[asterisk-users] [asterisk-dev] Grandstream blind transfer issue

[Klaus Darilion]

Haven't you read my email?

1. Wrong list
2. Missing log entries (set debug 4, set verbose 4)

klaus

Max Alex schrieb:
> Hi All,
> Thanks for your reply.
> I got this refer message in asterisk.
> but there is not any active channel of blind transfer.
> ----------------------
> REFER sip:1101 at 192.168.1.25 <mailto:sip%3A1101 at 192.168.1.25> SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
> From: <sip:7500 at 192.168.1.30:5060;transport=udp>;tag=3699e1bcbed17687
> To: "1101" <sip:1101 at 192.168.1.25 
> <mailto:sip%3A1101 at 192.168.1.25>>;tag=as32ed6c48
> Contact: <sip:7500 at 192.168.1.30:5060;transport=udp>
> Supported: replaces, path
> Refer-To: <sip:1631XXXXXXX at 192.168.1.25 
> <mailto:sip%3A1631XXXXXXX at 192.168.1.25>>
> Referred-By: <sip:7500 at 192.168.1.25 <mailto:sip%3A7500 at 192.168.1.25>>
> Call-ID: 4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25 
> <mailto:4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25>
> CSeq: 34526 REFER
> User-Agent: Grandstream BT200 1.1.6.46
> Max-Forwards: 70
> Allow: 
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> Content-Length: 0
> 
> <------------->
> --- (14 headers 0 lines) ---
> Call 4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25 
> <mailto:4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25> got a SIP call 
> transfer from caller: (REFER)!
> SIP transfer to extension 1631XXXXXXX at outgoing by 7500 at 192.168.1.25 
> <mailto:7500 at 192.168.1.25>
> localhost*CLI>
> <--- Transmitting (NAT) to 192.168.1.30:5060 <http://192.168.1.30:5060> --->
> SIP/2.0 202 Accepted
> Via: SIP/2.0/UDP 
> 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30
> From: <sip:7500 at 192.168.1.30:5060;transport=udp>;tag=3699e1bcbed17687
> To: "1101" <sip:1101 at 192.168.1.25 
> <mailto:sip%3A1101 at 192.168.1.25>>;tag=as32ed6c48
> Call-ID: 4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25 
> <mailto:4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25>
> CSeq: 34526 REFER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:1101 at 192.168.1.25 <mailto:sip%3A1101 at 192.168.1.25>>
> Content-Length: 0
> 
> 
> <------------>  
> ----------------------------------------
> Is there any options we need to enable in asterisk or grandstream phone?
> I have already user transfer option 'Tt' in dialplan of this.
> Please provide me some help.
> Thanks in advance!!
> 
> Thanks,
> Max Alex
> Voip Developer
> 
> 
> 
> On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion 
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
> 
>     Max Alex wrote:
>      > Hi All,
>      > I have working asterisk version 1.4.24.
>      > I have a blind transfer issue with grandstream bt200.
> 
>     Does it work with other phones? That means is it a Grandstream isue or a
>     general issue?
> 
>      > I have updated the latest firmware to the phone.
>      > The phone is sending the *refer* to asterisk but asterisk is not
>     able to
>      > connect with the another call
> 
>     Why? some log messages would help us helping you.
> 
>      > that i have checked in sip debug.
>      > I am using transfer button of the grandstream phone.
>      > Can anybody provide help for this issue?
> 
>     Please ask again on the user mailing lists and provide some log messages
> 
>      > Thanks in advance!!
>      >
>      > Thanks,
>      > Max Alex
>      > Voip Developer
>      >
>      >
>      >
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