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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:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 To: <sip:[EMAIL PROTECTED];user=phone> Contact: <sip:[EMAIL PROTECTED];user=phone> Supported: replaces, timer Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be Call-ID: [EMAIL PROTECTED] CSeq: 45925 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366" Content-Length: 0 Sip read: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be Contact: <sip:[EMAIL PROTECTED];user=phone> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 To: <sip:[EMAIL PROTECTED];user=phone> Contact: <sip:[EMAIL PROTECTED];user=phone> Supported: replaces, timer Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c" Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be Call-ID: [EMAIL PROTECTED] CSeq: 45926 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.3:5060 Sip read: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be Contact: <sip:[EMAIL PROTECTED];user=phone> Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e" Call-ID: [EMAIL PROTECTED] 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 How can I troubleshoot? What should I be looking at? Angus
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