I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060 Unmonitored 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 IP phone. relevant bit of sip.conf: [200] username=200 type=friend secret=1234 port=5060 nat=never dtmfmode=rfc2833 context=default callerid="Angus Comber" <200> host=dynamic disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 [202] username=202 type=friend secret=1234 port=5060 nat=never dtmfmode=rfc2833 context=default callerid="Sam Comber" <202> host=dynamic disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 But whenever I try to dial between phones I get this: Sip read: 0 headers, 0 lines Sip read: INVITE sip:777@192.168.0.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 To: <sip:777@192.168.0.13;user=phone> Contact: <sip:200@192.168.0.3;user=phone> Supported: replaces, timer Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45925 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 258 v=0 o=200 8000 8000 IN IP4 192.168.0.3 s=SIP Call c=IN IP4 192.168.0.3 t=0 0 m=audio 5004 RTP/AVP 18 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 13 headers, 13 lines Using latest request as basis request Sending to 192.168.0.3 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 To: <sip:777@192.168.0.13;user=phone>;tag=as668982be Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45925 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:777@192.168.0.13> Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366" Content-Length: 0 to 192.168.0.3:5060 Scheduling destruction of call '11e4ca07b25c9335@192.168.0.3' in 15000 ms Found user '200' Sip read: ACK sip:777@192.168.0.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 To: <sip:777@192.168.0.13;user=phone>;tag=as668982be Contact: <sip:200@192.168.0.3;user=phone> Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45925 ACK User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:777@192.168.0.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 To: <sip:777@192.168.0.13;user=phone> Contact: <sip:200@192.168.0.3;user=phone> Supported: replaces, timer Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:777@192.168.0.13;user=phone", nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c" Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45926 INVITE User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 258 v=0 o=200 8000 8001 IN IP4 192.168.0.3 s=SIP Call c=IN IP4 192.168.0.3 t=0 0 m=audio 5004 RTP/AVP 18 0 8 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 14 headers, 13 lines Using latest request as basis request Sending to 192.168.0.3 : 5060 (non-NAT) Found user '200' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.3:5004 Found description format G729 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 777 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 To: <sip:777@192.168.0.13;user=phone>;tag=as668982be Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45926 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:777@192.168.0.13> Content-Length: 0 to 192.168.0.3:5060 Sip read: ACK sip:777@192.168.0.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 To: <sip:777@192.168.0.13;user=phone>;tag=as668982be Contact: <sip:200@192.168.0.3;user=phone> Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:777@192.168.0.13;user=phone", nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e" Call-ID: 11e4ca07b25c9335@192.168.0.3 CSeq: 45926 ACK User-Agent: Grandstream GXP2000 1.0.1.9 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 12 headers, 0 lines Destroying call '11e4ca07b25c9335@192.168.0.3' How can I troubleshoot? What should I be looking at? Angus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050724/72898f79/attachment.htm
It appears from the debug that extension 200 is trying to call 777, not 202.
Your Asterisk server can't find an extension 777 and returns "404 not
found". That will explain why you can't call extension 777 from
extension 200. If you want to call extension 202, you will need to dial 202 on
extension 200, not 777.
Regards,
Derek
----- Original Message -----
From: Angus Comber
To: asterisk-users@lists.digium.com
Sent: Sunday, July 24, 2005 11:51 AM
Subject: [Asterisk-Users] Why can't sip/200 call sip/202
I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get:
sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
202/202 192.168.0.6 D 255.255.255.255 5060
Unmonitored
201/201 (Unspecified) D 255.255.255.255 5060
Unmonitored
200/200 192.168.0.3 D 255.255.255.255 5060
Unmonitored
200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 IP phone.
