SIP/2.0 400 Bad Request - 'Invalid IP Address'
Submitted by ufukguler on Sun, 06/22/2008 - 8:09am.
Hi,
I have trixbox and cisco router. When i call cisco router extension from trixbox or call trixbox extension from cisco router i get SIP/2.0 400 Bad Request - 'Invalid IP Address' error. I am sending trixbox debug output. What should i do to solve problem. Thank you.
Ufuk Guler
advansa*CLI>
-- Executing [2000@from-internal:1] Macro("SIP/5610-0a01cd10", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/5610-0a01cd10", "user-callerid: device 5610") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/5610-0a01cd10", "AMPUSER=5610") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/5610-0a01cd10", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/5610-0a01cd10", "1|Set|REALCALLERIDNUM=5610") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/5610-0a01cd10", "REALCALLERIDNUM is 5610") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/5610-0a01cd10", "AMPUSER=5610") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/5610-0a01cd10", "AMPUSERCIDNAME=Ufuk Guler") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/5610-0a01cd10", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/5610-0a01cd10", "AMPUSERCID=5610") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/5610-0a01cd10", "CALLERID(all)="Ufuk Guler" <5610>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/5610-0a01cd10", "REALCALLERIDNUM=5610") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/5610-0a01cd10", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/5610-0a01cd10", "TTL: ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/5610-0a01cd10", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/5610-0a01cd10", "Using CallerID "Ufuk Guler" <5610>") in new stack
-- Executing [2000@from-internal:2] Set("SIP/5610-0a01cd10", "_NODEST=") in new stack
-- Executing [2000@from-internal:3] Macro("SIP/5610-0a01cd10", "record-enable|5610|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/5610-0a01cd10", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/5610-0a01cd10", "recordingcheck|20080622-180802|1214147282.56") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080622-180802|1214147282.56: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/5610-0a01cd10", "No recording needed") in new stack
-- Executing [2000@from-internal:4] Macro("SIP/5610-0a01cd10", "dialout-trunk|2|2000||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/5610-0a01cd10", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/5610-0a01cd10", "0|Authenticate|") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/5610-0a01cd10", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/5610-0a01cd10", "DIAL_NUMBER=2000") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/5610-0a01cd10", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/5610-0a01cd10", "GROUP()=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/5610-0a01cd10", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/5610-0a01cd10", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/5610-0a01cd10", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/5610-0a01cd10", "outbound-callerid|2") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/5610-0a01cd10", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/5610-0a01cd10", "REALCALLERIDNUM is 5610") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/5610-0a01cd10", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/5610-0a01cd10", "USEROUTCID=5610") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/5610-0a01cd10", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/5610-0a01cd10", "TRUNKOUTCID=Advansa") in new stack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/5610-0a01cd10", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/5610-0a01cd10", "0?usercid") in new stack
-- Executing [s@macro-outbound-callerid:17] Set("SIP/5610-0a01cd10", "CALLERID(all)=Advansa") in new stack
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/5610-0a01cd10", "0?report") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/5610-0a01cd10", "CALLERID(all)=5610") in new stack
-- Executing [s@macro-outbound-callerid:20] GotoIf("SIP/5610-0a01cd10", "1?report:hidecid") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/5610-0a01cd10", "CallerID set to "" <5610>") in new stack
-- Executing [s@macro-dialout-trunk:12] AGI("SIP/5610-0a01cd10", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/5610-0a01cd10", "OUTNUM=2000") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/5610-0a01cd10", "custom=SIP/Advansa_Out") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/5610-0a01cd10", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/5610-0a01cd10", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/5610-0a01cd10", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/5610-0a01cd10", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/5610-0a01cd10", "SIP/Advansa_Out/2000|300|") in new stack
Audio is at 172.16.1.7 port 11224
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 88.255.xx.xxx:5060:
INVITE sip:2000@88.255.xx.xxx SIP/2.0
Via: SIP/2.0/UDP 172.16.1.7:5060;branch=z9hG4bK3746d80b;rport
From: "5610"
To:
Contact:
Call-ID: 4fc5f6c85ed8c9196a4919f53da6034e@172.16.1.7
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 22 Jun 2008 15:08:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 2425 2425 IN IP4 172.16.1.7
s=session
c=IN IP4 172.16.1.7
t=0 0
m=audio 11224 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called Advansa_Out/2000
dvansa*CLI>
<--- SIP read from 88.255.xx.xxx:5060 --->
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 172.16.1.7:5060;branch=z9hG4bK3746d80b;rport
From: "5610"
To:
Date: Sun, 22 Jun 2008 15:08:03 GMT
Call-ID: 4fc5f6c85ed8c9196a4919f53da6034e@172.16.1.7
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 400 "Bad Request - 'Invalid IP Address'" back from 88.255.xx.xxx
Transmitting (no NAT) to 88.255.xx.xxx:5060:
ACK sip:2000@88.255.xx.xxx SIP/2.0
Via: SIP/2.0/UDP 172.16.1.7:5060;branch=z9hG4bK3746d80b;rport
From: "5610"
To:
Contact:
Call-ID: 4fc5f6c85ed8c9196a4919f53da6034e@172.16.1.7
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/Advansa_Out-09fe5e70 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/5610-0a01cd10", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/5610-0a01cd10", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/5610-0a01cd10", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [2000@from-internal:5] Macro("SIP/5610-0a01cd10", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/5610-0a01cd10", "all-circuits-busy-now|noanswer") in new stack
--
Really destroying SIP dialog '4fc5f6c85ed8c9196a4919f53da6034e@172.16.1.7' Method: INVITE
-- Executing [s@macro-outisbusy:2] Playback("SIP/5610-0a01cd10", "pls-try-call-later|noanswer") in new stack
--
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/5610-0a01cd10' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/5610-0a01cd10'
 
