It's not an Asterisk configuration problem, it's an Asterisk problem
(I've experienced this before as well). FreeSWITCH has the same issue
currently. I'd report this to Digium but don't expect it to get fixed.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/26/2010 12:54 PM, Rene Pankratz wrote:
Hello list members,
we have successfully connected an asterisk as a gateway to our sipx
installation. This gateway is only used for outbound calls and
everything seems to be working fine.
The only problem we figured out is the attended transfer While blind
transfer works without any problems the attended transfer does not
work. Asterisk sends a "SIP/2.0 481 Call leg/transaction does not
exist" notification to the REFER message of my polycom telephone. With
snom i have exactly the same behaviour.
I know that this might be a problem with the asterisk configuration,
but mayber someone on the list already configured something like this.
I attached a pcap file of the transfer.
Here is my peer configuration in the sip.conf of asterisk:
---------------------------
[voip_block]
type=peer
fromdomain=voip.ikt-bs.de <http://voip.ikt-bs.de>
host=141.41.40.232
context=default
[authentication]
auth=999:[email protected]
<mailto:999%[email protected]>
----------------------------
--
-------------------------------------------------------------------
Dipl.-Ing. (FH) René Pankratz
IANT- APPLIED NGN-TECHNOLOGIES
Schlüsselfertige VoIP-Lösungen und mehr...
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Schumacher
IANT is Member of GROUPLINK
www.grouplink.de <http://www.grouplink.de>
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