relevant bit of sip.conf:
[200]
username=200
type=friend
secret=1234
port=5060
nat=never
dtmfmode=rfc2833
context=default
callerid="Angus Comber" <200>
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
[202]
username=202
type=friend
secret=1234
port=5060
nat=never
dtmfmode=rfc2833
context=default
callerid="Sam Comber" <202>
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
But whenever I try to dial between phones I get this:
Sip read:
0 headers, 0 lines
Sip read:
INVITE sip:777@192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777@192.168.0.13;user=phone>
Contact: <sip:200@192.168.0.3;user=phone>
Supported: replaces, timer
Call-ID: 11e4ca07b25c9335@192.168.0.3
CSeq: 45925 INVITE
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 258
v=0
o=200 8000 8000 IN IP4 192.168.0.3
s=SIP Call
c=IN IP4 192.168.0.3
t=0 0
m=audio 5004 RTP/AVP 18 0 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777@192.168.0.13;user=phone>;tag=as668982be
Call-ID: 11e4ca07b25c9335@192.168.0.3
CSeq: 45925 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:777@192.168.0.13>
Proxy-Authenticate: Digest realm="asterisk",
nonce="0c555366"
Content-Length: 0
to 192.168.0.3:5060
Scheduling destruction of call '11e4ca07b25c9335@192.168.0.3' in 15000
ms
Found user '200'
Sip read:
ACK sip:777@192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777@192.168.0.13;user=phone>;tag=as668982be
Contact: <sip:200@192.168.0.3;user=phone>
Call-ID: 11e4ca07b25c9335@192.168.0.3
CSeq: 45925 ACK
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
11 headers, 0 lines
Sip read:
INVITE sip:777@192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777@192.168.0.13;user=phone>
Contact: <sip:200@192.168.0.3;user=phone>
Supported: replaces, timer
Proxy-Authorization: Digest username="200",
realm="asterisk", algorithm=MD5,
uri="sip:777@192.168.0.13;user=phone", nonce="0c555366",
response="ee6088fb4e50da5fe412913ae40dd45c"
Call-ID: 11e4ca07b25c9335@192.168.0.3
CSeq: 45926 INVITE
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 258
v=0
o=200 8000 8001 IN IP4 192.168.0.3
s=SIP Call
c=IN IP4 192.168.0.3
t=0 0
m=audio 5004 RTP/AVP 18 0 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
14 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.3 : 5060 (non-NAT)
Found user '200'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.3:5004
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 777 in default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777@192.168.0.13;user=phone>;tag=as668982be
Call-ID: 11e4ca07b25c9335@192.168.0.3
CSeq: 45926 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:777@192.168.0.13>
Content-Length: 0
to 192.168.0.3:5060
Sip read:
ACK sip:777@192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777@192.168.0.13;user=phone>;tag=as668982be
Contact: <sip:200@192.168.0.3;user=phone>
Proxy-Authorization: Digest username="200",
realm="asterisk", algorithm=MD5,
uri="sip:777@192.168.0.13;user=phone", nonce="0c555366",
response="7fcb1024a81b3ea3bcc56baeca4bac3e"
Call-ID: 11e4ca07b25c9335@192.168.0.3
CSeq: 45926 ACK
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
12 headers, 0 lines
Destroying call '11e4ca07b25c9335@192.168.0.3'
How can I troubleshoot? What should I be looking at?
