17 Oct
2006
17 Oct
'06
6:42 p.m.
On Tue, 17 Oct 2006 10:36:20 -0400
Nathan Hawkins <utsl(a)quic.net> wrote:
Comments inline.
My gut feeling
is you've found a firmware bug, or it's not interacting right with
the other phone. I'd try calling between two Polycom's and see if
that works as expected.
Good luck!
no, it doesn't make a difference, i just tried it with a polycom 601
and 501 involved. While i guess your assumption that this is incorrect
behaviour of the polycom is true i'm not sure if this is the main
reason for the one-way audio issue i initially posted.
The sdp-part in the invite message on resuming the call on hold looks
like it would have to correct information to go through the nat, the
"c=a.b.c.d"-line is updated with the proxy ip and all other
nat-parameters are appended too. Could this be a problem of rtpproxy or
asterisk? The rtp-ports change a lot, seems some device sends the audio
to a port that has not been recognized by the firewall. Hope someone
has some additional input, i'll try to contact polycom but with their
strict reseller policy i guess i'll not get an answer...
best regards
christian
To clarify
what i'm talking about i've attached the whole sip-debug
of two calls, the first call is sent on hold when a second call
comes and resumed after the second call has ended(1111111801 calls
2222222804 and it answers, then 0891234567 calls 2222222804 and
1111111801 is set on hold and resumed later). Openser runs on .98,
which also serves the rtpproxy. .97 is a asterisk-gateway.
Thanks for the input
Christian
> What firmware version are you running?
>
> Nathan
>
> Benko wrote:
>
>> Hmm, actually this parameter is set to 0(default).
>> However, in practice it still uses the old standard(rfc2543).
>> There's no media direction parameter in the sdp-message sent by
>> the polycom for some reason, although the manual states to do so
>> (firmware 2.0.1). Is it possible to force this somehow?
>>
>> regards
>> christian
>>
>> On Sun, 15 Oct 2006 19:58:33 -0400
>> Nathan Hawkins <utsl(a)quic.net> wrote:
>>
>>
>>> There's an option for the Polycom phones to switch the hold
>>> behaviour.
>>>
>>> Set voIpProt.SIP.useRFC2543hold to 0, and it should use RFC3264
>>> rules for signalling hold instead of 0.0.0.0.
>>>
>>> Benko wrote:
>>>
>>>> Hello!
>>>>
>>>> I'm having a issue with NAT and rtpproxy. Usually my setup works
>>>> fine with natted clients, the Connection Information is
>>>> overwritten with the IP of the rtpproxy and audio passes through
>>>> in both directions. However, today i came across a problem where
>>>> the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the
>>>> private ip after resuming a call that was on hold(actually, the
>>>> other party is invited again) - and the force_rtp_proxy
>>>> ()-command on openser left the ip untouched instead of
>>>> overwriting it with the rtpproxy-ip. As a result the person that
>>>> was on hold had audio but the polycom user (with the "wrong"
ip)
>>>> hadn't.
>>>>
>>>> The false ip left aside, is it expected behaviour of
>>>> force_rtp_proxy to not touch 0.0.0.0?
>>>>
>>>> Just out of curiosity - does someone know the "on
hold"-problem
>>>> with polycoms?
>>>>
>>>> thx
>>>> christian
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>>
> ------------------------------------------------------------------------
>
> interface: bond0 (2XX.XXX.XXX.96/255.255.255.240)
> filter: ip and ( port 5060 and host 81.XXX.XXX.196 )
> #
> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060
> INVITE sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport. From: "801"
> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To:
> <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID:
> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 INVITE.
> Max-Forwards: 70.
> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>.
> P-Key-Flags: keys="3".
> User-Agent: snom190/3.60w.
> Accept: application/sdp.
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,
> SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer.
> Supported: timer, 100rel, replaces.
> Session-Expires: 3600.
> Content-Type: application/sdp.
> Content-Length: 268.
> .
> v=0.
> o=root 1814496751 1814496751 IN IP4 10.1.99.169.
> s=call.
> c=IN IP4 10.1.99.169.
> t=0 0.
