Am I reading the data below incorrectly, or does it appear that even
though I have the directive canreinvite=no set for the two asterisk
boxes, they are trying to do a reinvite (which fails) anyway?
Is this expected behaviour in this situation? If so, how can I prevent
this?
---- Lots of output ----
Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A)
has a sip ua (2608) attached which is generating a call, the other
machine (box B) has the final destination.
Sip config for the phone on box A (via Realtime):
pbx3*CLI> sip show peer 2608
* Name : 2608
Secret : <Set>
MD5Secret : <Not set>
Context : assigned-device
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : Yes
Callerid : "" <>
Expire : 386
Expiry : 900
Insecure : no
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.10.32 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 2608
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : OK (16 ms)
Useragent : Sipura/SPA841-0.9.1
Reg. Contact : sip:2608@192.168.10.32:5060
Sip config for Box B on box A:
pbx3*CLI> sip show peer pbx1
* Name : boxb
Secret : <Not set>
MD5Secret : <Not set>
Context : inter-system-inbound-main
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Expiry : 900
Insecure : no
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : <cut for public display>
Addr->IP : <cut for public display> Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : Unmonitored
Useragent :
Reg. Contact :
Sip config for Box A on Box B
pbx1*CLI> sip show peer pbx3
pbx1*CLI>
* Name : boxa
Secret : <Not set>
MD5Secret : <Not set>
Context : inter-system-inbound-main
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Expiry : 900
Insecure : no
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : <cut for public display>
Addr->IP : <cut for public display> Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username:
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : Unmonitored
Useragent :
Reg. Contact :
Dial command as appears on boxa
-- Executing Dial("SIP/2608-8049", "SIP/c1#1234@boxb")
in new stack
-- Called c1#1234@boxb
Aug 31 02:01:29 NOTICE[10496]: chan_sip.c:9028 handle_response: Failed
to authenticate on INVITE to '"2608"
<sip:2608@<boxa-ip-here>>;tag=as4124f74a'
-- SIP/boxb-ae96 is circuit-busy
SIP Debug as it appears on boxb from the call above
<-- SIP read from <boxa-ip-address>:5060:
INVITE sip:c1#1234@<boxb-ip-address> SIP/2.0
Via: SIP/2.0/UDP <boxa-ip-address>:5060;branch=z9hG4bK1d216175;rport
From: "2608" <sip:2608@boxa-ip-address>;tag=as4124f74a
To: <sip:c1#1234@boxb-ip-address>
Contact: <sip:2608@boxa-ip-address>
Call-ID: 3f1250096c1a12b0259689006888f106@<boxb-ip-address>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 09:01:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 214
v=0
=root 10496 10496 IN IP4 <boxa-ip-address>
s=session
c=IN IP4 <boxa-ip-address>
t=0 0
m=audio 19014 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (12 headers 10 lines)---
Using INVITE request as basis request -
3f1250096c1a12b0259689006888f106@<boxa-ip-address>
Sending to <boxa-ip-address> : 5060 (non-NAT)
Reliably Transmitting (NAT) to <boxa-ip-address>:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
<boxa-ip-address>:5060;branch=z9hG4bK1d216175;received=<boxa-ip-address>;rport=5060
From: "2608" <sip:2608@<boxa-ip-address>>;tag=as4124f74a
To: <sip:c1#1234@<boxb-ip-address>>;tag=as2ac1a098
Call-ID: 3f1250096c1a12b0259689006888f106@<boxa-ip-address>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:c1#1234@<boxb-ip-address>>
Proxy-Authenticate: Digest realm="asterisk",
nonce="382038a4"
Content-Length: 0
---
Scheduling destruction of call
'3f1250096c1a12b0259689006888f106@<boxa-ip-address>' in 15000 ms
Found user '2608'
pbx1*CLI>
<-- SIP read from <boxa-ip-address>:5060:
ACK sip:c1#1234@<boxb-ip-address> SIP/2.0
Via: SIP/2.0/UDP <boxa-ip-address>:5060;branch=z9hG4bK1d216175;rport
From: "2608" <sip:2608@<boxa-ip-address>>;tag=as4124f74a
To: <sip:c1#1234@<boxb-ip-address>>;tag=as2ac1a098
Contact: <sip:2608@<boxa-ip-address>>
Call-ID: 3f1250096c1a12b0259689006888f106@<boxa-ip-address>
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call
'3f1250096c1a12b0259689006888f106@<boxa-ip-address>'
pbx1*CLI>
