Re: [asterisk-users] Support for inbound UPDATE request

2017-07-08 Thread Daniel Tryba
On Fri, Jul 07, 2017 at 07:44:26PM +0530, Rahul MathuR wrote:
> Could you please let me know whether the latest Asterisk has a support for
> inbound UPDATE ?
> 
> In my case, the carrier is sending an UPDATE to change the codec which is
> replied by 5xx from Asterisk 11.17.1.

Asterisk 13/PJSIP supports inbound UPDATEs, I see them all the time as
inbound Session-Timer mechanism.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_direct_media_method
mentions support for SDP renegotiation via UPDATE if supported by the
endpoints being preferred above re-INVITEs.


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Re: [asterisk-users] AMI column widths

2017-07-08 Thread Marcelo Terres
You are using AMI to run CLI commands and that's the problem.

Try to use the equivalent AMI actions to get the information that you want.

My suggestion : get all channels in use (CoreShowChannels) and then filter
just the SIP, since there is not an action to do exactly what you need.


On 8 Jul 2017 9:17 am, "Antony Stone" 
wrote:

On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:

> There are no sip show channels on AMI. Also, the output that you sent is
> not a AMI output. Are u using AMI ou running commands on console?

I'm using AMI.

I have a connection to the Asterisk server on port 5038, initated with:

Action: Login
Username: x
Secret: y
Events: off

I receive back:

Response: Success
Message: Authentication accepted

I then issue:

Action: Command
Command: SIP show channels

and I get back:

Response: Follows
Privilege: Command
Peer User/ANR Call ID  Format
 Hold
Last MessageExpiry Peer

plus the data I quoted previously.

> Running commands on console and parsing the output is the worst way to
> obtain data, first because it is not easily parseable.

And also because it is very inefficient with connection setups, I believe.

> Second, it doesn't show you all data.
>
> Third, you can have these truncate problems, because that's not intention
> of CLI.
>
> Using proper AMI Actions you will probably achieve your goals
>
> https://wiki.asterisk.org/wiki/display/AST/AMI+Actions

Hm, I don't see anything there which will give me a list of the SIP channels
currently in use - what command should I be using for that?


Thanks,


Antony.

> On 7 Jul 2017 10:32 pm, Antony Stone wrote:
>
> Hi.
>
> I'm trying to get a list of the channels currently in use on an Asterisk
> server (1.8.32.1 if it matters) over AMI.
>
> I send the command "sip show channels", and I get back a response along
the
> lines of (* used to protect the innocent...):
>
> Peer User/ANR Call ID  Format   Hold
>  Last MessageExpiry Peer
> *8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
>  Tx: ACKSIPtrunkNu
> *.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
>  Rx: OPTIONS
> *.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
>  Rx: ACK200101
> *.1*.19.70 (None)   149831567021051  0x0 (nothing)No
>  Rx: REGISTER   
>
> So, firstly, the "Call ID" column is clearly truncated, because it should
> show more than is indicated above,
> but more importantly for me, the "Peer" column is truncated, and what
> should show as "SIPtrunkNumber8"
> is only shown as "SIPtrunkNu".
>
> How can I get the full column widths of these items shown in the output?
>
> Note that it is not a solution just to say "don't call it
> 'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
> this name has also been modified slightly to conceal the real name of the
> trunk, which is actually longer
> than "SIPtrunkNo8", but still with the most important information at the
> end.
>
> What I'm looking for is how to get the *full* details of all the channels
> shown.
>
> I have checked, and there is no "verbose" option to the "sip show
channels"
> command.
>
>
> Thanks,
>
>
> Antony.

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Re: [asterisk-users] AMI column widths

2017-07-08 Thread Antony Stone
On Saturday 08 July 2017 at 10:16:19, Antony Stone wrote:

> On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:
> > There are no sip show channels on AMI. Also, the output that you sent is
> > not a AMI output. Are u using AMI ou running commands on console?
> 
> I'm using AMI.
> 
> I have a connection to the Asterisk server on port 5038, initated with:
> 
>   Action: Login
>   Username: x
>   Secret: y
>   Events: off
> 
> I receive back:
> 
>   Response: Success
>   Message: Authentication accepted
> 
> I then issue:
> 
>   Action: Command
>   Command: SIP show channels
> 
> and I get back:
> 
>   Response: Follows
>   Privilege: Command
>   Peer User/ANR Call ID  Format   Hold
> Last MessageExpiry Peer
> 
> plus the data I quoted previously.
> 
> > Running commands on console and parsing the output is the worst way to
> > obtain data, first because it is not easily parseable.
> 
> And also because it is very inefficient with connection setups, I believe.
> 
> > Second, it doesn't show you all data.
> > 
> > Third, you can have these truncate problems, because that's not intention
> > of CLI.
> > 
> > Using proper AMI Actions you will probably achieve your goals
> > 
> > https://wiki.asterisk.org/wiki/display/AST/AMI+Actions
> 
> Hm, I don't see anything there which will give me a list of the SIP
> channels currently in use - what command should I be using for that?

