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I'm trying to call from Avaya to Asterisk (11.5.0) in SIP. Avaya calls to Asterisk, then Asterisk rings my softphone but when I answer the call (in the softphone), call drops immediately.

This is the SIP debug log when calling and answering

Also, something I'm really curious about this screen:

http://i.imgur.com/0zwxAZS.png

When Avaya extension calls me, I can see the display name, but not the domain or user, it just show an ANONYMOUS label.

Finally, if I turn the SIP debug off, when Avaya attemps to call, the CLI log show me this:

*ERROR[10732][C-000000ae]: sip/reqresp_parser.c:830 get_name_and_number: can not parse name and number from sip header.*

This is my SIP.CONF file:

[general]
context=default 
allowoverlap=no 
bindport=5060 
bindaddr = 0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
allow=all
dtmfmode=rfc2833
srvlookup=yes 
accept_outofcall_message=yes
;allowguest=yes
;insecure=very ;please
sendrpid=yes
trustrpid=yes
nat=force
allowguest=yes
ignoreregexpire=yes

[8930]
type=friend
;regexten=8930
context=phones
;secret=8930
;username=8930
;callerid="User1" <8930> 
host=dynamic
;canreinvite=no 
;allow=all
;dtmfmode=rfc2833
;language=es

[8929] 
type=peer
regexten=8929
context=default
bindaddr=5060
secret=8929 
username=8929 
callerid="User2" <8929> 
host=dynamic
canreinvite=no 
allow=all
dtmfmode=rfc2833
language=es

[Avaya2]
type=friend
context=assign
host=10.10.2.220
bindport=5060
allow=all

And this is my EXTENSION.CONF

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
TRUNKMSD=1

[default]

exten => 8930,1,Set(IP=${CUT(CUT(SIP_HEADER(Via), ,2),:,1)})


exten => 8929,1,Dial(SIP/8929)
exten => 8929,2,VoiceMail(8930@vm-demo)


exten => _6XXXX,1,Dial(SIP/${EXTEN}@Avaya2)
exten => _5XXXX,1,Dial(SIP/${EXTEN}@Avaya2)
exten => _4XXXX,1,Dial(SIP/${EXTEN}@Avaya2)
share|improve this question
    
Any chance you can provide an actual packet capture and not just the Asterisk debug? Also, can you tell me a little bit more about the setup? Are these on the same LAN? Is it over a VPN? How is the IP addressing scheme setup on both systems? –  Matt W Jul 29 '13 at 20:35
    
Hello Matt. We are trying to send calls from Avaya to Asterisk in SIP (one way only), the Asterisk and the Avaya servers are on the same LAN and the same network segments, there is no VPN. I think the problem comes when Asterisk is not receiving the anonymous Avaya's SIP protocol. We just figured it out when we install FreePBX and allow the Allow Anonymous Inbound SIP Calls option. But the problem here is, where is that option with out the FreePBX? I'm posting some images of the packet captures I made. i.imgur.com/94Y3g25.png and this one i.imgur.com/XneDgug.png Thank you Matt. –  Cyril Jul 29 '13 at 23:51

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