> m=audio 55428 RTP/AVP 8 0 18 101.
> a=rtpmap:8 pcma/8000.
> a=rtpmap:0 pcmu/8000.
> a=rtpmap:18 g729/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport=2051;received=81.XXX.XXX.196.
> From: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h.
> To:
>
<sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=2f7d8fa65b1e4a32386804888f034ca8.5f9f.
> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 INVITE.
> Proxy-Authenticate: Digest realm="sip.de.mydomain.org",
> nonce="4534dc22e41f8b3a51321f458eed7830772d2ea7". Server: OpenSer
> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392
> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907
> req_src_ip=81.XXX.XXX.196 req_src_port=2051
> in_uri=sip:0892222222804@sip.de.mydomain.org;user=phone
> out_uri=sip:0892222222804@sip.de.mydomain.org;user=phone
> via_cnt==1". .
>
> #
> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060
> ACK sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport. From: "801"
> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To:
>
<sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=2f7d8fa65b1e4a32386804888f034ca8.5f9f.
> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 ACK.
> Max-Forwards: 70.
> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>.
> Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060
> INVITE sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;branch=z9hG4bK-x9wr16oqr17g;rport. From: "801"
> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To:
> <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID:
> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE.
> Max-Forwards: 70.
> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>.
> P-Key-Flags: keys="3".
> User-Agent: snom190/3.60w.
> Accept: application/sdp.
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,
> SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer.
> Supported: timer, 100rel, replaces.
> Session-Expires: 3600.
> Proxy-Authorization: Digest
>
username="11111111801",realm="sip.de.mydomain.org",nonce="4534dc22e41f8b3a51321f458eed7830772d2ea7",uri="sip:0892222222804@sip.de.mydomain.org;user=phone",response="139c40e2fecbadf4412628412607a7d7",algorithm=md5.
> Content-Type: application/sdp. Content-Length: 268.
> .
> v=0.
> o=root 1814496751 1814496751 IN IP4 10.1.99.169.
> s=call.
> c=IN IP4 10.1.99.169.
> t=0 0.
> m=audio 55428 RTP/AVP 8 0 18 101.
> a=rtpmap:8 pcma/8000.
> a=rtpmap:0 pcmu/8000.
> a=rtpmap:18 g729/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;branch=z9hG4bK-x9wr16oqr17g;rport=2051;received=81.XXX.XXX.196.
> From: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h.
> To: <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID:
> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE.
> Server: OpenSer (1.1.0-tls (i386/linux)).
> Content-Length: 0.
> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20909
> req_src_ip=81.XXX.XXX.196 req_src_port=2051
> in_uri=sip:0892222222804@sip.de.mydomain.org;user=phone
> out_uri=sip:0892222222804@mgw.de.mydomain.org;user=phone
> via_cnt==1". .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;received=81.XXX.XXX.196;branch=z9hG4bK-x9wr16oqr17g;rport=2051.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=lxlk4gyz7h>. From:
> "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To:
> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e.
> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE.
> User-Agent: MyMediaGW.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Contact: <sip:0892222222804@2XX.XXX.XXX.97>.
> Content-Type: application/sdp.
> Content-Length: 303.
> .
> v=0.
> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97.
> s=session.
> c=IN IP4 2XX.XXX.XXX.98.
> t=0 0.
> m=audio 36578 RTP/AVP 0 18 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=nortpproxy:yes.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> INVITE sip:2222222804@10.1.99.161:5060 SIP/2.0.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From:
> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To:
> <sip:2222222804@sip.de.mydomain.org>. Contact:
> <sip:0891111111@2XX.XXX.XXX.97:5060>. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. CSeq: 102
> INVITE. User-Agent: MyMediaGW.
> Max-Forwards: 69.
> Date: Tue, 17 Oct 2006 13:30:30 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Content-Type: application/sdp.
> Content-Length: 396.
> .
> v=0.
> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97.
> s=session.
> c=IN IP4 2XX.XXX.XXX.98.
> t=0 0.
> m=audio 36580 RTP/AVP 0 18 4 3 97 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:4 G723/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=direction:active.
