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 host=141.41.40.232 context=default [authentication] auth=999:xxxxxxxx...@voip.ikt-bs.de <999%3axxxxxxxx...@voip.ikt-bs.de> ---------------------------- -- ------------------------------------------------------------------- Dipl.-Ing. (FH) René Pankratz IANT- APPLIED NGN-TECHNOLOGIES Schlüsselfertige VoIP-Lösungen und mehr... IANT GmbH Salzdahlumer Straße 46/48 D-38302 Wolfenbüttel Fon: +49/(0)5331/ 900989-450 Fax: +49/(0)5331/ 900989-499 Internet: www.iant.de Ust.-IdNr: DE264352710 HRB 201710, Amtsgericht Braunschweig Geschäftsführer: Prof. Dr.-Ing. Diederich Wermser, Dipl.-Ing. Jan Schumacher IANT is Member of GROUPLINK www.grouplink.de
transfer_fail_polycom.pcap
Description: Binary data
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