Hello
I use UAC module to route my international calls from SER to IP-COM mainly
for testing. The call is setup and it rings on my PSTN phone however after
20 seconds the call terminates. A classical ACK problem. So I did some
sniffing with NGREP and sees that there is actually an ACK being sent but it
is not accepted by the provider. It seems that the ACK from header does not
match the OK from header, could this be the cause.
This is the relevant config lines and route blocks for my setup, if uri is
an international phonenumber processing is sent to route block 7 below. At
the bottom you see the 200 ok and following ACK that dont work.
Any idea or pointers on what to change? I spent all day without detecting
where the problem is...
modparam("uac","credential","000000:sip.ip-com.co.uk:xxxxxxxxxx")
modparam("uac","from_store_param","xparam")
modparam("uac","from_restore_mode", 1)
route[1] {
# -----------------------------------------------------------------
# Default Message Handler
# -----------------------------------------------------------------
t_on_reply("1");
if (!t_relay()){
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[6] {
#
------------------------------------------------------------------------
# ACK Handler
#
------------------------------------------------------------------------
lookup("aliases");
if (uri!=myself) {
route(1);
break;
};
lookup("location");
route(1);
}
route[7] {
# -----------------------------------------------------------------
# PSTN Handler - International calls (Using UAC)
# -----------------------------------------------------------------
t_on_failure("1");
resetflag(1);
uac_replace_from("Serverhallen","sip:000000@sip.ip-com.co.uk");
#000000 is just to hide actul no.
rewritehost("sip.ip-com.co.uk");
route(1);
}
onreply_route[1] {
if ((isflagset(6) || isflagset(7)) &&
(status=~"(180)|(183)|2[0-9][0-9]")) {
if (!search("^Content-Length:\ +0")) {
use_media_proxy();
};
};
if (client_nat_test("1")) {
fix_nated_contact();
};
}
failure_route[1] {
if ( t_check_status("401|407") )
{
if (isflagset(1))
{
t_reply("503","Authentication failed");
break;
}
if (uac_auth())
{
setflag(1);
t_on_failure("1");
append_branch();
t_relay();
}
}
}
#
U 195.60.16.41:5060 -> 212.247.91.237:5060 SIP/2.0 200 OK.
Via: SIP/2.0/UDP 212.247.91.237;branch=z9hG4bK18cf.5a0c84f7.0,SIP/2.0/UDP
82.182.194.198:5062;branch=z9hG4bK-1098303a.
From: "Serverhallen"
<sip:000000@sip.ip-com.co.uk>;tag=dc0d090b4f97be8eo0;xparam=c2lwOjMzMDAwMUBz
aXAuc2VydmVyaGFsbGVuLmNv.
To: <sip:004640240252@sip.serverhallen.com>;tag=1C3FFA90-8D9.
Date: Tue, 26 Jul 2005 12:43:40 GMT.
Call-ID: c289f48f-a44228da(a)172.16.126.108.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 102 INVITE.
Contact: <sip:2383004640240252@213.228.196.221:5060;user=phone>.
Record-Route:
<sip:004640240252@195.60.16.41;ftag=dc0d090b4f97be8eo0;lr=on>,<sip:212.247.9
1.237;ftag=dc0d090b4f97be8eo0;lr=on>.
Content-Type: application/sdp.
Content-Length: 214.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 2038 8238 IN IP4 213.228.196.221.
s=SIP Call.
c=IN IP4 213.228.196.221.
t=0 0.
m=audio 17790 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 .
#
U 212.247.91.237:5060 -> 195.60.16.41:5060 ACK
sip:2383004640240252@213.228.196.221:5060;user=phone SIP/2.0.
Record-Route: <sip:212.247.91.237;ftag=dc0d090b4f97be8eo0;lr=on>.
Via: SIP/2.0/UDP 212.247.91.237;branch=0.
Via: SIP/2.0/UDP 82.182.194.198:5062;branch=z9hG4bK-7d2fae01.
From: Roger <sip:330001@sip.serverhallen.com>;tag=dc0d090b4f97be8eo0.
#SHOULD NOT BE THE SAME FROM AS IN THE OK?
To: <sip:004640240252@sip.serverhallen.com>;tag=1C3FFA90-8D9.
Call-ID: c289f48f-a44228da(a)172.16.126.108.
CSeq: 102 ACK.
Max-Forwards: 16.
Route: <sip:004640240252@195.60.16.41;ftag=dc0d090b4f97be8eo0;lr=on>.
Proxy-Authorization: Digest
username="330001",realm="sip.serverhallen.com",nonce="42e629ae8328119b5bd5fc
16ab4d0bc512c0db0c",uri="sip:004640240252@sip.serverhallen.com",algorithm=MD
5,response="00556d4ebdec58dc9156e61a61654028".
