[Users] Issues with calls using openser.
Klaus Darilion
klaus.mailinglists at pernau.at
Wed Jan 3 23:51:02 CET 2007
Hi!
I think there are several problems:
1. your config is broken. It looks like the caller is a gateway. Usually
gateways are not authenticated by the proxy with digest authentication but
use IP based authentication. Thus, as you challenge the reINVITE, the
gateway has no credentials to answer the challenge and hangs up.
2. The gateway offers µ-LAW and G.729. The Grandstream answers with G.729.
This looks like a normal call setup. Nevertheless the gateway sends a
reINVITE with explicitely offering only G.729. Why? Maybe the gateway does
not support receiving µ-LAW and sending G.729 - thus it sends a reINVITE
to force the received codec to G.729.
Conclusion - fix your config.
Use IP based authentication and TCP (e.g. is_fromgw from LCR module) to
authenticate your gateway to the proxy.
Use digest authentication to authenticate users.
reINVITEs (which are handled in the loose_route() block) can be also
authenticated using the above methods - but not authenticating reINVITEs
is usually no security issue as the totag acts somehow as a "cookie" to
the UAS.
regards
klaus
On Wed, January 3, 2007 22:25, Shane Burrell said:
> I changed the codec to ULAW which is defiantly supported. I'm thinking
> the
> reINVITE may be the problem but I'm pretty new at openser configuration
> and
> don't see a clear way to detect a reinvite and not auth it. I did capture
> a
> ngrep of a failed call. I'll also test xlite or a sipura tonight to see
> if
> it something specific to the grandstream.
>
> ngrep -q -t -W byline port 5060
> interface: eth0 (192.168.16.0/255.255.255.0)
> filter: (ip) and ( port 5060 )
>
>
>
>
>
>
> U 2007/01/03 16:18:41.764242 192.168.16.91:5060 -> 192.168.16.192:5060
> INVITE sip:7005874200 at siprt1.siptest.net:5060;user=phone SIP/2.0.
> t: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> Remote-Party-Id:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
> ;party=calling;privacy=off.
> Proxy-Require: privacy.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 70.
> m: <sip:7006311229 at 192.168.16.91:5060;user=phone>.
> k: replaces.
> c: application/sdp.
> Accept: application/sdp.
> Accept-Encoding: .
> Accept-Language: en.
> User-Agent: MSTSYLVAIPGW.
> l: 249.
> .
> v=0.
> o=MSTNT 536702936 536702936 IN IP4 192.168.16.91.
> s=Session SDP.
> c=IN IP4 192.168.16.91.
> t=0 0.
> m=audio 40878 RTP/AVP 18 0 101 .
> a=silenceSupp:off.
> a=ecan:b on g168.
> a=rtpmap:101 telephone-event/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
>
>
> U 2007/01/03 16:18:41.764646 192.168.16.192:5060 -> 192.168.16.91:5060
> SIP/2.0 100 Giving a try.
> t: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Server: OpenSer (1.2.0-dev12-notls (i386/linux)).
> Content-Length: 0.
> Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14845
> req_src_ip=192.168.16.91 req_src_port=5060
> in_uri=sip:7005874200 at siprt1.siptest.net:5060;user=phone
> out_uri=sip:7005874200 at 192.168.17.83;user=phone via_cnt==1".
> .
>
>
> U 2007/01/03 16:18:41.764673 192.168.16.192:5060 -> 192.168.17.83:5060
> INVITE sip:7005874200 at 192.168.17.83;user=phone SIP/2.0.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> t: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> Remote-Party-Id:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
> ;party=calling;privacy=off.
> Proxy-Require: privacy.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 69.
> m: <sip:7006311229 at 192.168.16.91:5060;user=phone>.
> k: replaces.
> c: application/sdp.
> Accept: application/sdp.
> Accept-Encoding: .
> Accept-Language: en.
> User-Agent: MSTSYLVAIPGW.
> l: 249.
