[Asterisk-Users] Sip Issue

[Bisker, Scott (7805)]

Michael,

Where in your extension definition to you dial a channel (SIP, Zap, or other)?  You are missing the dial entry.

-sb


-----Original Message-----
From: Lists [mailto:lists at uc9.net]
Sent: Saturday, November 29, 2003 10:53 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Sip Issue


Hi all I am having some issues with a gs 100 phone. It is on the same 
network as my * server. There is no firewall.

In extentions.conf
exten => 5,1,Answer
exten => 5,2,MusicOnHold(default)

When I dial 5 from the sip phone
    -- Executing Answer("SIP/mlh-2e75", "") in new stack
    -- Executing MusicOnHold("SIP/mlh-2e75", "default") in new stack
    -- Started music on hold, class 'default', on SIP/mlh-2e75
-----------about 7 secs.......................
    -- Stopped music on hold on SIP/mlh-2e75
    == Spawn extension (sip, 5, 2) exited non-zero on 'SIP/mlh-2e75'


In /var/log/asterisk/messages
Nov 29 23:01:46 WARNING[1142127920]: File chan_sip.c, Line 464 
(retrans_pkt): Maximum retries exceeded on call 
d84c13c5-97cc-a31a-81ba-9426fb234a44 at 66.93.1.239 for seqno 28503 
(Response)


Any Ideas?

Michael

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