Hi all,
How I can supress of uri user=phone ?
This ATA receive call from PSTN, but, when I send calls to PSTN, i don't
obtain response.
The integration to PSTN is on Asterisk.
Thanks
R2
Hi All
I have a scenario in which I want to deploy SER with mediaproxy & allow
everyone/anyone connecting with my SER to talk to each other. I am using
ONSIP gettingStarted document. In order to allow everyone to connect to my
SER without authentication I commented the following portion in ser.cfg:
in Route[2]
#if (!www_authorize("","subscriber")) {
# www_challenge("","0");
# break;
#};
#if (!check_to()) {
# sl_send_reply("401", "Unauthorized");
# break;
#};
# consume_credentials();
and in Route[3]
# if (!proxy_authorize("","subscriber")) {
# proxy_challenge("","0");
# break;
#} else if (!check_from()) {
# sl_send_reply("403", "Use From=ID");
# break;
#};
#consume_credentials();
Now when I try to connect 2 UA (Simpleopal) from behind a same NAT, they do
get registered, & records also appear in location table of DataBase. But
when I place call from UA1 to UA2 like "call sip:UA2@aa.bb.cc.dd", where
aa.bb.cc.dd is the public IP of my SER machine (mediaproxy is also running
on same machine), it ends up showing transport error. On SER side:
A) SER logs say:
on lookup() "UA2 not found in usrloc" and on is_local() "Realm 'ab.cc.dd.ee'
is not local". the is_lcoal appears again saying "Realm 192.168.0.241 is not
local" where 192.168.0.241 is UA2's local IP.
B)Mediaproxy log says:
command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio
203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info=
from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd
,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag:
session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: started. listening on
aa.bb.cc.dd:35000
command execution time: 9.35 ms
command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio
203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info=
from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd
,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag:
command execution time: 0.60 ms
session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: 0/0/0 packets, 0/0/0 bytes
(caller/called/relayed)
session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: ended (did timeout).
(203.82.51.9) is my LAN's public IP aa.bb.cc.dd is the public IP of SER
machine
C) using ngrep I found out that it sends an "SIP/2.0 408 Request Timeout"
cant figure out whats going on, anybody having any idea please help.
Thanx in advance
Regards
Muhammad Asif Ali
Dear all,
I am a student and am trying to setup a demonstration with TLS and DTLS
support between a SIP client and a Proxy.
Has anyone some information regarding adding DTLS support for a SIP
client/Proxy or a more generic one using OpenSSL. I just need it for
demonstration purposes and so error handling's if any need not be complete.
Just a simple support and a demonstration of both would do fine. If anyone
has any information regarding this, may be a how to start or any
introductory article would be of great help, kindly let me know. This would
be of great help. Any suggestions regarding the same are most welcome.
kindly let me know and thank you very much.
regards,
Pjothi
Hi!
I've read about the above topic in the arcives and now I've got the same
problem.
User A is behind a symmetric NAT -> a new port is bound for every
different
(external ip, external port) pair.
User B has a connection to the net without any NAT or firewall.
User A wants to have a SIP+RTP session with user B. How?
I guess basically what is needed is forming of a SIP
message like "send the your reply to the same ip and
same port from where you have received this" and user
A can send this to user B.
For the RTP session, a pretty similar thing would be
needed like, A says to B: "I send you the RTP stream
on your x port, send your rtp stream back to the same
ip and port from where you receive mine".
How to setup such a SIP packet?
Or maybe there is a standard solution for this already?
(STUN wont help in this case I guess, and TURN is not
an option for me.)
I've read about "symmetric SIP + symmetric RTP" maybe
that is the same as I've described above?
Thanks
Istvan
Hello to all
Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know
the best way of implement a centralized address book system.
Maybe the solution is LDAP, but these clients doesnt seem to support
LDAP.Who should contact the LDAP directory? the SIP clients or the SIP
server?
Thanks
Joao Pereira
Do you Know HSRP from Cisco System ?
http://www.cisco.com/univercd/cc/td/doc/cisintwk/ics/cs009.htm
Harry
--- Hendrik Scholz <hendrik.scholz(a)freenet-ag.de> a
écrit :
> Hi!
>
> hgaillac-sip(a)yahoo.fr wrote:
> > What about VRRP for failover ?
> >
> > SER1 (master)
> > sip agents == || === Asterisk Farm==sip/pstn
> gateway
> > SER2 (slave)
>
> What do you mean by that?
>
> Somewhat confused,
> Hendrik
>
> --
> freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114
> Kiel, Germany
> Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559
> Internet: http://www.freenet.de, eMail:
> hendrik.scholz(a)freenet-ag.de
>
___________________________________________________________________________
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I have two window messenger 5.1 clients on the same subnet as the SER server w/ pa module. They subscribe/notify to each other ok. HOwever they cannot IM each other. The callee, after receiveing INVITE, sent back a 100 trying, then 200 OK, and then immediately BYE.
I searched thru archives and looks like this is a known issue with wm. What is the solution or workaround to this?
Here is the messages I captured:
U 10.1.104.251:1172 -> 10.1.104.254:5060
INVITE sip:104pc1@10.1.104.254 SIP/2.0..Via: SIP/2.0/UDP 10.1.104.251:8053
..Max-Forwards: 70..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;
tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1
.104.254>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Cont
act: <sip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: applicati
on/sdp..Content-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.251..s=session
..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc2@10.1.104.254
..
#
U 10.1.104.254:5060 -> 10.1.104.251:8053
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.1.
