[asterisk-bugs] [JIRA] (ASTERISK-27003) chan_sip doesn't send CONNECTEDLINE info over sip trunk

[Dmitry Melekhov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27003?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237091#comment-237091 ] 

Dmitry Melekhov commented on ASTERISK-27003:
--------------------------------------------

Hello!

Yes, I did something wrong, looks like I forgot to reload dialplan.

If I set both name and number , then everything is fine, i.e. if I use CONNECTEDLINE  on asterisk and set both name and number,
then asterisk passes this info over chan_sip.

So, my bug report is incorrect, everything works as expected if call is terminated on asterisk.

But my real problem is transit call, what I want to have is to pass connected name over sip trunk in following scheme:

avaya pbx---isdn pri--asterisk--chan_sip---asterisk--isdn pri--avaya pbx.

If I use chan_ooh323, then connected name is passed, but looks like chan_sip needs both name and number.

Anyway, avaya sends only name:

[May 23 08:21:39] VERBOSE[22766] chan_dahdi.c: PRI Span: 1 < Display (CS6) (len=14) [ Out of Service ]

I'll attach log file , which contains call through right avaya to ast-lud to asterisk and then to left avaya.

Is it possible to send connected name over chan_sip if number is not provided?

Thank you!

> chan_sip doesn't send CONNECTEDLINE info over sip trunk
> -------------------------------------------------------
>
>                 Key: ASTERISK-27003
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27003
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.13.1
>         Environment: Centos 6 x86-64
>            Reporter: Dmitry Melekhov
>            Assignee: Dmitry Melekhov
>            Severity: Minor
>         Attachments: myDebugLog, myDebugLog-asterisk, myDebugLog-ast-lud
>
>
> I need to translate CONNECTEDLINE info from one asterisk to another and then to ISDN.
> Here is how asterisks are connected in my test environment:
> {noformat}
> asterisk(192.168.22.19)---IP link--ast-lud(192.168.72.254)--ISDN PRI----PBX.
> {noformat}
> As PBX user I place call to ast-lud to following dial-plan
> {noformat}
> exten => 5085,1,Dial(SIP/6000 at asterisk)
> ;exten => 5085,1,Dial(OOH323/6000)
> exten => 5085,n,Hangup
> {noformat}
> asterisk on ast-lud:
> {noformat}
> [asterisk]
> type=friend
> host=192.168.22.19             
> insecure=port,invite
> nat=no                         
> canreinvite=yes        
> disallow=all
> ;allow=all             
> allow=alaw             
> allow=ulaw
> ;allow=g729
> dtmfmode=rfc2833
> context=asterisk
> trustrpid = yes
> sendrpid = yes
> {noformat}
> on asterisk:
> {noformat}
> [ast-lud]
> type=friend
> host=192.168.72.254             
> insecure=port,invite
> nat=no                          
> canreinvite=yes         
> disallow=all
> ;allow=all                    
> allow=alaw                      
> allow=ulaw
> ;allow=g729
> dtmfmode=rfc2833
> context=h323
> trustrpid = yes
> sendrpid = yes
> {noformat}
> And then dialplan is:
> {noformat}
> exten => 6000,1,Set(CHANNEL(language)=ru)
> exten => 6000,n,Set(CONNECTEDLINE(name,i)=Conf. 6000)
> exten => 6000,n,Set(CONNECTEDLINE(pres)=allowed)
> exten => 6000,n,Answer
> exten => 6000,n,Meetme(6000,TL(10800000:60000))
> exten => 6000,n,Hangup
> {noformat}
> But, as I can see from debug CONNECTEDLINE info is not even sent by asterisk- 
> I'll upload debug logs.
> If I use ooh323 to connect asterisks, then I see Conf. 6000 on my phone..



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