> .
> v=0.
> o=MSTNT 536702936 536702936 IN IP4 192.168.16.91.
> s=Session SDP.
> c=IN IP4 192.168.16.91.
> t=0 0.
> m=audio 40878 RTP/AVP 18 0 101 .
> a=silenceSupp:off.
> a=ecan:b on g168.
> a=rtpmap:101 telephone-event/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
>
>
> U 2007/01/03 16:18:41.774319 192.168.17.83:5060 -> 192.168.16.192:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To: <sip:7005874200 at siprt1.siptest.net:5060;user=phone>.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Content-Length: 0.
> .
>
>
> U 2007/01/03 16:18:41.776363 192.168.17.83:5060 -> 192.168.16.192:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Content-Length: 0.
> .
>
>
> U 2007/01/03 16:18:41.776442 192.168.16.192:5060 -> 192.168.16.91:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Content-Length: 0.
> .
>
>
> U 2007/01/03 16:18:44.426461 192.168.17.83:5060 -> 192.168.16.192:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> Content-Type: application/sdp.
> Supported: replaces.
> Content-Length: 220.
> .
> v=0.
> o=7005874200 8000 8000 IN IP4 192.168.17.83.
> s=SIP Call.
> c=IN IP4 192.168.17.83.
> t=0 0.
> m=audio 10000 RTP/AVP 18 101.
> a=sendrecv.
> a=rtpmap:18 G729/8000.
> a=ptime:20.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-11.
>
>
> U 2007/01/03 16:18:44.426613 192.168.16.192:5060 -> 192.168.16.91:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 INVITE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> Content-Type: application/sdp.
> Supported: replaces.
> Content-Length: 220.
> .
> v=0.
> o=7005874200 8000 8000 IN IP4 192.168.17.83.
> s=SIP Call.
> c=IN IP4 192.168.17.83.
> t=0 0.
> m=audio 10000 RTP/AVP 18 101.
> a=sendrecv.
> a=rtpmap:18 G729/8000.
> a=ptime:20.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-11.
>
>
> U 2007/01/03 16:18:44.451021 192.168.16.91:5060 -> 192.168.16.192:5060
> ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 ACK.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 70.
> Route: <sip:7005874200 at 192.168.17.83;user=phone>.
> User-Agent: MSTSYLVAIPGW.
> l: 0.
> .
>
>
> U 2007/01/03 16:18:44.451221 192.168.16.192:5060 -> 192.168.17.83:5060
> ACK sip:7005874200 at 192.168.17.83;user=phone SIP/2.0.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979699 ACK.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.2.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 69.
> User-Agent: MSTSYLVAIPGW.
> l: 0.
> .
>
>
> U 2007/01/03 16:18:44.451328 192.168.16.91:5060 -> 192.168.16.192:5060
> INVITE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> Remote-Party-Id:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;screen=yes;id-type=subscriber
> ;party=calling;privacy=off.
> Proxy-Require: privacy.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979700 INVITE.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 70.
> Route: <sip:7005874200 at 192.168.17.83;user=phone>.
> m: <sip:7006311229 at 192.168.16.91:5060;user=phone>.
> c: application/sdp.
> Accept: application/sdp.
> Accept-Encoding: .
> Accept-Language: en.
> User-Agent: MSTSYLVAIPGW.
> l: 236.
> .
> v=0.
> o=MSTNT 536702936 536702937 IN IP4 192.168.16.91.
> s=Session SDP.
> c=IN IP4 192.168.16.91.
> t=0 0.
> m=audio 40878 RTP/AVP 18 101.
> a=silenceSupp:off.
> a=ecan:b on g168.
> a=ptime:20.
> a=rtpmap:101 telephone-event/8000.
> a=rtpmap:18 G729/8000.
