Hi!
I have never used the command line to use the t_uac_dlg fifo command, but just a script similar to the example present in the attached file FIFO.txt (it is from an older version of SER but the FIFO interface has not changed).
I also attach another document containing the explanation found in the source code about the FIFO interface.
I hope with these two documents you will be able to send SIP requests with t_uac_dlg
Samuel.
Unclassified.
>>> "michael p" <mikep3000(a)hotmail.com> 05/25/05 04:32PM >>>
i'm sorry but it didn't help me
i think it's different with serctl but thanks
i don't understand anybody never used this command line in serctl???
thanks
>From: "Greger V. Teigre" <greger(a)teigre.com>
>To: "michael p" <mikep3000(a)hotmail.com>, <ladia6(a)centrum.cz>,
><serusers(a)lists.iptel.org>
>Subject: Re: [serusers] fifo
>Date: Wed, 25 May 2005 16:26:44 +0200
>
>Maybe this can help you in the right direction:
>http://www.onsip.org/modules/xoopsfaq/index.php?cat_id=3#q2
>g-)
>
>michael p wrote:
>>only in the third link ther is my question but there is no answer
>>Does anyone know what correct format of a packet
>>should be pushed into fifo buffer? For example,
>>
>>serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx
>>'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx'
>>'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . .
>>The above command I've tried,but got errors.
>>
>>
>>and i tried the format into the serctl too
>>
>>
>>thanks for replying
>>
>>
>>
>>>From: Ladislav Andel <ladia6(a)centrum.cz>
>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz>
>>>To: "michael p" <mikep3000(a)hotmail.com>
>>>Subject: Re: [serusers] fifo
>>>Date: Wed, 25 May 2005 15:39:07 +0200
>>>
>>>I'm sorry, I think I've seen it in the list, but can't find it..
>>>
>>>try the links below
>>>
>>>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html
>>>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html
>>>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html
>>>
>>>mp> thanks but it didn't help me cause this topic not found in
>>>google in this
>>>mp> name maybe another one
>>>
>>>mp> so if you know which line i have to write it will be more simple
>>>
>>>mp> thanks
>>>
>>>
>>>
>>>>>From: Ladislav Andel <ladia6(a)centrum.cz>
>>>>>Reply-To: Ladislav Andel <ladia6(a)centrum.cz>
>>>>>To: "michael p" <mikep3000(a)hotmail.com>
>>>>>CC: serusers(a)lists.iptel.org
>>>>>Subject: Re: [serusers] fifo
>>>>>Date: Wed, 25 May 2005 14:55:45 +0200
>>>>>
>>>>>This topic was in the list recently. Try to google it out.
>>>>>Lada
>>>>>
>>>>>mp> i already saw the different options but i want to know how to
>>>>>use them to
>>>>>mp> send a message like INVITE or OPTIONS i don't know how i have
>>>>>to write the
>>>>>mp> commande
>>>>>
>>>>>mp> thanks
>>>>>
>>>>>>>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com>
>>>>>>>To: mikep3000(a)hotmail.com
>>>>>>>CC: serusers(a)lists.iptel.org
>>>>>>>Subject: RE: [serusers] fifo
>>>>>>>Date: Wed, 25 May 2005 14:20:27 +0200
>>>>>>>
>>>>>
>>>>>
>>>
>>>
>>>
>>
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hello List.
I have a question regarding the use of avp_radius and avpops. I'm
using avp_radius to obtain an AVP value from my database via radius. What i
what to do is replace this value for the RURI.
Here i have a couple of questions.
1.- The value returned by the avp_radius (the SIP-AVP) where is stored ? It
suppose that the SIP-AVP returned by radius has the form of "name:value".
That "name" refers to the name of what?.
For example i'm returning : "var1:sip:1234567@mydomain.com". What i see in
the debug is :
avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has been added
This is what i got in my ser.cfg (a snippet).
if (method=="INVITE" || method=="CANCEL") {
if( !avp_load_radius("caller")) {
log (1, "AVP_RADIUS: Fail on avp_radius\n");
};
if( !avp_pushto("$RURI", "s:var1/g")) {
log (1, "AVPOPS: Fail on AVPOPS\n");
};
};
Again the debug = 9 .
