[OpenSIPS-Users] 17 sec, recieve a bye and a hangup

Tue Oct 6 15:48:09 CEST 2009

I understand you can find it under this text.
as you can see, the call just disapeare, i see now that the bye appears when
i hang up the polycom phone.

I hope this information helps.

U 172.16.0.12:5060 -> 172.16.1.10:5090
INVITE sip:0624469780 at 172.16.1.10:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>.
CSeq: 1 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 247.
.
v=0.
o=- 1254823487 1254823487 IN IP4 172.16.0.12.
s=Polycom IP Phone.
c=IN IP4 172.16.0.12.
t=0 0.
m=audio 2222 RTP/AVP 0 8 18 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To:
<sip:0624469780 at 172.16.1.10;user=phone>;tag=67747a5e755302d1b99e6b9647717b58.aaf5.
CSeq: 1 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Proxy-Authenticate: Digest realm="172.16.1.10",
nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5".
Server: OpenSIPS (1.5.3-notls (i386/linux)).
Content-Length: 0.
.

U 172.16.0.12:5060 -> 172.16.1.10:5090
ACK sip:0624469780 at 172.16.1.10:5090 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To:
<sip:0624469780 at 172.16.1.10;user=phone>;tag=67747a5e755302d1b99e6b9647717b58.aaf5.
CSeq: 1 ACK.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Max-Forwards: 70.
Content-Length: 0.
.

U 172.16.0.12:5060 -> 172.16.1.10:5090
INVITE sip:0624469780 at 172.16.1.10:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>.
CSeq: 2 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Proxy-Authorization: Digest username="701", realm="172.16.1.10",
nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5",
uri="sip:0624469780 at 172.16.1.10:5090;user=phone",
response="7c019127f962f84b63ec55b27daac6e2", algorithm=MD5.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 247.
.
v=0.
o=- 1254823487 1254823487 IN IP4 172.16.0.12.
s=Polycom IP Phone.
c=IN IP4 172.16.0.12.
t=0 0.
m=audio 2222 RTP/AVP 0 8 18 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>.
CSeq: 2 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Server: OpenSIPS (1.5.3-notls (i386/linux)).
Content-Length: 0.
.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5484 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.0.12:5060 -> 172.16.1.10:5090
ACK sip:0624469780 at 172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK1b9cbb48F2BE247D.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Route: <sip:172.16.1.10:5090;lr=on>.
CSeq: 2 ACK.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Proxy-Authorization: Digest username="701", realm="172.16.1.10",
nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5",
uri="sip:0624469780 at 172.16.1.10:5090;user=phone",
response="7c019127f962f84b63ec55b27daac6e2", algorithm=MD5.
Max-Forwards: 70.
Content-Length: 0.
.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.0.12:5060 -> 172.16.1.10:5090
BYE sip:0624469780 at 172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Route: <sip:172.16.1.10:5090;lr=on>.
CSeq: 3 BYE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Proxy-Authorization: Digest username="701", realm="172.16.1.10",
nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5",
uri="sip:0624469780 at 172.16.1.10:5090;user=phone",
response="9f5e7c543f689494d444f0402a1eca13", algorithm=MD5.
Max-Forwards: 70.
Content-Length: 0.
.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 404 Not here.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
CSeq: 3 BYE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Server: OpenSIPS (1.5.3-notls (i386/linux)).
Content-Length: 0.
.

RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.0.12:5060 -> 172.16.1.10:5090
ACK sip:0624469780 at 172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK1b9cbb48F2BE247D.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Route: <sip:172.16.1.10:5090;lr=on>.
CSeq: 2 ACK.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Proxy-Authorization: Digest username="701", realm="172.16.1.10",
nonce="4acb4a6f000000003752ab36118b66f10da90b7ccd55e7e5",
uri="sip:0624469780 at 172.16.1.10:5090;user=phone",
response="7c019127f962f84b63ec55b27daac6e2", algorithm=MD5.
Max-Forwards: 70.
Content-Length: 0.
.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.1.10:5090 -> 172.16.0.12:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK9d585e9bF0A1CE7C.
Record-Route: <sip:172.16.1.10:5090;lr=on>.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
CSeq: 2 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:0624469780 at 172.16.1.10>.
Content-Type: application/sdp.
Content-Length: 260.
.
v=0.
o=root 5484 5485 IN IP4 172.16.1.10.
s=session.
c=IN IP4 172.16.1.10.
t=0 0.
m=audio 17896 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

