17 sec, recieve a bye and a hangup

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17 sec, recieve a bye and a hangup

Peter den Hartog
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.
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Re: 17 sec, recieve a bye and a hangup

Brett Nemeroff
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 <[hidden email]> 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.
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View this message in context: http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html
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Re: 17 sec, recieve a bye and a hangup

Peter den Hartog
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@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@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: 17 sec, recieve a bye and a hangup

Brett Nemeroff
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 <[hidden email]> 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
> <[hidden email]>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
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> 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
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Re: 17 sec, recieve a bye and a hangup

Peter den Hartog
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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>.
CSeq: 1 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@172.16.0.12.
Contact: <sip:701@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=67747a5e755302d1b99e6b9647717b58.aaf5.
CSeq: 1 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@172.16.1.10:5090 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK2867f2a43260B0C9.
From: "701" <sip:701@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=67747a5e755302d1b99e6b9647717b58.aaf5.
CSeq: 1 ACK.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@172.16.0.12.
Contact: <sip:701@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>.
CSeq: 2 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@172.16.0.12.
Contact: <sip:701@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>.
CSeq: 2 INVITE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK1b9cbb48F2BE247D.
From: "701" <sip:701@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@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@172.16.0.12.
Contact: <sip:701@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893.
From: "701" <sip:701@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@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@172.16.0.12.
Contact: <sip:701@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
CSeq: 3 BYE.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bK1b9cbb48F2BE247D.
From: "701" <sip:701@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@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@172.16.0.12.
Contact: <sip:701@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@172.16.1.10;user=phone>;tag=as431f0138.
Call-ID: 3f4ff522-ba85e2ff-6b6924f0@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@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@172.16.1.10 SIP/2.0.
Via: SIP/2.0/UDP 172.16.0.12;branch=z9hG4bKcd927226B5124893.
From: "701" <sip:701@172.16.1.10>;tag=A2EA31C5-DC95458E.
To: <sip:0624469780@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@172.16.0.12.
Contact: <sip:701@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@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@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@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
> > _______________________________________________
> > Users mailing list
> > Users@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@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: 17 sec, recieve a bye and a hangup

Brett Nemeroff
Ok, so as you can see from the trace, the 200OK is retransmitted.. you'll see:
200OK
ACK
200OK
200OK
200OK
200OK
BYE

This is because the ACK never made it to Astersk. It's a scripting error in Opensips. You may want to check your loose route block for errors. 

Also, it's worth mentioning that the behavior to hang up in 17 seconds is correct. It's saying "I never got confirmation that the call really got completed, so I'm just going to end it". What's really happening is that the 200OK is timing out.

On Tue, Oct 6, 2009 at 8:48 AM, Peter den Hartog <[hidden email]> wrote:

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.


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