6(23815) avp_load_user: AVP 'var1'='sip:1234567@mydomain.com' has been
added
6(23815) qm_free(0x8123400, 0x8166ccc), called from avp_radius.c:
load_avp_user(344)
6(23815) qm_free: freeing frag. 0x8166cb4 alloc'ed from avp_radius.c:
load_avp_user(330)
6(23815) DEBUG:avpops:pushto_avp: no avp found
6(23815) AVPOPS: Fail on AVPOPS
What i'm doing wrong?
Thanks!
Regards
Ricardo Martinez
I got the config file someone posted to this list a couple of days back and
it fits almost perfectly
for what we need ser to do (voicemail, with asterisk or similar AND nat
using mediaproxy).
However it required a few other modules that dont come with the standard SER
package (I am currently using
0.8.14 - which I have working for a simple config).
0(32034) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/avpops.so>:
0(32034) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/options.so>:
0(32034) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/speeddial.so>:
0(32034) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/uri_db.so>:
0(32034) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/xdz_tools.so>:
Can anyone let me know where these modules are to download or better yet,
email the tar file with them.
I have also tried onsip.org which seems to have some great info on
configuring ser with the same modules above as well but I can't download any
onf the files (and from the FAQ/Forum I see that its not just me but
everyone can't download from them).
Would really appreciate it if someone could help me find the above modules.
Thanks
Tony Dean
Hi all,
I'm using SER 0.8.14, FreeRadius 1.0.2. and XLite 1103. My question is about
MD5, why I can't see Digest-Algorithm = "MD5" in FreeRadius and XLite client
outputs.
Is this a problem in XLite or SER? Everything is configured as in HOW TO and
I get authenticated with FreeRadius so I suppose that configuration is OK.
Dictionaries are the same in radiusclient on SER machine and in FreeRadius.
I can provide more outputs if someone is interested to help.
Any help would be appreciated.
*****This is output from FreeRadius Server****** (no Digest-Algorithm =
"MD5" received)
...
rad_recv: Access-Request packet from host 161.53.0.131:33854, id=131,
length=207
User-Name = "djovanovic.srce"
Digest-Attributes = 0x0a11646a6f76616e6f7669632e73726365
Digest-Attributes = 0x0109737263652e6872
Digest-Attributes =
0x022a3432393439336464646235643930376433653433323765333063366566646566336436
6137643933
Digest-Attributes = 0x040d7369703a737263652e6872
Digest-Attributes = 0x030a5245474953544552
Digest-Response = "263ec6802b382ddf8d58b363fd0629ad"
Service-Type = Sip-Session
Sip-Uri-User = "djovanovic.srce"
NAS-IP-Address = 161.53.0.131
NAS-Port = 5060
Processing the authorize section of radiusd.conf
modcall: entering group authorize for request 0
modcall[authorize]: module "preprocess" returns ok for request 0
modcall[authorize]: module "chap" returns noop for request 0
modcall[authorize]: module "mschap" returns noop for request 0
rlm_digest: Converting Digest-Attributes to something sane...
Digest-User-Name = "djovanovic.srce"
Digest-Realm = "srce.hr"
Digest-Nonce = "429493dddb5d907d3e4327e30c6efdef3d6a7d93"
Digest-URI = "sip:srce.hr"
Digest-Method = "REGISTER"
rlm_digest: Adding Auth-Type = DIGEST
modcall[authorize]: module "digest" returns ok for request 0
...
*****This is output from XLite****** (no Digest-Algorithm = "MD5" sent)
SEND TIME: 203074718
SEND >> 191.153.206.58:5060
INVITE sip:537@srce.hr SIP/2.0
Via: SIP/2.0/UDP
191.153.206.57:5060;rport;branch=z9hG4bK8BB5DE10CC404EEBABE6761213B4AA45
From: davor <sip:djovanovic.srce@srce.hr>;tag=2361756632
To: <sip:537@srce.hr>
Contact: <sip:djovanovic.srce@191.153.206.57:5060>
Call-ID: 146A8327-BCC6-42E6-A027-83EA1832631B(a)191.153.206.57
CSeq: 40898 INVITE
Authorization: Digest
username="djovanovic.srce",realm="srce.hr",nonce="429492e156f0af6f7ae5e821d7
3df9be3e04f360",response="17a54d30cc2d5ecf1cc8e714d40210e0",uri="sip:537@src
e.hr"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 308
Hello all.
I'm new with this and I'm making test with 3 machines in the same network
using SER-SEMS. I receive the REGISTER messages from the users, but I can't
make an INVITE from one user to another.
