hi, all,
i want to call from pstn to the agent which registers at ser.
but when I call the sip agent from PSTN, the CISCO gateway get the 500
server errror from ser.
d05h: Received:
SIP/2.0 500 I'm terribly sorry, server error occured (1/SL)
Via: SIP/2.0/UDP 192.168.15.41:5060
From: "8010" <sip:8010@192.168.15.41>;tag=6483474-1312
To: <sip:8275@172.30.10.230;user=phone>;tag=d907c03782364451
b
Call-ID: EC4BBA87-15E311CC-82EADFF8-C6A5744F(a)192.168.15.41
CSeq: 101 INVITE
Server: Sip EXpress router (0.8.10 (i386/linux))
Content-Length: 0
Warning: 392 172.30.10.230:5060 "Noisy feedback tells: pid=2
168.15.41 in_uri=sip:8275@172.30.10.230:5060;user=phone out_
10.97 via_cnt==1"
and my sip agent can get the invite message, and the cisco gw can get my 180
after 500 error.
i have the following line in my ser.cfg
if(src_ip==192.168.15.41){
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
t_relay();
};
also I have add the usr and usr contact in the mysql db.
BTW, how can I read the debug information. In my ser.cfg, I set debug = 5;
Best Regards,
Jade
begin 666 ATT00084.htm
M/"%$3T-465!%($A434P@4%5"3$E#("(M+R]7,T,O+T141"!(5$U,(#,N,B\O
M14XB/@T*/$A434P^#0H\2$5!1#X-"CQ-151!($A45% M15%5258](D-O;G1E
M;G0M5'EP92((a)0T].5$5.5#TB=&5X="]H=&UL.R!C:&%R<V5T/6=B,C,Q,B(^
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M;W(@9G)O;2!S97(N/"]&3TY4/@T*/"]0/(a)T*#0H\4#X\1D].5"!325I%/3(^
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M4CX\1D].5"!325I%/3(^0V%L;"U)1#H(a)14,T0D)!.#<M,35%,S$Q0T,M.#)%
M041&1C@M0S9!-3<T-$9 ,3DR+C$V."XQ-2XT,3PO1D].5#X-"@T*/$)2/CQ&
M3TY4(%-)6D4],CY#4V5Q.B Q,#$@24Y6251%/"]&3TY4/@T*#0H\0E(^/$9/
M3E0@4TE:13TR/E-E<G9E<CH@4VEP($58<')E<W,@<F]U=&5R("@P+C@N,3 @
M*&DS.#8O;&EN=7(a)I*3PO1D].5#X-"@T*/$)2/CQ&3TY4(%-)6D4],CY#;VYT
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M/CQ&3TY4(%-)6D4],CXF;F)S<#L@=%]R96QA>2(a)I.SPO1D].5#X-"@T*/$)2
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B93PO1D].5#X-"CPO4#X-"@T*/"]"3T19/@T*/"](5$U,/@``
`
end
Folks,
This question isn't directly related to a SER, but does anybody know why
Cisco IOS inserts several identical `c=<IP>' lines in SDP which it
generates? No other UA I know about does that. I wonder because it
creates a problems with SDP rewriting in nathelper module and I need to
decide whether I have to modify nathelper to cope with that or open a
Cisco TAC case instead.
Any comments are appreciated!
-Maxim
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.104.130.102:5061
From:
<sip:0118000@67.104.130.102:5061;user=phone>;tag=bb4b39670c51e5c737c5de99b5450cc5
To: <sip:61412722777@64.180.102.72;user=phone>;tag=357A3908-2F6
Date: Tue, 17 Jun 2003 07:35:12 GMT
Call-ID: 2920698556(a)192.168.1.100
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 2 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:61412722777@67.104.130.107:5060>
Content-Type: application/sdp
Content-Length: 285
v=0
o=CiscoSystemsSIP-GW-UserAgent 7069 1539 IN IP4 67.104.130.107
s=SIP Call
c=IN IP4 67.104.130.107
^^^^^^^^^^^^^^^^^^^^^^^
t=0 0
m=audio 19044 RTP/AVP 4 19 101
c=IN IP4 67.104.130.107
^^^^^^^^^^^^^^^^^^^^^^^
a=rtpmap:4 G723/8000
a=rtpmap:19 CN/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Hi,
Should the server produce a 100 - TRYING (Stateful)/200 - OK
(Stateless) to the calling party when an OPTIONS message is sent to it.