> a=nortpproxy:yes.
>
> #
> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060
> ACK sip:0892222222804@2XX.XXX.XXX.97 SIP/2.0.
> Via: SIP/2.0/UDP
> 10.1.99.169:2051;branch=z9hG4bK-kqgbv28tcm8k;rport. Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=lxlk4gyz7h>. From: "801"
> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To:
> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e.
> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 ACK.
> Max-Forwards: 70.
> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>.
> Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From:
> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To:
> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. CSeq:
> 102 INVITE. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Contact:
> <sip:2222222804@10.1.99.161:5060>. Record-Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From:
> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To:
> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. CSeq:
> 102 INVITE. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Contact:
> <sip:2222222804@10.1.99.161:5060>. Record-Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. Allow: INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK,
> UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Type:
> application/sdp. Content-Length: 185. .
> v=0.
> o=- 1161091665 1161091665 IN IP4 10.1.99.161.
> s=Polycom IP Phone.
> c=IN IP4 10.1.99.161.
> t=0 0.
> m=audio 2266 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> ACK sip:2222222804@81.XXX.XXX.196:5060 SIP/2.0.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.2.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK77d83684;rport=5060. From:
> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To:
> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> Contact: <sip:0891111111@2XX.XXX.XXX.97>. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. CSeq: 102
> ACK. User-Agent: MyMediaGW.
> Max-Forwards: 69.
> Content-Length: 0.
> .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> INVITE sip:2222222804@10.1.99.161:5060 SIP/2.0.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From:
> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6.
> To: <sip:2222222804@sip.de.mydomain.org>. Contact:
> <sip:0891234567@2XX.XXX.XXX.97:5060>. Call-ID:
> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 102
> INVITE. User-Agent: MyMediaGW.
> Max-Forwards: 69.
> Date: Tue, 17 Oct 2006 13:30:38 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Content-Type: application/sdp.
> Content-Length: 396.
> .
> v=0.
> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97.
> s=session.
> c=IN IP4 2XX.XXX.XXX.98.
> t=0 0.
> m=audio 36584 RTP/AVP 0 18 4 3 97 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:4 G723/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=direction:active.
> a=nortpproxy:yes.
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From:
> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6.
> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB.
> CSeq: 102 INVITE. Call-ID:
> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact:
> <sip:2222222804@10.1.99.161:5060>. Record-Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From:
> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6.
> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB.
> CSeq: 102 INVITE. Call-ID:
> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact:
> <sip:2222222804@10.1.99.161:5060>. Record-Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Allow-Events:
> talk,hold,conference. Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 1 INVITE.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported:
> 100rel,replaces. Allow-Events: talk,hold,conference.
> Max-Forwards: 70.
> Content-Type: application/sdp.
> Content-Length: 181.
> .
> v=0.
> o=- 1161091665 1161091666 IN IP4 10.1.99.161.
> s=Polycom IP Phone.
> c=IN IP4 0.0.0.0.
> t=0 0.
> m=audio 2266 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C;rport=5060;received=81.XXX.XXX.196.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> CSeq: 1 INVITE. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Proxy-Authenticate: Digest realm="sip.de.mydomain.org",
> nonce="4534dc2bd45af3e3355df978e228223446d25d5b". Server: OpenSer
> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392
> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907
> req_src_ip=81.XXX.XXX.196 req_src_port=5060
> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060
> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 1 ACK.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKf40b74c7C2612792.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 2 INVITE.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported:
> 100rel,replaces. Allow-Events: talk,hold,conference.
> Proxy-Authorization: Digest username="2222222804",
> realm="sip.de.mydomain.org",
> nonce="4534dc2bd45af3e3355df978e228223446d25d5b",
> uri="sip:0891111111@2XX.XXX.XXX.97:5060",
> response="a690d1208542f7132ef16747f0abb98a", algorithm=MD5.
> Max-Forwards: 70. Content-Type: application/sdp. Content-Length:
> 181. . v=0. o=- 1161091665 1161091666 IN IP4 10.1.99.161.
> s=Polycom IP Phone.
