[asterisk-users] CONNECTEDLINE

[Magnus Benngård]

Gentlemen, 

Did tryout "CONNECTEDLINE" function, was exactly what I have been looking
for. But there are at least one thing I cant figure out. 

Did a very simple and "stupid" extension 0317998955 and ran a test. 

My phone (0317998975) dials 955, the display on my phone changes from
"955" to "Connected Line 955" when my call is answered,
shouldn't the display on my phone change to "Connected Line 0317998955"?

exten => 956,1,Goto(0317998956,1)

exten => 0317998956,1,Set(CONNECTEDLINE(all)=Connected Line )
exten => 0317998956,n,Answer()
exten => 0317998956,n,Wait(2)
exten => 0317998956,n,Hangup()

 -- Executing [...@inputinterior.se:1] Goto("SIP/0317998975-0004",
"0317998956,1") in new stack
 -- Goto (inputinterior.se,0317998956,1)
 -- Executing [0317998...@inputinterior.se:1]
Set("SIP/0317998975-0004", "CONNECTEDLINE(all)=Connected Line ") in new
stack
 -- Executing [0317998...@inputinterior.se:2]
Answer("SIP/0317998975-0004", "") in new stack
 -- Executing
[0317998...@inputinterior.se:3]
Wait("SIP/0317998975-0004", "2") in new stack
 -- Executing [0317998...@inputinterior.se:4]
Hangup("SIP/0317998975-0004", "") in new stack
 == Spawn extension (inputinterior.se, 0317998956, 4) exited non-zero on
'SIP/0317998975-0004' 

Asterisk SVN-trunk-r245147M built by root @ sip on a i686 running Linux on
2010-01-25 11:10:15 UTC 

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Re: [asterisk-users] CONNECTEDLINE

[Kyle Kienapfel]

You should take a look and see if any SIP packets are going out that
mention "Connected Line 0317998955" as either something is or isn't
sent out from the asterisk server.

On Sat, Feb 6, 2010 at 4:30 AM, Magnus Benngård
 wrote:
> Gentlemen,
>
> Did tryout "CONNECTEDLINE" function, was exactly what I have been looking
> for. But there are at least one thing I cant figure out.
>
> Did a very simple and "stupid" extension 0317998955 and ran a test.
>
> My phone (0317998975) dials 955, the display on my phone changes from "955"
> to "Connected Line 955" when my call is answered,
> shouldn't the display on my phone change to "Connected Line 0317998955"?
>
> exten => 956,1,Goto(0317998956,1)
>
> exten => 0317998956,1,Set(CONNECTEDLINE(all)=Connected Line <0317998955>)
> exten => 0317998956,n,Answer()
> exten => 0317998956,n,Wait(2)
> exten => 0317998956,n,Hangup()
>
>     -- Executing [...@inputinterior.se:1] Goto("SIP/0317998975-0004",
> "0317998956,1") in new stack
>     -- Goto (inputinterior.se,0317998956,1)
>     -- Executing [0317998...@inputinterior.se:1]
> Set("SIP/0317998975-0004", "CONNECTEDLINE(all)=Connected Line
> <0317998955>") in new stack
>     -- Executing [0317998...@inputinterior.se:2]
> Answer("SIP/0317998975-0004", "") in new stack
>     -- Executing [0317998...@inputinterior.se:3]
> Wait("SIP/0317998975-0004", "2") in new stack
>     -- Executing [0317998...@inputinterior.se:4]
> Hangup("SIP/0317998975-0004", "") in new stack
>   == Spawn extension (inputinterior.se, 0317998956, 4) exited non-zero on
> 'SIP/0317998975-0004'
>
> Asterisk SVN-trunk-r245147M built by root @ sip on a i686 running Linux on
> 2010-01-25 11:10:15 UTC
>
> /Magnus
>
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[asterisk-users] CONNECTEDLINE endpoint support

