device availability in my dialplan?

Greets,
Jacek

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Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

[Jacek Konieczny]

On 2016-08-09 10:06, Faheem Muhammad wrote:

Jacek,
This might be a bug or configuration issue, but you need to understand
the SIP Session Timers. With Session Timers you can control the round
trip time and Call Setup time of SIP Requests.


I don't think you really mean SIP Session Timers 
(https://tools.ietf.org/html/rfc4028) these do not affect RTT or call 
setup, but provide kind of 'keepalive' and session expiration for 
established calls.



In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms default). Similarly you can reduce the
Call setup time by configuring 'T2' upto you choice as per you telephony
network. Configure t1min, timert1 and timerb according to your network.
Also set session-type=uas.


Yes, tweaking the T1 and T2 timers may work for me. I'll try that, 
though the old 'qualify' magic with chan_sip was quite convenient. I 
wonder why it doesn't work with chan_pjsip.


Jacek

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Ludovic Gasc]

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
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