Auto Call forwarding not working with Audiocodes MP114

johnny
Posts: 313
Member Since:
2006-08-29

Does anybody encounter similar problems with Audiocodes gateway MP114 (2 FXO & 2 FXS)?

Incoming call ==> PSTN Line 1 ==> Audiocodes FXO3 ==> SIP Exten 208 ==> Auto forwarding to (XXXXXX external handphone number) ==> dial to Audiocodes FXO4 ==> PSTN Line2.

The above auto forwarding is working for TDM cards with 2 FXO ports but not for Audiocodes. Below are the syslog errors I received from Audiocodes:

Dec 16 11:28:44 192.168.15.166 ( sip_stack)(965 ) AcSIPParser: Problem in SIP Message Headers
Dec 16 11:28:44 192.168.15.166 ( sip_stack)(965 ) AcSIPParser: Problem in SIP Message Headers
Dec 16 11:28:44 192.168.15.166 ( sip_stack)(966 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol ';'. Expected ' Dec 16 11:28:44 192.168.15.166 ( sip_stack)(966 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol ';'. Expected ' Dec 16 11:28:44 192.168.15.166 ( sip_stack)(967 ) !! [ERROR] Message type: INVITE
Dec 16 11:28:44 192.168.15.166 ( sip_stack)(967 ) !! [ERROR] Message type: INVITE
Dec 16 11:28:44 192.168.15.166 ( sip_stack)(968 ) !! [ERROR] Source header:
Dec 16 11:28:44 192.168.15.166 ( sip_stack)(969 ) !! [ERROR] Line: 12. Column: 13

Really have no idea what does it mean........

thanks in advance.
johnny

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Johnny,
Pan Tone Communications. (Pte) Ltd.
www.pan-tone.com.sg



johnny
Posts: 313
Member Since:
2006-08-29
by the way two separately

by the way two separately concurrent incoming / outgoing calls are working fine, but not for auto forwarding......

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Johnny,
Pan Tone Communications. (Pte) Ltd.
www.pan-tone.com.sg



joshpatten
Posts: 733
Member Since:
2007-01-20
It appears that (not

It appears that (not surprisingly, I suppose) Asterisk is sending a malformed SIP message to the Audiocodes unit. We would have to see the SIP messages in order to go any further. One way to do this is with a hub (not a switch) in between your network and the Audiocodes, and a computer running wireshark plugged into the hub to capture all packets. You could probably debug the SIP stack on Asterisk and get the same information as well. Your choice.



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