[marek cervenka]

hi,

i'm testing CONNECTEDLINE function

https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

example dialplan

same => n,set(CONNECTEDLINE(name,i)=aastra)
same => n,set(CONNECTEDLINE(name-pres,i)=allowed)
same => n,Set(CONNECTEDLINE(num,i)=5551212)
same => n,Set(CONNECTEDLINE(num-pres)=allowed)
same => n,dial(SIP/sipline501,,I)

it only works with mitel(aastra 6767i) phones

i tested - grandstream 2130, jitsi, blink, microsip, zoiper - nothing worked

what devices working for you with CONNECTEDLINE function?

thanks

Marek



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[asterisk-users] CONNECTEDLINE and panasonic 500

[Dmitry Melekhov]

Hello!

I have following connections over isdn pri:

avaya definity---pri--asterisk--pri-panasonic 500

Just because panasonic 500 can't send user's names.

I also want to have reverse callerid for avaya users.

But if there is no answer in dial plan:

exten => _,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})})
;exten => _,n,Answer
exten => _,n,Dial(DAHDI/g4/${EXTEN})
exten => _,n,Hangup

there is no name on avaya display after panasonic user pick up...

With answer name appears immediately but this is not what we want :-(


Is it possible to do something? :-)

Thank you!


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[asterisk-users] CONNECTEDLINE(name) ISDN problem

[jg]

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only 
CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I 
did get back a name and a number and everything was displayed correctly. So I think the calling 
site should basically be able to handle all connected line info.


Looking at a pcap trace of the D-channel data, I see that CONNECTEDLINE(num) maps to the 
connected number information element and CONNECTEDLINE(name) to the display element. The pcap 
trace does actually contain my CONNECTEDLINE(name) plus a leading byte with a value of 0xB1 (or 
\261 in octal notation). This additional byte is part of the announced string length. Now I 
wonder, whether this byte is causing the trouble.


Does anybody know what this leading byte is actually doing there?

jg

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Re: [asterisk-users] CONNECTEDLINE endpoint support

[Doug Lytle]

>>> On Oct 19, 2016, at 8:37 AM, marek cervenka cerva...@gmail.com wrote:

>>> i'm testing CONNECTEDLINE function
>>> what devices working for you with CONNECTEDLINE function?

Cisco and Polycom

Doug

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[asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

Hi,

I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:

- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a SIP divert is in place? If so, how?

- Is it possible to have the COLP/COLR information updated when a SIP
attended transfer is completed? If so how?

In both of the above cases, there is no obvious dialplan execution
when the calls are redirected, diverted or masqueraded, so we cannot
update the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?

Thanks,
Steve

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Re: [asterisk-users] CONNECTEDLINE and panasonic 500

[Richard Mudgett]

On Mon, Nov 18, 2013 at 1:21 AM, Dmitry Melekhov  wrote:

> Hello!
>
> I have following connections over isdn pri:
>
> avaya definity---pri--asterisk--pri-panasonic 500
>
> Just because panasonic 500 can't send user's names.
>
> I also want to have reverse callerid for avaya users.
>
> But if there is no answer in dial plan:
>
> exten => _,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})})
> ;exten => _,n,Answer
> exten => _,n,Dial(DAHDI/g4/${EXTEN})
> exten => _,n,Hangup
>
> there is no name on avaya display after panasonic user pick up...
>
> With answer name appears immediately but this is not what we want :-(
>

You need to use the 'I' Dial option to inhibit the connected line date from
the
Dial from overwriting the values setup by your dialplan.  See [1].

Richard

[1]
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
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Re: [asterisk-users] CONNECTEDLINE and panasonic 500

[Dmitry Melekhov]

18.11.2013 20:51, Richard Mudgett пишет:




On Mon, Nov 18, 2013 at 1:21 AM, Dmitry Melekhov > wrote:


Hello!

I have following connections over isdn pri:

avaya definity---pri--asterisk--pri-panasonic 500

Just because panasonic 500 can't send user's names.

I also want to have reverse callerid for avaya users.

But if there is no answer in dial plan:

exten => _,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})})
;exten => _,n,Answer
exten => _,n,Dial(DAHDI/g4/${EXTEN})
exten => _,n,Hangup

there is no name on avaya display after panasonic user pick up...

