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:[EMAIL PROTECTED])
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/[EMAIL PROTECTED]: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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: "223" <sip:[EMAIL PROTECTED]:5061>;tag=as01e72172
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5061>;tag=as01e72172
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5061>;tag=as01e72172
To: <sip:[EMAIL PROTECTED]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==11"
9 headers, 0 lines
-- Got SIP response 513 "Message too big" back from 192.219.85.57
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a
From: "223" <sip:[EMAIL PROTECTED]:5061>;tag=as01e72172
To: <sip:[EMAIL PROTECTED]>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab
Contact: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
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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