>
>
> U 2007/01/03 16:18:44.453398 192.168.16.192:5060 -> 192.168.16.91:5060
> SIP/2.0 407 Proxy Authentication Required.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979700 INVITE.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Proxy-Authenticate: Digest realm="192.168.16.91",
> nonce="459c1ee0731e3291a5704b9666ffded6acb20bb5".
> Server: OpenSer (1.2.0-dev12-notls (i386/linux)).
> Content-Length: 0.
> Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14844
> req_src_ip=192.168.16.91 req_src_port=5060
> in_uri=sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598
> out_uri=sip:7005874200 at 192.168.17.83;user=phone via_cnt==1".
> .
>
>
> U 2007/01/03 16:18:44.471821 192.168.16.91:5060 -> 192.168.16.192:5060
> ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979700 ACK.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 70.
> User-Agent: MSTSYLVAIPGW.
> l: 0.
> .
>
>
> U 2007/01/03 16:18:44.472045 192.168.16.91:5060 -> 192.168.16.192:5060
> BYE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979701 BYE.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 70.
> Route: <sip:7005874200 at 192.168.17.83;user=phone>.
> Accept: application/sdp.
> Accept-Encoding: .
> Accept-Language: en.
> User-Agent: MSTSYLVAIPGW.
> l: 0.
> .
>
>
> U 2007/01/03 16:18:44.474251 192.168.16.192:5060 -> 192.168.17.83:5060
> BYE sip:7005874200 at 192.168.17.83;user=phone SIP/2.0.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> t:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> f:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> i: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979701 BYE.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0.
> v: SIP/2.0/UDP 192.168.16.91:5060.
> Max-Forwards: 69.
> Accept: application/sdp.
> Accept-Encoding: .
> Accept-Language: en.
> User-Agent: MSTSYLVAIPGW.
> l: 0.
> .
>
>
> U 2007/01/03 16:18:44.506433 192.168.17.83:5060 -> 192.168.16.192:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979701 BYE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> Supported: replaces.
> Content-Length: 0.
> .
>
>
> U 2007/01/03 16:18:44.506549 192.168.16.192:5060 -> 192.168.16.91:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.16.91:5060.
> Record-Route: <sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598>.
> From:
> <sip:7006311229 at 192.168.16.91:5060;user=phone>;tag=d0b60bd7-1ffd6fd8-5b10359
> 8.
> To:
> <sip:7005874200 at siprt1.siptest.net:5060;user=phone>;tag=6f6a15c679df8aa8.
> Call-ID: 1068bc1b-1bb-1ffd6fd8 at 192.168.16.91.
> CSeq: 12979701 BYE.
> User-Agent: Grandstream HT496 1.0.3.64 FXS0.
> Contact: <sip:7005874200 at 192.168.17.83;user=phone>.
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> Supported: replaces.
> Content-Length: 0.
> .
>
>
> U 2007/01/03 16:18:47.652882 192.168.17.150:5060 -> 192.168.16.192:5060
> ..................
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> -----Original Message-----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> Sent: Wednesday, January 03, 2007 7:51 AM
> To: Shane Burrell
> Cc: users at openser.org
> Subject: Re: [Users] Issues with calls using openser.
>
> The log shows that you challenge reINVITEs. MAybe this breaks the
> grandstream. Please try without challenging the reINVITE. If this helps,
> then it is a probably a grandstream bug.
>
> But of course there is the question why the grandstream sends a reINVITE
> at all? This is often a codec problem.
>
> Please also try to use other client (e.g. xlite) and post complete ngrep
> dumps: "ngrep -q -t -W byline port 5060"
>
> regards
> klaus
>
>
> Shane Burrell wrote:
>> The SIP UA was a grandstream ATA running the latest stable firmware.
> Prior
>> to upgrading to 1.1 and moving to mediaproxy it worked well with the
>> exception of good nat support which is why I would really like
>> mediaproxy
> to
>> work. Is there anything I should look for in the sip dialog to
>> determine
> if
>> the client, sip proxy, or the gateway is the culprit on disconnecting
>> the
>> call?