Contact: Roger <sip:330001@82.182.194.198:5062>.
User-Agent: Linksys/PAP2-2.0.12(LS).
Content-Length: 0.
Hi!
1. Please always Cc: to the mailing list - others might also be
interested in this topic and may help you.
2. I'm not sure if I understand the problem correctly
The SIP Proxy should forward all INVITEs to the corresponding SIP
client. Then the SIP client may choose to receive a second sessions
(voice, video or IM) or not.
regards,
klaus
Abhijit A.Mahajani wrote:
> Thanks Klaus!
> My basic problem is:
> If a SIP call has been established by an UAC (say windows msgr 5.0). Hence
> Now All Instant msgs sent by windows msgr will be processed by the SIP
> server.Now if some other UAC tries to send IM , then since that IM would
> contain INVITE , should our SIP server process that IM? I think answer
> should be know...SIP server needs to check the two Call IDs or not?
> Windows msgr 5.0 sends IM with SDP=x-ms-message but if other UAC does not
> send that then will that be OK?
> If It is OK, then should SIP server entertain the SIP IM if its call id and
> the call id in the session INVITEd by first UAC are different?
>
> Thanks and Regards.
> Abhijit
>
>
> ----- Original Message -----
> From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at>
> To: "Abhijit A.Mahajani" <abhijit_m(a)citilindia.com>
> Cc: <serusers(a)lists.iptel.org>
> Sent: Tuesday, July 26, 2005 1:58 PM
> Subject: Re: [Serusers] Queries regarding SIP IM
>
>
>
>>Abhijit A.Mahajani wrote:
>>
>>>I have few doubts regarding SIP IM:
>>>1.Can SIP IM initiate a SIP dialog on its own?
>>
>>The RFC states, that MESSAGE does not create a dialog. Nevertheless, in
>>windows Messegner 4.6/4.7, the MESSAGE created a dialog.
>>
>>
>>>2.Without any INVITE or any Call established can SIP ua send a SIP IM?
>>
>>Yes.
>>
>>
>>>3.Like Windowds Msgr , For the First SIP IM , x-ms-invite is sent which
>>>iniates the converation, But These session should be treated as same
>>>media session would have been iniated by any other INIVTE?
>>
>>Where das "x-ms-invite" appear? AFAIK Windows Messenger 5.0 creates
>>first a dialog using INVITE and an SDP with "m=x-ms-message". Of course
>>these INVITE should be treated as any other INVITE, except that you need
>>not enforce the rtpproxy (in case of NAT traversal)
>>
>>
>>>4.How should SIP Server respond to the Normal INIVITE and x-ms-invite?
>>>should treat same or diffrerent?
>>
>>same
>>
>>regards,
>>Klaus
>>
>>
>>>Hoping for your reply.
>>>with Regards.
>>>Abhijit
>>>
>>>
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>Serusers mailing list
>>>serusers(a)lists.iptel.org
>>>http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
Hello:
I'm trying to build the pa module on Redhat ES4 with SER 0.9.4-rc1. I
keep
encountering the following errors. These seem to be due to incomplete
variable
definitions in the pa module makefile. Before I begin making changes has
anyone
else experienced these errors? If so and you corrected them can you
explain how?
Thanks.
make[1]: Entering directory `/usr/sip_router/modules/pa'
../../Makefile.rules:80: pidf.d: No such file or directory
../../Makefile.rules:80: publish.d: No such file or directory
publish.c:57:27: libxml/parser.h: No such file or directory
publish.c:58:26: libxml/xpath.h: No such file or directory
pidf.c:32:27: libxml/parser.h: No such file or directory
pidf.c:33:26: libxml/xpath.h: No such file or directory
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
Hello, Guys:
i am trying to use acc modules to get call records. I reloaded acc
modules and did modparam. but there is still not data in acc table in
mysql. Does anyone konw how to check the records?
thanks lot!
lizhong
Thanks Klaus!
My basic problem is:
If a SIP call has been established by an UAC (say windows msgr 5.0). Hence
Now All Instant msgs sent by windows msgr will be processed by the SIP
server.Now if some other UAC tries to send IM , then since that IM would
contain INVITE , should our SIP server process that IM? I think answer
should be know...SIP server needs to check the two Call IDs or not?
Windows msgr 5.0 sends IM with SDP=x-ms-message but if other UAC does not
send that then will that be OK?
If It is OK, then should SIP server entertain the SIP IM if its call id and
the call id in the session INVITEd by first UAC are different?
Thanks and Regards.
