[Serusers] Call Transfers in SER + Asterisk architecture

Ricardo - this is an Asterisk issue. Configure your SIP.conf in * creating a sip peer with insecure=very fotr the SER peer, like this: [ser-peer] type = friend ; should you want to receive and make calls to SER context=from-ser ; the context in dialplan with extensions allowed to be accessed by SER. Here you must have PSTN extensions capability. host=200.XXX.XXX.XXX ; the IP address for SER fromdomain=domain.com ;the Domain part of uri to be verified by asterisk on the INVITE received by SER. qualify= yes ; just to check the latency between SER and Asterisk (like this, if over 2000ms Ast will report as unavailable peer). disallow=all allow=alaw allow=g729 insecure= very ; this line garantees that any username part of Request URI sent by SER in INVITE to Asterisk will be accepted by Ast and routed to the dialplan. So, if SER send an INVITE to HYPERLINK "mailto:5531332818847@domain.com.br"5531332818847@domain.com.br , Asterisk will look for a section of type =user in SIP conf to match the user part first, it won't find, then it will look for a type=peer. It will find and try to match the IP address as in host= ...line and will accept any username part as per insecure=very line. Then, if context=from-ser in the Ast dialplan allows this dialing string (553132818847), it will proceed from there. Hope it helps. At. Walter -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.16/552 - Release Date: 26/11/2006 11:30