thr3ads.net - asterisk users - [Asterisk-Users] Calls arent handled by asterisk

If this information is useful, please help other people find it:
Share via:

Göran Törnqvist

2004-Dec-15 06:15 UTC

[Asterisk-Users] Calls arent handled by asterisk - destruction of call

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:MyUser@my-sip-provider/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:

Scheduling destruction of call
'397F980B-4DCF11D9-9E359155-12491096@my-sip-providers-ip' in 15000 ms


WARNING[4863]: chan_sip.c:706 retrans_pkt: Maximum retries exceeded on call
221201580f4bd104062df83a2437e145@MY-IP for seqno 102 (Non-critical Request)

 

Sip.conf:

[general]

context=demo

 

[my-sip-provider]

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:s@192.1.1.1 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:0123456789@212.112.162.50;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:9999999999@212.112.162.22>;tag=2EBE3E60-1646

To: <sip:0123456789@212.112.162.50>

Date: Wed, 15 Dec 2004 10:10:11 GMT

Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22

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:9999999999@212.112.162.22>;party=calling;screen=yes;privacy=off

Timestamp: 1103105411

Contact: <sip:9999999999@212.112.162.22:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 288

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22

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:9999999999@212.112.162.22>;tag=2EBE3E60-1646

To: <sip:0123456789@212.112.162.50>;tag=as3c0db481

Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:s@192.1.1.1>

Proxy-Authenticate: Digest realm="asterisk",
nonce="59e60c89"

Content-Length: 0

 

 

 to 213.132.103.213:5060

Scheduling destruction of call
'56F68A14-4DB811D9-95A69155-12491096@212.112.162.22' in 15000 ms

 

Sip read:

ACK sip:s@192.1.1.1 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:0123456789@212.112.162.50>;tag=as3c0db481

From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646

Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22

CSeq: 101 ACK

Content-Length: 0

 

 

9 headers, 0 lines

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041217/fb6ef7f2/attachment.htm

test

2004-Dec-16 01:59 UTC

head link

[Asterisk-Users] Calls arent handled by asterisk - destruction of call

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:MyUser@my-sip-provider/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:
Scheduling destruction of call
'397F980B-4DCF11D9-9E359155-12491096@my-sip-providers-ip' in 15000 ms

WARNING[4863]: chan_sip.c:706 retrans_pkt: Maximum retries exceeded on call
221201580f4bd104062df83a2437e145@MY-IP for seqno 102 (Non-critical Request)

Can you guys help me? Thanks :) 

Sip.conf:

[general]

context=demo

 [my-sip-provider]

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:s@192.1.1.1 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:0123456789@212.112.162.50;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:9999999999@212.112.162.22>;tag=2EBE3E60-1646

To: <sip:0123456789@212.112.162.50>

Date: Wed, 15 Dec 2004 10:10:11 GMT

Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22

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:9999999999@212.112.162.22>;party=calling;screen=yes;privacy=off

Timestamp: 1103105411

Contact: <sip:9999999999@212.112.162.22:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 288

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22

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:9999999999@212.112.162.22>;tag=2EBE3E60-1646

To: <sip:0123456789@212.112.162.50>;tag=as3c0db481

Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:s@192.1.1.1>

Proxy-Authenticate: Digest realm="asterisk",
nonce="59e60c89"

Content-Length: 0

 

 

 to 213.132.103.213:5060

Scheduling destruction of call
'56F68A14-4DB811D9-95A69155-12491096@212.112.162.22' in 15000 ms

 

Sip read:

ACK sip:s@192.1.1.1 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:0123456789@212.112.162.50>;tag=as3c0db481

From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646

Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22

CSeq: 101 ACK

Content-Length: 0

 

 

9 headers, 0 lines

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041216/2ea77417/attachment.htm

Maybe Matching Threads

Search for more possibly parallel threads

asterisk users - Dec 2004 - Calls arent handled by asterisk - destruction of call

[Asterisk-Users] Calls arent handled by asterisk - destruction of call

[Asterisk-Users] Calls arent handled by asterisk - destruction of call

Maybe Matching Threads