[Asterisk-Users] SER & Asterisk & SIP =513 "Message Too Big"

[David Waugh]

Using Asterisk 1.0.9

When I try to make an outgoing call with SIP I get the message " 513 Message
too big" back from SER. Any ideas what I am doing wrong?
Debug below.

SER and Asterisk are running on the same Server
SER is on port 5060
Asterisk is on port 5061

In my extension.conf I have the line

SERADDRESS=192.219.85.57:5060 
in Globals

and am using
exten =>_5XXX,2,Dial(sip/${EXTEN:1}@${SERADDRESS})

to dial out.

Here is the sip debug.

    -- Executing Ringing("H323/ip$192.219.85.57:2680/5746", "") in new stack
    -- Executing Dial("H323/ip$192.219.85.57:2680/5746",
"sip/290 at 192.219.85.57:5060") in new stack
We're at 192.219.85.57 port 13054
Answering/Requesting with root capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:290 at 192.219.85.57 SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: "223" <sip:223 at 192.219.85.57:5061>;tag=as01e72172
To: <sip:290 at 192.219.85.57>
Contact: <sip:223 at 192.219.85.57:5061>
Call-ID: 395c707b65d5166f633441b949d8ba9a at 192.219.85.57
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 25 Jul 2005 14:33:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 30548 30548 IN IP4 192.219.85.57
s=session
c=IN IP4 192.219.85.57
t=0 0
m=audio 13054 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 192.219.85.57:5060
    -- Called 290 at 192.219.85.57:5060


Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: "223" <sip:223 at 192.219.85.57:5061>;tag=as01e72172
To: <sip:290 at 192.219.85.57>
Call-ID: 395c707b65d5166f633441b949d8ba9a at 192.219.85.57
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 192.219.85.57:5060 "Noisy feedback tells:  pid=19732
req_src_ip=192.219.85.57 req_src_port=5061 in_uri=sip:290 at 192.219.85.57
out_uri=sip:290 at 192.219.85.57 via_cnt==1"


9 headers, 0 lines


Sip read:
SIP/2.0 513 Message too big
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: "223" <sip:223 at 192.219.85.57:5061>;tag=as01e72172
To: <sip:290 at 192.219.85.57>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab
Call-ID: 395c707b65d5166f633441b949d8ba9a at 192.219.85.57
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 192.219.85.57:5060 "Noisy feedback tells:  pid=19732
req_src_ip=192.219.85.57 req_src_port=5060 in_uri=sip:290 at 192.219.85.57
out_uri=sip:290 at 192.219.85.57 via_cnt==11"


9 headers, 0 lines
    -- Got SIP response 513 "Message too big" back from 192.219.85.57
Transmitting:
ACK sip:290 at 192.219.85.57 SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: "223" <sip:223 at 192.219.85.57:5061>;tag=as01e72172
To: <sip:290 at 192.219.85.57>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab
Contact: <sip:223 at 192.219.85.57:5061>
Call-ID: 395c707b65d5166f633441b949d8ba9a at 192.219.85.57
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.219.85.57:5060
  == No one is available to answer at this time

Incoming calls from a soft SIP phone to SER and then through to asterisk
work fine.


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