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:1234567 at 172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From:
""<sip:unknown at 172.16.0.1>;tag=57677d009236
ed33..To: <sip:1234567 at 172.16.0.1>..Contact:
<sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:1234567 at 172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From:
""<sip:unknown at 172.16.0.1>;tag=57677d009236
ed33..To: <sip:1234567 at 172.16.0.1>..Contact:
<sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:1234567 at 172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP
172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From:
""<sip:unknown at 172.16.0.1>;tag=57677d009236
ed33..To: <sip:1234567 at 172.16.0.1>..Contact:
<sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:unknown at 172.16.0.1>;tag=57677d00923
6ed33..To: <sip:1234567 at 172.16.0.1>..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:1234567 at 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:unknown at 172.16.0.1>;tag=57677d00923
6ed33..To: <sip:1234567 at 172.16.0.1>..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:1234567 at 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:unknown at 172.16.0.1>;tag=57677d00923
6ed33..To: <sip:1234567 at 172.16.0.1>..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:1234567 at 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:unknown at 172.16.0.1>;tag=57677d0092
36ed33..To: <sip:1234567 at 172.16.0.1>;tag=as69addf84..Call-ID:
02075d60f895e8264904b3133107aa47 at 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:1234567 at 172.16.0.1>..Cont