[11.2 processing of OPTIONS request @ rfc3261 - "The response code
chosen MUST be the same that would have been chosen had the request been
an INVITE"]
Currently the server just forwards the OPTIONS message (to the called
party) without sending a (200/100) back to the calling party.
Thanks,
Platform: GNU/Linux 2.4.18-14 i686
Server: Sip EXpress router (0.8.11pre29 (i386/linux))
-- Start OPTIONS message --
2003/06/19 12:40:05.819450 192.1.2.88:5060 -> 192.1.2.17:5060
OPTIONS sip:9727610001@192.1.2.17 SIP/2.0.Via: SIP/2.0/UDP
192.1.2.88:5060;
branch=z9hG4bKkdjuw.Max-Forwards: 70.From: "9727619271"
<sip:9727619271@
192.1.2.88>;tag=323.To: sip:9727610001@192.1.2.17.Call-ID:
517928895(a)192.1.
2.88.CSeq: 1 OPTIONS..
-- End OPTIONS message --
I installed ser yesterday and I was trying to get an UA to connect to it. I created just one user:
<sip:marcelo@marcelo.homeunix.com> and used an iptel.org account to send a message, I got:
483 Too Many Hops. I attached the ser.cfg and a capture file from tcpdump.
Thanks,
Marcelo.
Hi,
I have just upgraded our ser 0.8.10, which worked fine, to ser 0.8.011Pre29,
with the Jabber and PA modules running.
The problems I am encountering are:
when in debug mode I often get error messages: XJAB:xjab_check_workers:
worker has exited, status=0, err=-1, errno=10. I have not found the meaning
of this error code.
More serious: ser seems to create many processes (up to more than a hundred
after some hours of use), which after a day or so crashes the servers it
runs on (Linux Redhat 9, bi PIII-500). Do you know where this can come from
?
Thanks in advance,
Damien Vicq
Bell Labs
Hi,
I have been running ser-0.8.10/FreeBSD-4.7 for 6 months
without problem. I changed ser default port,i.e.,5060 to
5065 on ser.cfg because some sip phone needs 5060 for
their STUN/NAT traversal now. It works fine for sip hardphones
using UDP:5065 of ser for REGISTER, but MSN Messenger4.6
on windows98 could not REGISTER eventhough the REGISTER
packet was sent from the Messenger4.6 to the ser's 5065
as bellow;
-------------------sample -------------------
U 2003/06/19 10:10:08.426264 192.168.0.21:1079 -> 192.168.0.26:5065
REGISTER sip:tel.no-ip.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.21:14236..Fr
om: <sip:050021@tel.no-ip.com>;tag=acbb1c1c-a23c-11d7-8627-0090997e7578..To
: <sip:050021@tel.no-ip.com>..Call-ID: acbb1c1d-a23c-11d7-8627-0090997e7578
@192.168.0.21..CSeq: 2 REGISTER..Contact: <sip:192.168.0.21:14236>;methods=
"INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"..User
-Agent: Windows RTC/1.0..Expires: 0..Content-Length: 0....
--------------------sample end-----------------
Messenger4.6 does not accept "aaa@foo.bar.com:5065" and never
change UDP port:
REGISTER sip:foo.bar.com
From: <sip:aaa@foo.bar.com>
To: <sip:aaa@foo.bar.com>
I tried "iptel.org SIP Express Router v0.8.10 -- Admin's Guide,
Troubleshooting,Q: Windows Messenger authentication fails. "
but it does not work.
Does someone know how to register Messenger4.6 to non-default
port of sip proxy?
Regards,
Zen Kato
Hello:
I have some Cisco phones registered to SER running on
FreeBSD 4.7. Users of these phones want to dial a telephone
number and have the call correctly routed out to the PSTN
via a Cisco gateway on the network. The users of course
only key-in a telephone number. No domain name or
IP address. I thought the following code would correctly
append the IP address and port of the gateway to their
telephone number so the call could get routed correctly. This
is not working. Instead either the IP address of the calling
phone or the domain name for my domain is appended. Any
thought on how to correct this would be appreciated.
# check if it's about PSTN destinations through our gateway;
if (uri=~"sip:9{10}")
{
route(4);
break;
};
route[4]
.....
some account statements followed by
.....
rewritehostport("ip-address-of-pstn-gateway:5060");
Thanks,Steve