The debug output is not
Sun, 06/22/2008 - 11:58am
The debug output is not useful all it shows is that Asterisk can't get to any SIP peers on that machine.
In order to help you we need, versions of trixbox, trixbox trunk or peer config to the cisco. Cisco config and output of 'sho ver' and 'show voice port summary'
Above all please use the code tags (see input format below comment window) to make the output readable.
Scott
--
Scott
aka "Skyking"
 
Asterisk and Cisco Router Info
Sun, 06/22/2008 - 12:51pm
Hi,
I am sending detailed information...Thanks for help...
1 - My Trixbox CE version is 2.6.1.1
2 - Trunk Config is;
[Cisco_Out]
host=88.255.xx.xxx
type=peer
dtmfmode=rfc2833
disallow=all
allow=g729
[Cisco_In]
type=user
context=from-trunk
dtmfmode=rfc2833
disallow=all
allow=g729
3 - Cisco Pots Config is;
dial-peer voice 11 pots
destination-pattern [23]...
no digit-strip
direct-inward-dial
port 1/0:15
forward-digits all
4- Show version output;
Cisco IOS Software, C1700 Software (C1700-IPVOICEK9-M), Version 12.4(10), RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2006 by Cisco Systems, Inc.
Compiled Tue 15-Aug-06 23:46 by prod_rel_team
ROM: System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
HOME uptime is 1 minute
System returned to ROM by reload at 22:34:54 ath Sun Jun 22 2008
System restarted at 22:34:33 ath Sun Jun 22 2008
System image file is "flash:c1700-ipvoicek9-mz.124-10.bin"
This product contains cryptographic features and is subject to United
States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to
export@cisco.com.
Cisco 1760 (MPC860P) processor (revision 0x500) with 89228K/9076K bytes of memory.
Processor board ID FOC08082NBS (3736364615), with hardware revision 0000
MPC860P processor: part number 5, mask 2
1 FastEthernet interface
31 Serial interfaces
1 Channelized E1/PRI port
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
Configuration register is 0x2102
5 - Show voice port summary;
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
============== == ============ ===== ==== ======== ======== ==
1/0:15 01 isdn-voice up dorm none none y
1/0:15 02 isdn-voice up dorm none none y
1/0:15 03 isdn-voice up dorm none none y
1/0:15 04 isdn-voice up dorm none none y
1/0:15 05 isdn-voice up dorm none none y
1/0:15 06 isdn-voice up dorm none none y
1/0:15 07 isdn-voice up dorm none none y
1/0:15 08 isdn-voice up dorm none none y
1/0:15 09 isdn-voice up dorm none none y
1/0:15 10 isdn-voice up dorm none none y
1/0:15 11 isdn-voice up dorm none none y
1/0:15 12 isdn-voice up dorm none none y
1/0:15 13 isdn-voice up dorm none none y
1/0:15 14 isdn-voice up dorm none none y
1/0:15 15 isdn-voice up dorm none none y
1/0:15 17 isdn-voice up dorm none none y
1/0:15 18 isdn-voice up dorm none none y
1/0:15 19 isdn-voice up dorm none none y
1/0:15 20 isdn-voice up dorm none none y
1/0:15 21 isdn-voice up dorm none none y
1/0:15 22 isdn-voice up dorm none none y
1/0:15 23 isdn-voice up dorm none none y
1/0:15 24 isdn-voice up dorm none none y
1/0:15 25 isdn-voice up dorm none none y
1/0:15 26 isdn-voice up dorm none none y
1/0:15 27 isdn-voice up dorm none none y
1/0:15 28 isdn-voice up dorm none none y
1/0:15 29 isdn-voice up dorm none none y
1/0:15 30 isdn-voice up dorm none none y
1/0:15 31 isdn-voice up dorm none none y
 
I am going to have to check
Sun, 06/22/2008 - 1:06pm
I am going to have to check some of my notes.
Just from the start the Asterisk config needs to be type = friend, insecure = very.
I will find my notes, you might as well post your cisco config also (show run).
Is the router with the PRI card in the same network as the trixbox?
Scott
--
Scott
aka "Skyking"
 
Member Since:
2007-06-09