Hi All,

I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk replied to all the INVITE's it received before it says Ringing. Really need help on this badly, anyone has an idea. Thank you in advance.

Regards
Ron


U 172.16.0.6:5068 -> 172.16.0.1:5060
INVITE sip:1234...@172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: ""<sip:unkn...@172.16.0.1>;tag=57677d009236 ed33..To: <sip:1234...@172.16.0.1>..Contact: <sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0. 6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 1.3.4.9..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 306....v=0..o=system 8003 8000 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 172.16.0.6.. t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap :3 GSM/8000..a=ptime:20..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..
#
U 172.16.0.6:5068 -> 172.16.0.1:5060
INVITE sip:1234...@172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: ""<sip:unkn...@172.16.0.1>;tag=57677d009236 ed33..To: <sip:1234...@172.16.0.1>..Contact: <sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0. 6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 1.3.4.9..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 306....v=0..o=system 8003 8001 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 172.16.0.6.. t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap :3 GSM/8000..a=ptime:20..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..
#
U 172.16.0.6:5068 -> 172.16.0.1:5060
INVITE sip:1234...@172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: ""<sip:unkn...@172.16.0.1>;tag=57677d009236 ed33..To: <sip:1234...@172.16.0.1>..Contact: <sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0. 6..CSeq: 28907 INVITE..User-Agent: Grandstream GXW4108 (HW 1.1, Ch:3) 1.3.4.9..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,S UBSCRIBE,UPDATE,PRACK..Content-Type: application/sdp..Content-Length: 306....v=0..o=system 8003 8002 IN IP4 172.16.0.6..s=SIP Call..c=IN IP4 172.16.0.6.. t=0 0..m=audio 5016 RTP/AVP 0 8 18 4 3 101..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap :3 GSM/8000..a=ptime:20..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..
#



==============================================================


U 172.16.0.1:5060 -> 172.16.0.6:5068
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: ""<sip:unkn...@172.16.0.1>;tag=57677d00923 6ed33..To: <sip:1234...@172.16.0.1>..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, A CK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Contact: <sip:1234...@172.16.0.1>..Content-Length: 0....
#
U 172.16.0.1:5060 -> 172.16.0.6:5068
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: ""<sip:unkn...@172.16.0.1>;tag=57677d00923 6ed33..To: <sip:1234...@172.16.0.1>..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, A CK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Contact: <sip:1234...@172.16.0.1>..Content-Length: 0....
#
U 172.16.0.1:5060 -> 172.16.0.6:5068
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: ""<sip:unkn...@172.16.0.1>;tag=57677d00923 6ed33..To: <sip:1234...@172.16.0.1>..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, A CK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Contact: <sip:1234...@172.16.0.1>..Content-Length: 0....
#
U 172.16.0.1:5060 -> 172.16.0.6:5068
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2;received=172.16.0.6..From: ""<sip:unkn...@172.16.0.1>;tag=57677d0092 36ed33..To: <sip:1234...@172.16.0.1>;tag=as69addf84..Call-ID: 02075d60f895e8264904b3133107a...@172.16.0.6..cseq: 28907 INVITE..User-Agent: Asterisk PBX.. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Contact: <sip:1234...@172.16.0.1>..Cont

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to