With answer name appears immediately but this is not what we want :-(


You need to use the 'I' Dial option to inhibit the connected line date 
from the

Dial from overwriting the values setup by your dialplan. See [1].

Richard

[1] 
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information




Hello!

Thank you!
It works :-)

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Re: [asterisk-users] CONNECTEDLINE(name) ISDN problem

[Richard Mudgett]

On Thu, Mar 13, 2014 at 1:42 PM, jg  wrote:

> When I set CONNECTEDLINE() info for an incoming ISDN call, the calling
> party sees only CONNECTEDLINE(num) and the name does not get displayed.
> Some time ago I called a number, where I did get back a name and a number
> and everything was displayed correctly. So I think the calling site should
> basically be able to handle all connected line info.
>
> Looking at a pcap trace of the D-channel data, I see that
> CONNECTEDLINE(num) maps to the connected number information element and
> CONNECTEDLINE(name) to the display element. The pcap trace does actually
> contain my CONNECTEDLINE(name) plus a leading byte with a value of 0xB1 (or
> \261 in octal notation). This additional byte is part of the announced
> string length. Now I wonder, whether this byte is causing the trouble.
>
> Does anybody know what this leading byte is actually doing there?
>

AFAIK, It is some kind of character set code.  That byte is intentionally
put there for
switches that are not Q.SIG or ETSI.  Technically, the display IE is only
to be sent
from the network to the user.  So sending it to the network is undefined
and switch
dependent.

Connected line name support is fully supported only by Q.SIG since it
actually
defines how to pass the name.  Using the display IE is a defacto standard
but
is only really going to work in the network to user direction.

Richard
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Re: [asterisk-users] CONNECTEDLINE(name) ISDN problem

[jg]

AFAIK, It is some kind of character set code.  That byte is intentionally put 
there for
switches that are not Q.SIG or ETSI.  Technically, the display IE is only to be 
sent
from the network to the user.  So sending it to the network is undefined and 
switch
dependent.
Got it. The single instance where I saw a string on the display was when I called my telco's 
technical support.


Connected line name support is fully supported only by Q.SIG since it actually
defines how to pass the name.  Using the display IE is a defacto standard but
is only really going to work in the network to user direction.
Too bad, I was already dreaming of offering some customers a few nifty features, like greetings, 
email addresses for further contacts, booking confirmation numbers for hotels, etc...


I shall still go through all the charset options to see whether there is any effect. Meanwhile I 
managed to get a pcap trace of a calling device and the display element is not at all present. I 
guess they are filtering it out.


Thank you, Richard.

jg

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Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

[Richard Mudgett]

> I have read the excellent information here:
> 
> https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
> and believe I have an understanding of what is offered. I have a
> couple of questions:
> 
> - Is it possible to update COLP/COLR when a SIP redirect occurs, or
> when a SIP divert is in place? If so, how?

All redirecting activity is valid only before the associated calls are
answered.  After the calls are answered, it is connected-line updates.
The redirecting interception macros are invoked before the outgoing call is
answered when the outgoing call is redirected by an entity further down
the line.  If your Asterisk server is redirecting the call, the REDIRECTING
information is updated by normal dialplan activity before placing the next
outgoing call to the redirected to party.

> - Is it possible to have the COLP/COLR information updated when a SIP
> attended transfer is completed? If so how?

Transfers generate connected line update events automatically.  The connected
line interception macros give you a chance to alter the connected line
information as it is passed between the connected endpoints of the bridge.

> In both of the above cases, there is no obvious dialplan execution
> when the calls are redirected, diverted or masqueraded, so we cannot
> update the CONNECTEDLINE() information trivially. Or am I missing an
> obvious trick?

This is the purpose of the interception macros.

