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Hello, I�m trying to get started
with asterisk/SIP so I was trying the demo that is provided in the extensions
config file, but the call isn�t �answered� by my server when I try calling the
number that I registered at my SIP provider. I�ve registered with register =>
John.Doe:MyPass:[EMAIL PROTECTED]/1000 in sip.conf and if I use �sip debug� I
can see the call is coming in but then nothing more happens (see debug output
below). Also get these error
messages: Sip.conf: context=demo type=peer fromuser=MyUser secret=MyPass fromdomain=my-sip-provider context=demo extensions.conf: [demo] ; ; All the stuff in the
demo� ; exten =>
s,1,Wait,1
; Wait a second, just for fun exten =>
s,n,Answer
; Answer the line exten =>
s,n,DigitTimeout,5
; Set Digit Timeout to 5 seconds exten =>
s,n,ResponseTimeout,10 ; Set
Response Timeout to 10 seconds �and so
on� That�s all I have�have I missed
something? Debug output from
call: 192.1.1.1=my
server 0123456789=my number at
SIP-provider 9999999999=the number I�m calling
from 213.132.103.213, 212.112.162.50=my SIP providers
IPs ========================================== Sip
read: INVITE sip:[EMAIL PROTECTED]
SIP/2.0 Record-Route:
<sip:213.132.103.213:5060;transport=UDP;lr=true> Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b Record-Route:
<sip:[EMAIL PROTECTED];ftag=2EBE3E60-1646;lr> Via: SIP/2.0/UDP
212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP
212.112.162.22:5060 From:
<sip:[EMAIL PROTECTED]>;tag=2EBE3E60-1646 To:
<sip:[EMAIL PROTECTED]> Date: Wed, 15 Dec 2004 10:10:11
GMT Call-ID:
[EMAIL PROTECTED] Supported:
timer,100rel Min-SE:
1800 Cisco-Guid:
1458717796-1303908825-2510524757-306778262 User-Agent:
Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL,
ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101
INVITE Max-Forwards:
9 Remote-Party-ID:
<sip:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off Timestamp:
1103105411 Contact:
<sip:[EMAIL PROTECTED]:5060> Expires:
180 Allow-Events:
telephone-event Content-Type:
application/sdp Content-Length:
288 v=0 o=CiscoSystemsSIP-GW-UserAgent 1486
s=SIP
Call c=IN IP4
212.112.162.22 t=0 0 m=audio 16842 RTP/AVP 18 0
101 c=IN IP4
212.112.162.22 a=rtpmap:18
G729/8000 a=fmtp:18
annexb=no a=rtpmap:0
PCMU/8000 a=rtpmap:101
telephone-event/8000 a=fmtp:101
0-16 24 headers, 12
lines Using latest request as basis
request Sending to 213.132.103.213 : 5060
(non-NAT) Found RTP audio format
18 Found RTP audio format
0 Found RTP audio format
101 Peer audio RTP is at port
212.112.162.22:16842 Found description format
G729 Found description format
PCMU Found description format
telephone-event Capabilities: us - 0x8000e
(gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing),
combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1
(g723), peer - 0x1 (g723), combined - 0x1 (g723) Found peer
'wx3.se' Reliably Transmitting (no
NAT): SIP/2.0 407 Proxy Authentication
Required Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b Via: SIP/2.0/UDP
212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP
212.112.162.22:5060 From:
<sip:[EMAIL PROTECTED]>;tag=2EBE3E60-1646 To:
<sip:[EMAIL PROTECTED]>;tag=as3c0db481 Call-ID:
[EMAIL PROTECTED] CSeq: 101
INVITE User-Agent: Asterisk
PBX Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER Contact:
<sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest
realm="asterisk", nonce="59e60c89" Content-Length:
0 to
213.132.103.213:5060 Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000
ms Sip
read: ACK sip:[EMAIL PROTECTED]
SIP/2.0 User-Agent:
sapphire/1.6.2.0253 Max-Forwards:
70 Via: SIP/2.0/UDP
213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b To:
<sip:[EMAIL PROTECTED]>;tag=as3c0db481 From:
<sip:[EMAIL PROTECTED]>;tag=2EBE3E60-1646 Call-ID:
[EMAIL PROTECTED] CSeq: 101
ACK Content-Length:
0 9 headers, 0
lines |
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