> c=IN IP4 0.0.0.0.
> t=0 0.
> m=audio 2266 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;branch=z9hG4bKf40b74c7C2612792;rport=5060;received=81.XXX.XXX.196.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> CSeq: 2 INVITE. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Server:
> OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0.
> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907
> req_src_ip=81.XXX.XXX.196 req_src_port=5060
> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060
> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;rport=5060;received=81.XXX.XXX.196;branch=z9hG4bKf40b74c7C2612792.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> CSeq: 2 INVITE. User-Agent: MyMediaGW.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Contact: <sip:0891111111@2XX.XXX.XXX.97>.
> Content-Type: application/sdp.
> Content-Length: 232.
> .
> v=0.
> o=root 6068 6069 IN IP4 2XX.XXX.XXX.97.
> s=session.
> c=IN IP4 2XX.XXX.XXX.98.
> t=0 0.
> m=audio 36580 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=nortpproxy:yes.
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK4ec65e93B39DC51E.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 2 ACK.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Proxy-Authorization:
> Digest username="2222222804", realm="sip.de.mydomain.org",
> nonce="4534dc2bd45af3e3355df978e228223446d25d5b",
> uri="sip:0891111111@2XX.XXX.XXX.97:5060",
> response="a690d1208542f7132ef16747f0abb98a", algorithm=MD5.
> Max-Forwards: 70. Content-Length: 0. .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From:
> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6.
> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB.
> CSeq: 102 INVITE. Call-ID:
> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact:
> <sip:2222222804@10.1.99.161:5060>. Record-Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. Allow: INVITE, ACK,
> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK,
> UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Type:
> application/sdp. Content-Length: 185. .
> v=0.
> o=- 1161091674 1161091674 IN IP4 10.1.99.161.
> s=Polycom IP Phone.
> c=IN IP4 10.1.99.161.
> t=0 0.
> m=audio 2268 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> ACK sip:2222222804@81.XXX.XXX.196:5060 SIP/2.0.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.2.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK4365390f;rport=5060. From:
> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6.
> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB.
> Contact: <sip:0891234567@2XX.XXX.XXX.97>. Call-ID:
> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 102
> ACK. User-Agent: MyMediaGW.
> Max-Forwards: 69.
> Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> BYE sip:0891234567@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKc2fe28198F44BA84.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB.
> To: "0891234567"
> <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. Route:
> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. CSeq: 1 BYE.
> Call-ID: 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;received=81.XXX.XXX.196;branch=z9hG4bKc2fe28198F44BA84.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=4245106-795C21FB>.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB.
> To: "0891234567"
> <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. Call-ID:
> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 1 BYE.
> User-Agent: MyMediaGW. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
> REFER, SUBSCRIBE, NOTIFY. Contact: <sip:0891234567@2XX.XXX.XXX.97>.
> Content-Length: 0.
> X-Asterisk-HangupCause: Normal Clearing.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 3 INVITE.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported:
> 100rel,replaces. Allow-Events: talk,hold,conference.
> Proxy-Authorization: Digest username="2222222804",
> realm="sip.de.mydomain.org",
> nonce="4534dc2bd45af3e3355df978e228223446d25d5b",
> uri="sip:0891111111@2XX.XXX.XXX.97:5060",
> response="4fafb927667083777a520a6f717fc8d3", algorithm=MD5.
> Max-Forwards: 70. Content-Type: application/sdp. Content-Length:
> 247. . v=0. o=- 1161091665 1161091667 IN IP4 10.1.99.161.
> s=Polycom IP Phone.
> c=IN IP4 10.1.99.161.
> t=0 0.
> a=sendrecv.
> m=audio 2266 RTP/AVP 0 8 18 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3;rport=5060;received=81.XXX.XXX.196.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> CSeq: 3 INVITE. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Proxy-Authenticate: Digest realm="sip.de.mydomain.org",
> nonce="4534dc3073903ac300e605f4fd2834968e7ef932". Server: OpenSer
> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392
> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20903
> req_src_ip=81.XXX.XXX.196 req_src_port=5060
> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060
> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 3 ACK.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK3fa5014e6D19C349.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 4 INVITE.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported:
> 100rel,replaces. Allow-Events: talk,hold,conference.