Richard

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Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

On 25 April 2012 16:55, Richard Mudgett  wrote:
[snip]
>
>> - Is it possible to have the COLP/COLR information updated when a SIP
>> attended transfer is completed? If so how?
>
> Transfers generate connected line update events automatically.  The connected
> line interception macros give you a chance to alter the connected line
> information as it is passed between the connected endpoints of the bridge.
>
>> In both of the above cases, there is no obvious dialplan execution
>> when the calls are redirected, diverted or masqueraded, so we cannot
>> update the CONNECTEDLINE() information trivially. Or am I missing an
>> obvious trick?
>
> This is the purpose of the interception macros.

Ah, thank you. I was looking at it back-to-front.

The key bit is "Transfers generate connected line update events
automatically." - I can now see this in the source code in
ast_do_masquerade() and elsewhere. This then lets you use the macros
as you describe.

A further question... It appears that for SIP endpoints, this facility
only updates RPID and PAI headers? I have found that there appear to
be 4 different SIP CID-update mechanisms "out there" as follows:

- Update RPID and PAI (ITSP and trunks often understand this)
- Update Contact: header (Aastra handsets use this)
- A SIP INFO packet if "Supported: callerid" is specified (Older snom
firmware uses this)
- Update From: header if "Supported: from-change" is specified (RFC
4916, snom, Yealink)

Are there existing plans to support any of these other methods? If
not, I will almost certainly add them for my own use, and submit the
code.

Regards,
Steve

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Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

[Kevin P. Fleming]

On 04/25/2012 11:54 AM, Steve Davies wrote:


A further question... It appears that for SIP endpoints, this facility
only updates RPID and PAI headers? I have found that there appear to
be 4 different SIP CID-update mechanisms "out there" as follows:

- Update RPID and PAI (ITSP and trunks often understand this)
- Update Contact: header (Aastra handsets use this)
- A SIP INFO packet if "Supported: callerid" is specified (Older snom
firmware uses this)
- Update From: header if "Supported: from-change" is specified (RFC
4916, snom, Yealink)

Are there existing plans to support any of these other methods? If
not, I will almost certainly add them for my own use, and submit the
code.


No, we have no plans at this time to go beyond RPID and PAI support. 
Those two appear to cover all the current endpoints that we have been 
able to test with, and many community members have also used with other 
endpoints and had success.


Changing the Contact header seems quite wrong; the display-name in a URI 
in the Contact header is pretty much irrelevant. Changing the From 
header also seems wrong; that should indicate who sent the initial 
INVITE, not who redirected it. I don't think we'd want to merge patches 
that added support for either of those mechanisms.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

On 25 April 2012 18:05, Kevin P. Fleming  wrote:
> On 04/25/2012 11:54 AM, Steve Davies wrote:
>
>> A further question... It appears that for SIP endpoints, this facility
>> only updates RPID and PAI headers? I have found that there appear to
>> be 4 different SIP CID-update mechanisms "out there" as follows:
>>
>> - Update RPID and PAI (ITSP and trunks often understand this)
>> - Update Contact: header (Aastra handsets use this)
>> - A SIP INFO packet if "Supported: callerid" is specified (Older snom
>> firmware uses this)
>> - Update From: header if "Supported: from-change" is specified (RFC
>> 4916, snom, Yealink)
>>
>> Are there existing plans to support any of these other methods? If
>> not, I will almost certainly add them for my own use, and submit the
>> code.
>
>
> No, we have no plans at this time to go beyond RPID and PAI support. Those
> two appear to cover all the current endpoints that we have been able to test
> with, and many community members have also used with other endpoints and had
> success.

Thanks for that, I'll have to test further and see whether all the
devices we use support RPID/PAI. It would certainly be easier than
messing about with headers that should not really be changed!

> Changing the Contact header seems quite wrong; the display-name in a URI in
> the Contact header is pretty much irrelevant. Changing the From header also
> seems wrong; that should indicate who sent the initial INVITE, not who
> redirected it. I don't think we'd want to merge patches that added support
> for either of those mechanisms.

The From: header change is a relatively recent RFC, but I've seen
several handsets supporting it, and several non-Asterisk SIP stacks
using this to achieve COLP updates. I completely agree that changing
the Contact: header is daft, and I have no idea why Aastra use this
method.