104.251:8053..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d4
6c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.2
54>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Server: Si
p EXpress router (0.10.99-dev30-tm-timers-pa-3 (i386/linux))..Content-Leng
th: 0..Warning: 392 10.1.104.254:5060 "Noisy feedback tells: pid=23866 re
q_src_ip=10.1.104.251 req_src_port=1172 in_uri=sip:104pc1@10.1.104.254 out
_uri=sip:10.1.104.248:13018 via_cnt==1"....
#
U 10.1.104.254:5060 -> 10.1.104.248:13018
INVITE sip:10.1.104.248:13018 SIP/2.0..Record-Route: <sip:10.1.104.254;fta
g=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Via: SIP/2.0/UDP 10.1.104.254;b
ranch=z9hG4bK2216.fea17c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..Max-Forw
ards: 16..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d46c61
cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.254>.
.Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Contact: <sip:
10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: application/sdp..Co
ntent-Length: 119..P-hint: usrloc applied....v=0..o=- 0 0 IN IP4 10.1.104.
251..s=session..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc
2(a)10.1.104.254..
#
U 10.1.104.248:1334 -> 10.1.104.254:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17
c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <si
p:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d136
2..To: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Cal
l-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..User-Agent: RTC/1.
3..Content-Length: 0....
#
U 10.1.104.248:1334 -> 10.1.104.254:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17c01.
0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <sip:10
;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T">4pc2(a)10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T
o: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Call-ID
: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Record-Route: <sip:10.
1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Contact: <sip:10.1
.104.248:13018>..User-Agent: RTC/1.3..Content-Type: application/sdp..Conte
nt-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.248..s=session..c=IN IP4 10
.1.104.248..t=0 0..m=message 5060 sip sip:104pc1@10.1.104.254..
#
U 10.1.104.248:1334 -> 10.1.104.254:5060
BYE sip:10.1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on SIP/2.0..
Via: SIP/2.0/UDP 10.1.104.248:13018..Max-Forwards: 70..From: <sip:104pc1@1
0.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..To: "104pc2(a)10.1.104.25
4" <sip:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=692
00d1362..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 BYE..Route: <s
ip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Length: 0....
##
Hi,
I configured openser to handle redirect responses and re-route calls based
on the Contact header from the 3xx response using the uac_redirect module.
In addition to that, I would like to transfer some headers from the 3xx
response into the redirected INVITE (see diagram):
UAC openser UAS1 UAS2
| | | |
|---INVITE--->| | |
| |---INVITE--->| |
| |<--3XX-------| |
| | with Diversion header |
| |---ACK------>| |
| |---------------INVITE------->|
| | | with Diversion header
| | | from 3xx response from UAS1
Is there an API that will allow me to search for headers inside a
response and save them in to an AVP?
Thx,
-ovi
great, thanks for the clarification...so simply,
rewrite("host") would trigger SRV lookup.
--- Klaus Darilion <klaus.mailinglists(a)pernau.at>
wrote:
> Hi Alan!
>
> That's the correct behaviour. Please read RFC 3263.
> If the URI contains a
> port, e.g. inside.acme.com:5060 there must not be an
> SRV lookup. Because
> the SRV lookup leads to a port, which is already
> defined.
>
> regards
> klaus
>
> On Tue, January 31, 2006 20:28, alan chiu said:
> > Server: Sip EXpress router (0.8.12
> (i86pc/solaris))
> >
> > when endpoint sends 18005551212(a)sip.acme.com, SER
> > (without rewriting of R-URI, looks it up with SRV
> > ***************
> >
> > 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP
> > Request: INVITE sip:18005551212@sip.acme.com, with
> > session description
> > 386.657307 10.100.0.111 -> 192.168.1.246 SIP
> Status:
> > 100 trying -- your call is important to us
> > 88.811658 10.100.0.111 -> 192.168.1.11 DNS
> Standard
> > query SRV _sip._udp.sip.acme.com
> > 98.825208 192.168.1.11 -> 10.100.0.111 DNS
> Standard
> > query response, No such name
> > 98.825496 10.100.0.111 -> 192.168.1.11 DNS
> Standard
> > query A sip.acme.com
> > 108.845136 192.168.1.11 -> 10.100.0.111 DNS
> Standard
> > query response, No such name
> > 108.845468 10.100.0.111 -> 192.168.1.11 DNS
> Standard
> > query AAAA sip.acme.com
> > 118.865215 192.168.1.11 -> 10.100.0.111 DNS
> Standard
> > query response, No such name
> > 416.711408 10.100.0.111 -> 192.168.1.246 SIP
> Status:
> > 478 Unresolveable destination (478/TM)
> > 407.921364 192.168.1.246 -> 10.100.0.111 SIP
> Request:
> > ACK sip:18005551212@sip.acme.com
> >
> >
>
***********************************************************************************
> >
> >
> > *************
> > after rewritehostport("inside.acme.com:5060"), SER
> > only does an A lookup
> > ***************
> >
> > 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP
> > Request: INVITE sip:18005551212@nat.foo.com, with
> > session descripti
> > on
> > 415.971953 10.100.0.111 -> 192.168.1.246 SIP
> Status:
> > 100 trying -- your call is important to us
> > 127.006508 10.100.0.111 -> 192.168.1.11 DNS
> Standard
> > query A inside.foo.com
> > 127.007557 192.168.1.11 -> 10.100.0.111 DNS
> Standard
> > query response A 192.168.1.211
> > 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP
> > Request: INVITE
> sip:18005551212@inside.foo.com:5060,
> > with session descriptio
> > n
> > 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP
> > Request: INVITE
> sip:18005551212@inside.foo.com:5060,
> > with session descriptio
> > n
> >
> >
>
******************************************************************
> >
> > How can I get SER to consistently do SRV lookup
> after
> > rewriting of R-URI?
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