> Proxy-Authorization: Digest username="2222222804",
> realm="sip.de.mydomain.org",
> nonce="4534dc3073903ac300e605f4fd2834968e7ef932",
> uri="sip:0891111111@2XX.XXX.XXX.97:5060",
> response="e6f5f00560921aceca9a0329333b720f", algorithm=MD5.
> Max-Forwards: 70. Content-Type: application/sdp. Content-Length:
> 247. . v=0. o=- 1161091665 1161091667 IN IP4 10.1.99.161.
> s=Polycom IP Phone.
> c=IN IP4 10.1.99.161.
> t=0 0.
> a=sendrecv.
> m=audio 2266 RTP/AVP 0 8 18 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:101 telephone-event/8000.
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;branch=z9hG4bK3fa5014e6D19C349;rport=5060;received=81.XXX.XXX.196.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> CSeq: 4 INVITE. Call-ID:
> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Server:
> OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0.
> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20903
> req_src_ip=81.XXX.XXX.196 req_src_port=5060
> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060
> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;rport=5060;received=81.XXX.XXX.196;branch=z9hG4bK3fa5014e6D19C349.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> CSeq: 4 INVITE. User-Agent: MyMediaGW.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Contact: <sip:0891111111@2XX.XXX.XXX.97>.
> Content-Type: application/sdp.
> Content-Length: 303.
> .
> v=0.
> o=root 6068 6070 IN IP4 2XX.XXX.XXX.97.
> s=session.
> c=IN IP4 2XX.XXX.XXX.98.
> t=0 0.
> m=audio 36580 RTP/AVP 0 18 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=nortpproxy:yes.
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKca5c1daDE079395.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 4 ACK.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Proxy-Authorization:
> Digest username="2222222804", realm="sip.de.mydomain.org",
> nonce="4534dc3073903ac300e605f4fd2834968e7ef932",
> uri="sip:0891111111@2XX.XXX.XXX.97:5060",
> response="e6f5f00560921aceca9a0329333b720f", algorithm=MD5.
> Max-Forwards: 70. Content-Length: 0. .
>
> #
> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060
> BYE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKcda270c0BA153C5B.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>.
> CSeq: 5 BYE.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> Contact: <sip:2222222804@10.1.99.161:5060>.
> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291.
> Proxy-Authorization: Digest username="2222222804",
> realm="sip.de.mydomain.org",
> nonce="4534dc3073903ac300e605f4fd2834968e7ef932",
> uri="sip:0891111111@2XX.XXX.XXX.97:5060",
> response="d1e585adb491da5b0643fcdb7a5d8cef", algorithm=MD5.
> Max-Forwards: 70. Content-Length: 0. .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 10.1.99.161:5060;received=81.XXX.XXX.196;branch=z9hG4bKcda270c0BA153C5B.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>.
> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A.
> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a.
> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org.
> CSeq: 5 BYE. User-Agent: MyMediaGW.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Contact: <sip:0891111111@2XX.XXX.XXX.97>.
> Content-Length: 0.
> X-Asterisk-HangupCause: Normal Clearing.
> .
>
> #
> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051
> BYE sip:11111111801@81.XXX.XXX.196:2051 SIP/2.0.
> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as0fd69c4e>.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bKf278.897153c6.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK13f9319f;rport=5060. From:
> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e.
> To: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h.
> Contact: <sip:0892222222804@2XX.XXX.XXX.97>. Call-ID:
> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 102 BYE.
> User-Agent: MyMediaGW.
> Max-Forwards: 69.
> Content-Length: 0.
> .
>
> #
> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bKf278.897153c6.0.
> Via: SIP/2.0/UDP
> 2XX.XXX.XXX.97:5060;branch=z9hG4bK13f9319f;rport=5060. From:
> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e.
> To: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h.
> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 102 BYE.
> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>.
> User-Agent: snom190/3.60w.
> Content-Length: 0.
> .
>
> exit
> 78 received, 0 dropped
>