Regards,
Steve

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[asterisk-users] CONNECTEDLINE and ooh323, do it work?

[Dmitry Melekhov]

Hello!

Just read

http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE

tried on dahdi, it works, i.e. if I call asterisk user from my pbx 
connected phone I see what I set in

Set(CONNECTEDLINE(name)=

But if I call the same user over h323 ( no matter is it asterisk with 
ooh323 or cisco gateway) I don't see this.


Could you tell me is it possible?

Thank you!

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Re: [asterisk-users] CONNECTEDLINE and ooh323, do it work?

[Dmitry Melekhov]

30.10.2013 13:25, Dmitry Melekhov пишет:

Hello!

Just read

http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE

tried on dahdi, it works, i.e. if I call asterisk user from my pbx 
connected phone I see what I set in

Set(CONNECTEDLINE(name)=

But if I call the same user over h323 ( no matter is it asterisk with 
ooh323 or cisco gateway) I don't see this.


Could you tell me is it possible?

Thank you!

OK, I don't know how I get it works on asterisk-asterisk h323 connection 
and transit to pri.

I need this on cisco-asterisk connection with cisco connected to pri too.
I understand that cisco configuration is not topic of this list, but may 
be somebody know what I need to do on cisco to match

asterisk? :-)

Thank you!


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[asterisk-users] CONNECTEDLINE(), progressinband=no and 183 before 180 (with latest trunk)

[crjw]

I don't expect my SIP provider to provide useful "Remote-Party-ID" information.
Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to 
populate remote party information from a local database.
I am also using the "I" (upper case "i") option for Dial.
Generally I like to see to see the remote party name appear on the phone's 
display as soon as a call is dialed... even if the the remote party is BUSY.
I ran into a problem where the Remote-Party-ID does not get displayed on the 
caller's phone until the remote phone is answered.
I finally tracked this down to several things:
a) My SIP provider sends "183 Session Progress" and inband ringback prior to 
sending a "180 Ringing".
b) The default sip.conf file that ships with asterisk suggests using 
"progressinband=no" for polycom phones.
c) The "progessinband=no" setting prevents the "180 Ringing" from being 
forwarded to the phone if it is received after the "183 Session Progress".
d) Called-Parity-ID appears to be only sent to the phone with "180 Ringing" and 
"200 OK" responses.
# this the sequence of events that transpire:
-caller places call
-asterisk receives "183 Trying" from SIP provider and forwards it to the 
caller's phone
-asterisk receives inband ringback from SIP provider and forwards it to the 
phone (RTP)
-asterisk receives "180 Ringing" from SIP provider but does "not" forward it to 
the phone.
-asterisk continues to receive more inband ringback from SIP provider and it 
continues to forward it to the phone (RTP)
-remote party answers the phone
-asterisk receives "200 OK" from SIP provider; asterisk inserts 
"Called-Party-ID" and then forwards it to the calling phone.
-the display on the caller's phone is finally updated; ringback stops and 
someone at the other end says "hello".
There are two workarounds which will make the Called-Party-ID show up on the 
phone before the call is answered:
i) Use "progressinband=never" even though the default sip.conf file recommends 
against it.
The recommendation is presumably based on some old bugs in the Polycom phones 
that no longer exists.
I am using recent Polycom firmware and did not notice any bugs.
Note however that the the display on the phone won't be updated if the remote 
phone is "BUSY", which in my case is not ideal.
ii) Use the "r" option to "Dial". e.g. Dial(SIP/${ext...@x,300,Ir);
This has the advantage of updating the phone very quickly without waiting for 
any respones from the SIP provider.
This may have side effects: ringback could hypothetically be produced when it 
shouldn't be.
Questions:
Is there a reason why "Remote-Party-ID" is not sent to the phone as part of the 
"183 Trying" message?
Could this be a configurable option?
Should the example sip.conf file continue to recommend "progressinband=no" for 
Polycom phones?
-crjw-- 
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