Hm, Action: CoreShowChannels looks like it can be made to work - it's not 
specifically SIP, but I can parse that out of the channel name.  Strange that 
there is a DAHDIShowChannels command, and a CoreShowChannels, but no 
SIPShowChannels..

If anyone has a better idea, please let me know...


> Thanks,
> 
> 
> Antony.
> 
> > On 7 Jul 2017 10:32 pm, Antony Stone wrote:
> > 
> > Hi.
> > 
> > I'm trying to get a list of the channels currently in use on an Asterisk
> > server (1.8.32.1 if it matters) over AMI.
> > 
> > I send the command "sip show channels", and I get back a response along
> > the lines of (* used to protect the innocent...):
> > 
> > Peer User/ANR Call ID  Format   Hold
> >  Last MessageExpiry Peer
> > *8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
> >  Tx: ACKSIPtrunkNu
> > *.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
> >  Rx: OPTIONS
> > *.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
> >  Rx: ACK200101
> > *.1*.19.70 (None)   149831567021051  0x0 (nothing)No
> >  Rx: REGISTER   
> > 
> > So, firstly, the "Call ID" column is clearly truncated, because it should
> > show more than is indicated above,
> > but more importantly for me, the "Peer" column is truncated, and what
> > should show as "SIPtrunkNumber8"
> > is only shown as "SIPtrunkNu".
> > 
> > How can I get the full column widths of these items shown in the output?
> > 
> > Note that it is not a solution just to say "don't call it
> > 'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
> > this name has also been modified slightly to conceal the real name of the
> > trunk, which is actually longer
> > than "SIPtrunkNo8", but still with the most important information at the
> > end.
> > 
> > What I'm looking for is how to get the *full* details of all the channels
> > shown.
> > 
> > I have checked, and there is no "verbose" option to the "sip show
> > channels" command.
> > 
> > 
> > Thanks,
> > 
> > 
> > Antony.

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Re: [asterisk-users] AMI column widths

2017-07-08 Thread Antony Stone
On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:

> There are no sip show channels on AMI. Also, the output that you sent is
> not a AMI output. Are u using AMI ou running commands on console?

I'm using AMI.

I have a connection to the Asterisk server on port 5038, initated with:

Action: Login
Username: x
Secret: y
Events: off

I receive back:

Response: Success
Message: Authentication accepted

I then issue:

Action: Command
Command: SIP show channels

and I get back:

Response: Follows
Privilege: Command
Peer User/ANR Call ID  Format   
Hold 
Last MessageExpiry Peer

plus the data I quoted previously.

> Running commands on console and parsing the output is the worst way to
> obtain data, first because it is not easily parseable.

And also because it is very inefficient with connection setups, I believe.

> Second, it doesn't show you all data.
> 
> Third, you can have these truncate problems, because that's not intention
> of CLI.
> 
> Using proper AMI Actions you will probably achieve your goals
> 
> https://wiki.asterisk.org/wiki/display/AST/AMI+Actions

Hm, I don't see anything there which will give me a list of the SIP channels 
currently in use - what command should I be using for that?


Thanks,


Antony.

> On 7 Jul 2017 10:32 pm, Antony Stone wrote:
> 
> Hi.
> 
> I'm trying to get a list of the channels currently in use on an Asterisk
> server (1.8.32.1 if it matters) over AMI.
> 
> I send the command "sip show channels", and I get back a response along the
> lines of (* used to protect the innocent...):
> 
> Peer User/ANR Call ID  Format   Hold
>  Last MessageExpiry Peer
> *8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
>  Tx: ACKSIPtrunkNu
> *.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
>  Rx: OPTIONS
> *.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
>  Rx: ACK200101
> *.1*.19.70 (None)   149831567021051  0x0 (nothing)No
>  Rx: REGISTER   
> 
> So, firstly, the "Call ID" column is clearly truncated, because it should
> show more than is indicated above,
> but more importantly for me, the "Peer" column is truncated, and what
> should show as "SIPtrunkNumber8"
> is only shown as "SIPtrunkNu".
> 
> How can I get the full column widths of these items shown in the output?
> 
> Note that it is not a solution just to say "don't call it
> 'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
> this name has also been modified slightly to conceal the real name of the
> trunk, which is actually longer
> than "SIPtrunkNo8", but still with the most important information at the
> end.
> 
> What I'm looking for is how to get the *full* details of all the channels
> shown.
> 
> I have checked, and there is no "verbose" option to the "sip show channels"
> command.
> 
> 
> Thanks,
> 
> 
> Antony.

-- 
Having been asked for a reference for this man,
I can confirm that you will be very lucky indeed if you can get him to work for 
you.

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

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

New to Asterisk? Start here:
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