|
Hi, I am initiating a SIP call
from Asterisk. After about 10 minutes, I loose audio in both directions but
the call seem to stay up. Can someone please help me understand what is
happening here. Been struggling on this for a while now. This one is
preventing me from fully enjoying my Asterisk installation L Here are the 2 last debug
items from the console. <-- SIP read from 62.123.211.31:5060: INFO sip:[EMAIL PROTECTED] SIP/2.0 t: "STEPHANE RICARD"
<sip:[EMAIL PROTECTED]>;tag=3Das1ea35b0b f:
<sip:[EMAIL PROTECTED]>;tag=3D47270277584177094 i: [EMAIL PROTECTED] CSeq: 76518 INFO v: SIP/2.0/UDP =
62.123.211.31:5060;branch=3Dz9hG4bK566fcdf06c789739901a9d3fdb0cd2f5 Max-Forwards: 18 x-nt-corr-id:
[EMAIL PROTECTED] k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec l: 0 Receiving INFO! Transmitting (no NAT) to 62.123.211.31:5060: SIP/2.0 403 Unauthorized Via: SIP/2.0/UDP =
62.123.211.31:5060;branch=3Dz9hG4bK566fcdf06c789739901a9d3fdb0cd2f5;recei= ved=3D62.123.211.31 From:
<sip:[EMAIL PROTECTED]>;tag=3D47270277584177094 To: "STEPHANE RICARD"
<sip:[EMAIL PROTECTED]>;tag=3Das1ea35b0b Call-ID: [EMAIL PROTECTED] CSeq: 76518 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 X-Asterisk-HangupCause: Thanks in advance. Stephane |
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