U 172.16.0.12:5060 -> 172.16.1.10:5090
BYE sip:0624469780 at 172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893.
From: "701" <sip:701 at 172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780 at 172.16.1.10;user=phone>;tag=as431f0138.
Route: <sip:172.16.1.10:5090;lr=on>.
CSeq: 3 BYE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0 at 172.16.0.12.
Contact: <sip:701 at 172.16.0.12>.
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049.
Proxy-Authorization: Digest username="701", realm="172.16.1.10",
nonce="4acb4a6f00000000

Brett Nemeroff wrote:
> 
> A trace of the whole call setup to hangup would be very helpful
> 
> On Tue, Oct 6, 2009 at 8:31 AM, Peter den Hartog
> <peterdenhartog at gmail.com>wrote:
> 
>>
>> Thanks alot for you reply.
>>
>> Asterisk is used because we have some agi stuff happening on incomming
>> calls. The sip trunk is registered on Asterisk. If i dial out, opensips
>> uses
>> Asterisk because the extention is not in opensips (if i understand it
>> correctly) then Asterisk just uses his own sip trunk to dial outside.
>>
>> But for me it would be fine to use Opensips directly to make the
>> connection
>> with the sip trunk, we  can leave asterisk out for now.
>>
>> 1. There is two way audio, i can hear the other person talking, and he
>> can
>> hear me 2.
>> 2. no reinvite, i see a ok, and then a bye
>> 3. i don't know this yet, i can test it, i think i saw a empty ACK
>>
>>
>>
>> Brett Nemeroff wrote:
>> >
>> > I guess the question here is, what is asterisk doing for you? I
>> personally
>> > would prefer the sip trunks right on opensips.. Asterisk is a kinda
>> funny
>> > bottleneck in your architecture unless it's acting as some sort of
>> media
>> > server (or TDM gateway).
>> > Some potential issues:
>> > 1. Do you have 2 way audio, some providers (gateways) will disco the
>> call
>> > if
>> > there is one way audio for X seconds.
>> > 2. Do you see any reinvites happening? Some providers will re-invite
>> calls
>> > after they are up and if the reinvite fails, it will tear down the
>> call.
>> > 3. Where is the BYE coming from? Do you see any other signaling after
>> the
>> > 200OK/ACKs you get? Do you see retransmissions of either the 200OK or
>> ACK?
>> > If the signaling indicating the call was connected doesn't finish a
>> proper
>> > ACK in both directions, the call will likely get hung up on.
>> >
>> >
>> > On Tue, Oct 6, 2009 at 8:17 AM, Peter den Hartog
>> > <peterdenhartog at gmail.com>wrote:
>> >
>> >>
>> >> I'm trying to intergrate opensips with a allready running Asterisk
>> >> server.
>> >> The two servers are both on the same machine.
>> >>
>> >> I can recieve calls fine, Asterisk send them to my opensips
>> installation,
>> >> and the opensips forwards the phone call to the right user. I can call
>> >> between the users on the network, with out any issue's so far so good.
>> >>
>> >> I have a sip trunk registered on Asterisk, and i use that for my in
>> and
>> >> outgoing calls.
>> >>
>> >> But when i make an outside call, the call ends after 17 seconds.
>> Looking
>> >> at
>> >> the sip messages i see that i recieve a bye, then the call is gone.
>> >>
>> >> Am i doing something wrong, should the sip trunk be directly in
>> opensips?
>> >> and add that as a rewritehost? Or is this an Asterisk issue?
>> >>
>> >> My opensips is running on port 5090 (so are the phones) and my
>> >> asterisk+outside trunk is on 5060.
>> >> --
>> >> View this message in context:
>> >>
>> http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html
>> >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>> >>
>> >> _______________________________________________
>> >> Users mailing list
>> >> Users at lists.opensips.org
>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >>
>> >
>> > _______________________________________________
>> > Users mailing list
>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>> >
>>
>> --
>> View this message in context:
>> http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3775048.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

-- 
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.