Angus
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> I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: > > sip show peers > Name/username Host Dyn Nat ACL Mask Port Status > 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored > 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored > 200/200 192.168.0.3 D 255.255.255.255 5060 Unmonitored > > 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 IP phone. > > relevant bit of sip.conf: > > [200] > username=200 > type=friend > secret=1234 > port=5060 > nat=never > dtmfmode=rfc2833 > context=default > callerid="Angus Comber" <200> > host=dynamic > disallow=all > allow=ulaw > allow=alaw > allow=g723.1 > allow=g729 > > [202] > username=202 > type=friend > secret=1234 > port=5060 > nat=never > dtmfmode=rfc2833 > context=default > callerid="Sam Comber" <202> > host=dynamic > disallow=all > allow=ulaw > allow=alaw > allow=g723.1 > allow=g729 > > > But whenever I try to dial between phones I get this: > > > Sip read: > > 0 headers, 0 lines > > > Sip read: > INVITE sip:777@192.168.0.13;user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:777@192.168.0.13;user=phone> > Contact: <sip:200@192.168.0.3;user=phone> > Supported: replaces, timer > Call-ID: 11e4ca07b25c9335@192.168.0.3 > CSeq: 45925 INVITE > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=200 8000 8000 IN IP4 192.168.0.3 > s=SIP Call > c=IN IP4 192.168.0.3 > t=0 0 > m=audio 5004 RTP/AVP 18 0 8 101 > a=sendrecv > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > 13 headers, 13 lines > Using latest request as basis request > Sending to 192.168.0.3 : 5060 (non-NAT) > Reliably Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:777@192.168.0.13;user=phone>;tag=as668982be > Call-ID: 11e4ca07b25c9335@192.168.0.3 > CSeq: 45925 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:777@192.168.0.13> > Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366" > Content-Length: 0 > > > to 192.168.0.3:5060 > Scheduling destruction of call '11e4ca07b25c9335@192.168.0.3' in 15000 ms > Found user '200' > > > Sip read: > ACK sip:777@192.168.0.13;user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:777@192.168.0.13;user=phone>;tag=as668982be > Contact: <sip:200@192.168.0.3;user=phone> > Call-ID: 11e4ca07b25c9335@192.168.0.3 > CSeq: 45925 ACK > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Length: 0 > > > 11 headers, 0 lines > > > Sip read: > INVITE sip:777@192.168.0.13;user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:777@192.168.0.13;user=phone> > Contact: <sip:200@192.168.0.3;user=phone> > Supported: replaces, timer > Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5,uri="sip:777@192.168.0.13;user=phone", nonce="0c555366",> response="ee6088fb4e50da5fe412913ae40dd45c" > Call-ID: 11e4ca07b25c9335@192.168.0.3 > CSeq: 45926 INVITE > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=200 8000 8001 IN IP4 192.168.0.3 > s=SIP Call > c=IN IP4 192.168.0.3 > t=0 0 > m=audio 5004 RTP/AVP 18 0 8 101 > a=sendrecv > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > 14 headers, 13 lines > Using latest request as basis request > Sending to 192.168.0.3 : 5060 (non-NAT) > Found user '200' > Found RTP audio format 18 > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 101 > Peer audio RTP is at port 192.168.0.3:5004 > Found description format G729 > Found description format PCMU > Found description format PCMA > Found description format telephone-event > Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0(nothing), combined - 0x10c (ulaw|alaw|g729)> Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) > Looking for 777 in default > Reliably Transmitting (no NAT): > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:777@192.168.0.13;user=phone>;tag=as668982be > Call-ID: 11e4ca07b25c9335@192.168.0.3 > CSeq: 45926 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:777@192.168.0.13> > Content-Length: 0 > > > to 192.168.0.3:5060 > > > Sip read: > ACK sip:777@192.168.0.13;user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > From: "Angus Comber" <sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:777@192.168.0.13;user=phone>;tag=as668982be > Contact: <sip:200@192.168.0.3;user=phone> > Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5,uri="sip:777@192.168.0.13;user=phone", nonce="0c555366",> response="7fcb1024a81b3ea3bcc56baeca4bac3e" > Call-ID: 11e4ca07b25c9335@192.168.0.3 > CSeq: 45926 ACK > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Length: 0 > > > 12 headers, 0 lines > Destroying call '11e4ca07b25c9335@192.168.0.3' > > > How can I troubleshoot? What should I be looking at?In the debug trace shown above, I see: Looking for 777 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found It would appear something is trying to dial "777" in the default context (in extensions.conf), and that extension isn't defined, therefor you are getting a "404 Not Found". Without looking at your extensions.conf contents, can't guess any closer on the problem. Also, until you get your arms around diagnosing problems, I'd suggest starting with a single codec, like: disallow=all allow=ulaw ; allow=alaw ; allow=g723.1 ; allow=g729 When the basics are well understood, then go back and experiment with various codecs. I'm also a strong believer in _not_ using a context name such as "default". Asterisk will frequently try to do something with a specified context and if it fails, fall back to the "default" context without you noticing. Unless you happen to see that in the CLI, you form an opinion that your configuration is working fine, but it really isn't.
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