>>
>> Thanks,
>>
>> Shane
>>
>> -----Original Message-----
>> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
>> Sent: Wednesday, January 03, 2007 4:52 AM
>> To: Shane Burrell
>> Cc: users at openser.org
>> Subject: Re: [Users] Issues with calls using openser.
>>
>>
>> Maybe a bug in the caller's SIP client?
>>
>> regards
>> klaus
>>
>> Shane Burrell wrote:
>>> I recently installed the latest version of openser and this time used
>>> mediaproxy rather than rtpproxy. Everything seems to work but if a sip
>>> device is called the phone rings and is instantally disconnected and
>>> the
>> far
>>> end is left off-hook. This worked before but I did modify my script to
>> work
>>> with mediaproxy. Below is the wireshark decode of the sip messagining.
>> Any
>>> help or suggestions on where to look would be great. Calls from the
>>> sip
>>> device works flawlessly. I am using a MaxTNT as the gateway.
>>>
>>>
>>>
>>>
>>>
>>> |Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
>>>
>>> |22.031 | INVITE SDP ( telephone-event) |
>>> |SIP From: sip:8385021101 at 192.168.16.91:5060 To:sip: 8385024200@
>>> siprt1.me.net:5060
>>>
>>> | |(5060) ------------------> (5060) | |
>>>
>>> |22.031 | 100 Giving a try |
> |SIP
>>> Status
>>>
>>> | |(5060) <------------------ (5060) | |
>>>
>>> |22.031 | | INVITE SDP ( telephone-event)
>>> |SIP Request
>>>
>>> | | |(5060) ------------------> (5060) |
>>>
>>> |22.040 | | 100 Trying|
> |SIP
>>> Status
>>>
>>> | | |(5060) <------------------ (5060) |
>>>
>>> |22.042 | | 180 Ringing
> |SIP
>>> Status
>>>
>>> | | |(5060) <------------------ (5060) |
>>>
>>> |22.042 | 180 Ringing |
> |SIP
>>> Status
>>>
>>> | |(5060) <------------------ (5060) | |
>>>
>>> |25.244 | | 200 OK SDP ( telephone-event)
>>> |SIP Status
>>>
>>> | | |(5060) <------------------ (5060) |
>>>
>>> |25.245 | 200 OK SDP ( telephone-event) |
>>> |SIP Status
>>>
>>> | |(5060) <------------------ (5060) | |
>>>
>>> |25.269 | ACK | |
> |SIP
>>> Request
>>>
>>> | |(5060) ------------------> (5060) | |
>>>
>>> |25.269 | | ACK |
> |SIP
>>> Request
>>>
>>> | | |(5060) ------------------> (5060) |
>>>
>>> |25.269 | INVITE SDP ( telephone-event) |
>>> |SIP From: sip: 8385021101 at 152.93.36.91:5060 To:sip: 8385024200@
>>> siprt1.me.net:5060
>>>
>>> | |(5060) ------------------> (5060) | |
>>>
>>> |25.270 | 407 Proxy Authentication Required |
>>> |SIP Status
>>>
>>> | |(5060) <------------------ (5060) | |
>>>
>>> |25.291 | ACK | |
> |SIP
>>> Request
>>>
>>> | |(5060) ------------------> (5060) | |
>>>
>>> |25.291 | BYE | |
> |SIP
>>> Request
>>>
>>> | |(5060) ------------------> (5060) | |
>>>
>>> |25.293 | | BYE |
> |SIP
>>> Request
>>>
>>> | | |(5060) ------------------> (5060) |
>>>
>>> |25.326 | | 200 OK |
> |SIP
>>> Status
>>>
>>> | | |(5060) <------------------ (5060) |
>>>
>>> |25.327 | 200 OK | |
> |SIP
>>> Status
>>>
>>> | |(5060) <------------------ (5060) | |
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Shane
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Klaus Darilion
> nic.at
>
>
More information about the Users
mailing list