I've tried with stateless forwarding, with no result.
Please, some help needed. Could anybody tell me what's missing or wrong ?
This is the ser.cfg that i'm using.
Thanks in advance
Ro
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
# Uncomment these lines to enter debugging mode
debug=9
fork=no
log_stderror=yes
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
#loadmodule "/usr/local/lib/ser/modules/dbtext.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
modparam("usrloc", "db_url","sql://ser:heslo@localhost/ser")
modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ----------------- setting module-specific parameters ---------------
modparam("voicemail", "db_url","sql://ser:heslo@localhost/ser")
#modparam("vm", "db_url", "sql://ser:heslo@localhost/ser")
modparam("voicemail", "email_column", "email_address")
modparam("voicemail", "subscriber_table", "subscriber")
modparam("voicemail", "user_column", "username")
modparam("voicemail", "domain_column", "domain")
# ------------------------- request routing logic -------------------
alias="sip.vivophone.com"
alias="201.249.37.26"
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("70")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("sip.vivophone.com", "subscriber")) {
www_challenge("sip.vivophone.com", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
#if (!lookup("location")) {
# sl_send_reply("404", "Not Found");
# break;
#};
# Voicemail specific configuration - begin
if( (!lookup("location")) && (method=="ACK" || method=="INVITE" ||
method=="BYE" || method=="REFER") ) {
if(t_newtran()){
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE" || method=="REFER"){
log("**************** vm start - begin ******************\n");
}
if(method=="BYE"){
log("**************** vm end/refer - begin ******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the media server\n");
t_reply("500","could not contact the media server");
};
log("**************** vm end/refer - end ********************\n");
break;
};
}
else {
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
};
};
# Voicemail specific configuration - end
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
only in the third link ther is my question but there is no answer
Does anyone know what correct format of a packet
should be pushed into fifo buffer? For example,
serctl fifo t_uac_dlg BYE sip:1111 at xxx.xxx.xxx
'sip:from:1111 at xxx.xxx.xxx' 'sip:to:2222 at xxx.xxx.xxx'
'callid:xxxxxxxxxxxx' 'Cseq:xxxxxx' . .
The above command I've tried,but got errors.
and i tried the format into the serctl too
thanks for replying
>From: Ladislav Andel <ladia6(a)centrum.cz>
>Reply-To: Ladislav Andel <ladia6(a)centrum.cz>
>To: "michael p" <mikep3000(a)hotmail.com>
>Subject: Re: [serusers] fifo
>Date: Wed, 25 May 2005 15:39:07 +0200
>
>I'm sorry, I think I've seen it in the list, but can't find it..
>
>try the links below
>
>http://lists.iptel.org/pipermail/serusers/2004-August/010850.html
>http://lists.iptel.org/pipermail/serusers/2004-August/010825.html
>http://lists.iptel.org/pipermail/serusers/2005-April/018897.html
>
>mp> thanks but it didn't help me cause this topic not found in google in
>this
>mp> name maybe another one
>
>mp> so if you know which line i have to write it will be more simple
>
>mp> thanks
>
>
>
> >>From: Ladislav Andel <ladia6(a)centrum.cz>
> >>Reply-To: Ladislav Andel <ladia6(a)centrum.cz>
> >>To: "michael p" <mikep3000(a)hotmail.com>
> >>CC: serusers(a)lists.iptel.org
> >>Subject: Re: [serusers] fifo
> >>Date: Wed, 25 May 2005 14:55:45 +0200
> >>
> >>This topic was in the list recently. Try to google it out.
> >>Lada
> >>
> >>mp> i already saw the different options but i want to know how to use
>them
> >>to
> >>mp> send a message like INVITE or OPTIONS i don't know how i have to
>write
> >>the
> >>mp> commande
> >>
> >>mp> thanks
> >>
> >> >>From: Llanos Serna García-Conde <llanosserna(a)hotmail.com>
> >> >>To: mikep3000(a)hotmail.com
> >> >>CC: serusers(a)lists.iptel.org
> >> >>Subject: RE: [serusers] fifo
> >> >>Date: Wed, 25 May 2005 14:20:27 +0200
> >> >>
> >>
> >>
>
>
>
Hello all,
is it possible to limit the scale of t_on_failure / failure_route[x] to
certain SIP response codes?
In the concrete example, I would like to exclude certain Response Codes
(mainly 486) from trying at a second location.
With best regards,
Martin
