connecting Asterisk k SIP provider
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Сообщений: 19
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connecting Asterisk k SIP provider
mne nado connectitsea k SIP gate-y , u menea vzeali moi IP servaka asteriska , i dali IP adress gate.
Ya propisal v sip.conf IP adress gate-a i v extensions.conf prefix cherez kotorii vihoju na SIP gate, no u menea nichevo ni poluchaetsea. Kto znaet v cheom problema?
Spasibo
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Откуда: Москва
Сообщений: 3421
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Re: connecting Asterisk k SIP provider
Лог звонка привел бы ради приличия...
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
izvineayusi
koneshno:
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Setting NAT on RTP to 524288
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Setting NAT on VRTP to 524288
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Checking SIP call limits for device 201
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Stopping retransmission on '1828686C-AF46-41E0-9B6A-CC9449F4CEC6@10.0.0.51' of Response 52179: Match Found
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
izvineayusi
koneshno:
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Setting NAT on RTP to 524288
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Setting NAT on VRTP to 524288
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Checking SIP call limits for device 201
Jan 27 01:42:03 DEBUG[25299] chan_sip.c: Stopping retransmission on '1828686C-AF46-41E0-9B6A-CC9449F4CEC6@10.0.0.51' of Response 52179: Match Found
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
a gate eto cisco 5300
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Откуда: Санкт-Петербург
Сообщений: 203
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Re: connecting Asterisk k SIP provider
set verbose 5
set debug 5
sip debug
и еще раз повторить :)
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
:)
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (11 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Proxy-Authenticate: Digest realm="asterisk", nonce="06a5e7eb"
Content-Length: 0
---
Scheduling destruction of call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51' in 15000 ms
Found user '200'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
ia: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
Proxy-Authorization: Digest username="200",realm="asterisk",nonce="06a5e7eb",response="01d4a0ed5ee518828a8df23258591fde",uri="sip:66373250540@asterisk"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
ontent-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (12 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.51:8000
Peer video RTP is at port 10.0.0.51:65535
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x1c000e (gsm|ulaw|alaw|h261|h263|h263p), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 66373250540 in from-internal (domain asterisk)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
:)
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (11 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Proxy-Authenticate: Digest realm="asterisk", nonce="06a5e7eb"
Content-Length: 0
---
Scheduling destruction of call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51' in 15000 ms
Found user '200'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
ia: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
Proxy-Authorization: Digest username="200",realm="asterisk",nonce="06a5e7eb",response="01d4a0ed5ee518828a8df23258591fde",uri="sip:66373250540@asterisk"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
ontent-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (12 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.51:8000
Peer video RTP is at port 10.0.0.51:65535
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x1c000e (gsm|ulaw|alaw|h261|h263|h263p), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 66373250540 in from-internal (domain asterisk)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
:)
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (11 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Proxy-Authenticate: Digest realm="asterisk", nonce="06a5e7eb"
Content-Length: 0
---
Scheduling destruction of call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51' in 15000 ms
Found user '200'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
ia: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
Proxy-Authorization: Digest username="200",realm="asterisk",nonce="06a5e7eb",response="01d4a0ed5ee518828a8df23258591fde",uri="sip:66373250540@asterisk"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
ontent-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (12 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.51:8000
Peer video RTP is at port 10.0.0.51:65535
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x1c000e (gsm|ulaw|alaw|h261|h263|h263p), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 66373250540 in from-internal (domain asterisk)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
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Сообщений: 19
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Re: connecting Asterisk k SIP provider
:)
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (11 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Proxy-Authenticate: Digest realm="asterisk", nonce="06a5e7eb"
Content-Length: 0
---
Scheduling destruction of call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51' in 15000 ms
Found user '200'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bKCB6917DB435146E5B2AD1F25F5DD5D0D
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55058 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
<-- SIP read from 80.97.56.5:5060:
INVITE sip:66373250540@asterisk SIP/2.0
ia: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
Proxy-Authorization: Digest username="200",realm="asterisk",nonce="06a5e7eb",response="01d4a0ed5ee518828a8df23258591fde",uri="sip:66373250540@asterisk"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
ontent-Length: 298
v=0
o=200 10528005 10528027 IN IP4 10.0.0.51
s=X-Lite
c=IN IP4 10.0.0.51
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (12 headers 14 lines)---
Using INVITE request as basis request - 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
Sending to 10.0.0.51 : 5060 (NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.51:8000
Peer video RTP is at port 10.0.0.51:65535
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x1c000e (gsm|ulaw|alaw|h261|h263|h263p), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 66373250540 in from-internal (domain asterisk)
Reliably Transmitting (NAT) to 80.97.56.5:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.51:5060;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E;received=80.97.56.5;rport=5060
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:66373250540@80.97.56.6>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
ACK sip:66373250540@asterisk SIP/2.0
Via: SIP/2.0/UDP 10.0.0.51:5060;rport;branch=z9hG4bK926ACF4E1D0E4532A5291F8411D9049E
From: ion <sip:200@asterisk>;tag=2118044773
To: <sip:66373250540@asterisk>;tag=as2a134fad
Contact: <sip:200@10.0.0.51:5060>
Call-ID: 11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51
CSeq: 55059 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
Destroying call '11297A36-9CCB-40BB-91C3-6725CE554EBE@10.0.0.51'
asterisk1*CLI>
<-- SIP read from 80.97.56.5:5060:
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