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Summary:ASTERISK-13873: Getting the wrong peer when INVITE
Reporter:Vic Jolin (vctor)Labels:
Date Opened:2009-04-01 02:19:12Date Closed:2011-06-07 14:03:08
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I have a network with some snom phones and I have xlite. The snom peer is 999100095 and my xlite is 999100078

The problem is when I try to dial out, I get the message like this Call from '999100095' to extension '6043762643' rejected because extension not found.

Now,please don't worry about the extension not found case, that is not the issue. The issue is that asterisk is showing the wrong peer!

I have below the sip debug message

****** ADDITIONAL INFORMATION ******

Reliably Transmitting (NAT) to 119.93.92.11:60446:
OPTIONS sip:999100078@119.93.92.11:60446;rinstance=1583617c90746cfa SIP/2.0
Via: SIP/2.0/UDP 193.67.129.60:5060;branch=z9hG4bK6ea56a19;rport
From: "asterisk" <sip:asterisk@193.67.129.60>;tag=as0791044c
To: <sip:999100078@119.93.92.11:60446;rinstance=1583617c90746cfa>
Contact: <sip:asterisk@193.67.129.60>
Call-ID: 34871cc8150b9f3f079f2d0653b770cc@193.67.129.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 01 Apr 2009 07:12:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
vpbx1*CLI>
<--- SIP read from 119.93.92.11:60446 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.67.129.60:5060;branch=z9hG4bK6ea56a19;rport=5060
Contact: <sip:192.168.0.101:60446>
To: <sip:999100078@119.93.92.11:60446;rinstance=1583617c90746cfa>;tag=77787d78
From: "asterisk"<sip:asterisk@193.67.129.60>;tag=as0791044c
Call-ID: 34871cc8150b9f3f079f2d0653b770cc@193.67.129.60
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '34871cc8150b9f3f079f2d0653b770cc@193.67.129.60' Method: OPTIONS
   -- Got SIP response 603 "Declined (no dialog)" back from 92.68.55.155
   -- Registered SIP '999100793' at 213.34.244.29 port 5069 expires 60
   -- Added extension '999100793' priority 1 to sipregistration
   -- Registered SIP '999100793' at 213.34.244.29 port 5061 expires 60
   -- Added extension '999100793' priority 1 to sipregistration
vpbx1*CLI>
<--- SIP read from 119.93.92.11:60446 --->
INVITE sip:6043762643@193.67.129.60 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:60446;branch=z9hG4bK-d8754z-7229e72a2b02dd7f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:999100078@119.93.92.11:60446>
To: "6043762643"<sip:6043762643@193.67.129.60>
From: "999100078"<sip:999100078@193.67.129.60>;tag=441f5404
Call-ID: YzdhMWY0YjNhOGUzNTliZDAyNzIzM2U0MDY4NDA1YTE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 425

v=0
o=- 7 2 IN IP4 192.168.0.101
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.101
t=0 0
m=audio 23582 RTP/AVP 107 119 100 106 0 105 98 8 3 101
a=alt:1 1 : EySTStq8 OPFRgdzQ 192.168.0.101 23582
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 16 lines) ---
Sending to 119.93.92.11 : 60446 (NAT)
Using INVITE request as basis request - YzdhMWY0YjNhOGUzNTliZDAyNzIzM2U0MDY4NDA1YTE.
Found peer '999100095'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.101:23582
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.101:23582
Looking for 6043762643 in 12connect (domain 193.67.129.60)

<--- Reliably Transmitting (NAT) to 119.93.92.11:60446 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.101:60446;branch=z9hG4bK-d8754z-7229e72a2b02dd7f-1---d8754z-;received=119.93.92.11;rport=60446
From: "999100078"<sip:999100078@193.67.129.60>;tag=441f5404
To: "6043762643"<sip:6043762643@193.67.129.60>;tag=as14ab59b9
Call-ID: YzdhMWY0YjNhOGUzNTliZDAyNzIzM2U0MDY4NDA1YTE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Apr  1 09:12:54] NOTICE[4354]: chan_sip.c:14383 handle_request_invite: Call from '999100095' to extension '6043762643' rejected because extension not found.
Comments:By: Vic Jolin (vctor) 2009-04-06 06:04:51

Anyone who can help me on this?
By: Joshua C. Colp (jcolp) 2009-04-06 09:19:32

Please include your sip.conf configuration minus passwords.
By: Vic Jolin (vctor) 2009-04-06 23:11:06

This is sip.conf general
[general]
context=default
allowoverlap=no
bindport=5060
srvlookup=yes
checkmwi=300
pedantic=yes
nat=yes
qualify=5000
canreinvite=yes
rtcachefriends=yes
rtsavesysname=yes
displaysystemname=yes
rtupdate=yes
rtautoclear=yes
regcontext=sipregistration
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729

I am using Asterisk Realtime
sip_peers table is like this
name=999101305  
host=dynamic  
nat=yes  
type=peer  
accountcode=11046  
amaflags=NULL  
callgroup=NULL  
callerid=201 <31852014911>  
cancallforward=yes  
canreinvite=yes  
context=12connect  
defaultip=NULL  
dtmfmode=auto  
fromuser=NULL  
fromdomain=NULL  
insecure=NULL  
language=NULL  
mailbox=999101305@12connect  
md5secret=NULL
deny=NULL  
permit=NULL  
mask=NULL  
musiconhold=NULL  
pickupgroup=NULL  
qualify=yes  
regexten=NULL  
restrictid=NULL  
rtptimeout=NULL  
rtpholdtimeout=NULL  
secret=********
setvar=NULL  
disallow=all  
allow=alaw;ulaw;gsm;g729    
fullcontact=
ipaddr=0.0.0.0  
port=0  
regserver=connect1  
regseconds=1239035042  
username=999101305
By: Vic Jolin (vctor) 2009-04-20 02:19:37

Im still getting wrong peer on invite, is there in anyway I can prevent this from happening even just in the configuration?
By: Vic Jolin (vctor) 2009-04-22 04:22:22

HI any updates on this? This is creating problems on our setup because the calling phone is different from the real peer
By: Leif Madsen (lmadsen) 2009-06-10 12:48:54

Changed back to new since the reporter has provided the requested information.
By: Mark Michelson (mmichelson) 2009-06-12 15:27:53

I think the problem here is that both of the peers have the same IP address. SIP entries with "type = peer" are matched using IP address, not name. If you change these to be type = user, then that might make your problem disappear.
By: Leif Madsen (lmadsen) 2009-06-16 12:29:29

I agree with mmichelson. This appears to be classic sip.conf matching confusion. I would try what mmichelson suggested, and if you still can't get it to work, you're going to have better support and help by posting your question and issue on the asterisk-users mailing list. Thanks!