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Hello All,
I have gone thru all the resources I could find on
google on asterisk + iconnect and managed to get outgoing calls working.
However,
I cannot get incoming calls to work at all. With
the sip debug on, I can see that something is happening everytime a call is
received
from iconnecthere, but I get an invalid tone on the
caller side. The call never rings anywhere on the asterisk. Would appreciate any
help on this. Thanks
Below is my sip file
register=442087926805:[EMAIL PROTECTED]:5060
[iconnecthere] type=friend secret=somepassword username=11232634 host=sipauth.deltathree.com canreinvite=no ;nat=yes context=default ;dtmfmode=inband disallow=all ;allow=all allow=gsm allow=ulaw allow=alaw allow=g726 allow=g723 This is the sip debug info when a call comes in
from iconnecthere :
11 headers, 0 lines
Reliably Transmitting: REGISTER sip:sipauth.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9 From: <sip:[EMAIL PROTECTED]>;tag=as5c70755c To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:[EMAIL PROTECTED]> Event: registration Content-Length: 0 (no NAT) to
213.137.73.140:5060
localhost*CLI> Sip read:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9 To: <sip:[EMAIL PROTECTED]> From: <sip:[EMAIL PROTECTED]>;tag=as5c70755c Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER Content-Length: 0 7 headers, 0 lines localhost*CLI> Sip read:
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9 From: <sip:[EMAIL PROTECTED]>;tag=as5c70755c To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER Contact: <sip:[EMAIL PROTECTED] :5060>;expires=120 Contact: <sip:[EMAIL PROTECTED] :5060>;expires=14 Expires: 120 Content-Length: 0 Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173 Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4- 1 Via: SIP/2.0/UDP 213.137.81.27:5060;received=213.137.81.27 To: <sip:[EMAIL PROTECTED]> From: <sip:[EMAIL PROTECTED]>;tag=DF81964C-1341 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Contact: <sip:[EMAIL PROTECTED]:5060> Record-Route: <sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.173> Record-Route: <sip:[EMAIL PROTECTED] .27:5060;maddr=213.137.73.176> Content-Type: application/sdp Content-Length: 146 v=0
o=CiscoSystemsSIP-GW-UserAgent 5851 2446 IN IP4 213.137.81.27 s=SIP Call c=IN IP4 213.137.81.27 t=0 0 m=audio 18958 RTP/AVP 4 0 8 2 101 13 headers, 6 lines
Using latest request as basis request Sending to 213.137.73.140 : 5060 (non-NAT) Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 213.137.81.27:18958 Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x1d(G723|ULAW|ALAW |G726)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'iconnecthere' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173 Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4- 1 Via: SIP/2.0/UDP 213.137.81.27:5060;received=213.137.81.27 From: <sip:[EMAIL PROTECTED]>;tag=DF81964C-1341 To: <sip:[EMAIL PROTECTED]>;tag=as34968f1d 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="252c7e0a" Content-Length: 0 Sip read:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173 Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4- 1 From: <sip:[EMAIL PROTECTED]>;tag=DF81964C-1341 To: <sip:[EMAIL PROTECTED]>;tag=as34968f1d Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Content-Length: 0 8 headers, 0 lines localhost*CLI> Sip read:
REGISTER sip:192.168.1.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca From: <sip:[EMAIL PROTECTED]>;tag=d96a1d9a3a8eb4af To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 402 REGISTER Expires: 120 User-Agent: Grandstream BT100 1.0.4.67 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.1.60 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca From: <sip:[EMAIL PROTECTED]>;tag=d96a1d9a3a8eb4af To: <sip:[EMAIL PROTECTED]>;tag=as5926604e Call-ID: [EMAIL PROTECTED] CSeq: 402 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.1.60:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca From: <sip:[EMAIL PROTECTED]>;tag=d96a1d9a3a8eb4af To: <sip:[EMAIL PROTECTED]>;tag=as5926604e Call-ID: [EMAIL PROTECTED] CSeq: 402 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="007140b3" Content-Length: 0 Sip read:
REGISTER sip:192.168.1.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK9356e31bb2078147 From: <sip:[EMAIL PROTECTED]>;tag=d96a1d9a3a8eb4af To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Authorization: DIGEST username="ext100", realm="asterisk", algorithm=MD5, uri="s ip:192.168.1.250", nonce="007140b3", response="5d56be19a6b63ed92390724df782f89a" Call-ID: [EMAIL PROTECTED] CSeq: 403 REGISTER Expires: 120 User-Agent: Grandstream BT100 1.0.4.67 ax-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 13 headers, 0 lines Using latest request as basis request Sending to 192.168.1.60 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK9356e31bb2078147 From: <sip:[EMAIL PROTECTED]>;tag=d96a1d9a3a8eb4af To: <sip:[EMAIL PROTECTED]>;tag=as5926604e Call-ID: [EMAIL PROTECTED] CSeq: 403 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.1.60:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK9356e31bb2078147 From: <sip:[EMAIL PROTECTED]>;tag=d96a1d9a3a8eb4af To: <sip:[EMAIL PROTECTED]>;tag=as5926604e Call-ID: [EMAIL PROTECTED] CSeq: 403 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:[EMAIL PROTECTED]>;expires=120 Date: Sun, 05 Sep 2004 17:13:34 GMT Content-Length: 0 to 192.168.1.60:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK047cfd59 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as4757cd3d To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no
Voicemail: 0/0 (no NAT) to 192.168.1.60:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms localhost*CLI> Sip read:
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK047cfd59 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as4757cd3d To: <sip:[EMAIL PROTECTED]>;tag=bcd972b14ed2943b Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Grandstream BT100 1.0.4.67 Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 10 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' |
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