Abhijit
>
> ----- Original Message -----
> From: "Klaus Darilion" <klaus.mailinglists(a)pernau.at>
> To: "Abhijit A.Mahajani" <abhijit_m(a)citilindia.com>
> Cc: <serusers(a)lists.iptel.org>
> Sent: Tuesday, July 26, 2005 1:58 PM
> Subject: Re: [Serusers] Queries regarding SIP IM
>
>
> > Abhijit A.Mahajani wrote:
> > >
> > > I have few doubts regarding SIP IM:
> > > 1.Can SIP IM initiate a SIP dialog on its own?
> >
> > The RFC states, that MESSAGE does not create a dialog. Nevertheless, in
> > windows Messegner 4.6/4.7, the MESSAGE created a dialog.
> >
> > > 2.Without any INVITE or any Call established can SIP ua send a SIP IM?
> >
> > Yes.
> >
> > > 3.Like Windowds Msgr , For the First SIP IM , x-ms-invite is sent
which
> > > iniates the converation, But These session should be treated as same
> > > media session would have been iniated by any other INIVTE?
> >
> > Where das "x-ms-invite" appear? AFAIK Windows Messenger 5.0 creates
> > first a dialog using INVITE and an SDP with "m=x-ms-message". Of course
> > these INVITE should be treated as any other INVITE, except that you need
> > not enforce the rtpproxy (in case of NAT traversal)
> >
> > > 4.How should SIP Server respond to the Normal INIVITE and x-ms-invite?
> > > should treat same or diffrerent?
> >
> > same
> >
> > regards,
> > Klaus
> >
> > >
> > > Hoping for your reply.
> > > with Regards.
> > > Abhijit
> > >
> > >
> > >
> >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
>
Hello,
please use mailing list serusers(a)lists.iptel.org or serweb-users(a)lists.iptel.org for
your further questions.
Yes I encountered this error. Try set these config variables in file
config/config_data_layer.php:
/**
* Needs to be set when you are useing MySQL >= 4.1
* see mysql manual for more info
*
* $config->data_sql->collation = "utf8_general_ci";
*/
$config->data_sql->collation = "";
/**
* Set to true when you are useing MySQL >= 4.1
* This option set mysql system variables character_set_client,
* character_set_connection, and character_set_results to charset
* used in serweb
*
* $config->data_sql->set_charset = true;
*/
$config->data_sql->set_charset = false;
regards Karel
seehoe yee napsal(a):
> Good day Karel,
>
> I'm a new SER user, I've successfully installed SERweb but after logging
> into SERweb I was displayed with
> DB Error: unknown error
>
> on my_account.php and pages like accounting.php, phonebook.php
>
> I've logged the error below is the log:
>
> # -----
> Jul 27 03:25:09 serweb [debug] Html form was not assigned to APU
> apu_aliases3. Useing default.
> Jul 27 03:25:09 serweb [debug] Html form was not assigned to APU
> apu_acl4. Useing default.
> Jul 27 03:25:09 serweb [error] file:
> /var/www/html/serweb/application_layer/apu_aliases.php:112: DB Error:
> unknown error - select username, domain from aliases where
> lower(contact)=lower('sip:77771111@192.168.0.106') order by username
> [nativecode=1267 ** Illegal mix of collations
> (latin1_swedish_ci,IMPLICIT) and (utf8_general_ci,COERCIBLE) for
> operation '=']
> # -----
>
> I've google through thought it was MySQL problem as I searched "Illegal
> mix of collations". Therefore, I upgrade from 4.1.12 -> 4.1.13 but still
> no go. The same error message I get.
>
> Just wondering if you have encountered this issue before?
>
> Regards,
> See Hoe
>
I have few doubts regarding SIP IM:
1.Can SIP IM initiate a SIP dialog on its own?
2.Without any INVITE or any Call established can SIP ua send a SIP IM?
3.Like Windowds Msgr , For the First SIP IM , x-ms-invite is sent which iniates the converation, But These session should be treated as same media session would have been iniated by any other INIVTE?
4.How should SIP Server respond to the Normal INIVITE and x-ms-invite? should treat same or diffrerent?
Hoping for your reply.
with Regards.
Abhijit
Hi,
I am running redhat 9.0, php version 4.2.2 and each
time I try to load the serweb page, I get the
following error:
Function aggregate_methods() doesn't exists. Try
install Classkit extension.
http://pecl.php.net/package/classkit
Great, I go the web site pecl.php.net and I download
the stuff. But guess what, I can't figure out how to
install it because all it has is a .c, .h. .m4 file.
So I read a posting on newsgroup and go and downloag
the php-pecl rpm from iptel.org. But guess what, I try
to install it and it says that it requires "php
>=5.0.3". Can someone please guide me.
____________________________________________________
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Here is the b2bua log
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