[Asterisk-Users] canreinvite=no being ignored?

Wed Aug 31 02:19:25 MST 2005

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 at 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 at boxb") in new stack
    -- Called c1#1234 at 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 at boxa-ip-address>;tag=as4124f74a
To: <sip:c1#1234 at boxb-ip-address>
Contact: <sip:2608 at 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>