Hello,
I really need help with Serweb.
I use ser 0.10.99-dev5 with latest Serweb from CVS but
I can't use most of features missed call,
message stored, account, ...
I spent time with apache ser and mysql logs but I
can't fix problems.
I read INSTALL, mailing list, ...
I can get missed calls because of I changed a php
script in data_layer.
Either my serweb configuration is wrong or Serweb
need really some fixings.
Regards
Harry
Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails !
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Hi All.
The AVPOPS documentation says that any header can be written to an AVP.
For example
avp_write("$hdr[My-header]","i:11");
This would right the body of the "My-header" header to AVP i:11
But this does not work.
I want to write the Call-ID header to an AVP. How do I do this?
Regards,
Paul
I have a SER Proxy that is routing calls to a SIP Redirector which then
informs SER as to which gateway to send calls to.
GW1 GW2
| |
SERProxy------->SIP Redirector
|
IP Phone
SER is currently sending the redirect to the IP Phone.. .which is
unauthorized to send SIP messages to the gateway.
I would like for SER to take a 302 redirect and then proxy the call to
the gateway per the redirect..
How do I configure SER to actually proxy the call per the redirect on
behalf of the IP Phone?
Thank you very much Samuel I didn't know that we should add this line to
the ser.cfg.
If we don't want to use aliases how should I avoid ser use it?
Regards
Alberto Cruz
Samuel Osorio Calvo wrote:
>The problem is that you don't have
> lookup("aliases")
>in your config file and therefore SER thinks you are not using aliases
>and do not load the aliases tables.
>
>To enable aliases you have to use the lookup("aliases") somewhere in
>your config file.
>
>I think this issue has been solved somewhere in the mailing list,
>
>Samuel.
>
>
>Unclassified.
>
>
>>>>Alberto Cruz <acruz(a)tekbrain.com> 03/22/05 05:15PM >>>
>>>>
>>>>
>Please list let me know if someone is receiving this message, at least
>
>give me a clue where to search for information about this error.
>
>I really need your help.
>
>Regards
>
>-------- Original Message --------
>Subject: getting "error: 400; check if you use aliases in SER"
>when
>adding a new user with serctl
>Date: Mon, 21 Mar 2005 16:25:55 -0600
>From: Alberto Cruz <acruz(a)tekbrain.com>
>To: 'serusers(a)lists.iptel.org' <serusers(a)lists.iptel.org>
>
>
>
>Hi list maybe this is an stupid question but I can't find any clue what
>
>I'm doing wrong.
>
>When I try to add a new user using serctl add I'm getting the following
>
>error:
>error: 400; check if you use aliases in SER
>
>What does "error: 400" mean? I have looked it at messages and I'm no
>receiving any log information. I'm using two aliases at my ser.cfg
>I'm attaching my ser.cfg
>
>How should I handle this error?
>
>Regards
>
>Alberto Cruz
>
>
>
>
>
>
Nicolas,
Most probably you pass some invalid parameter to the function - note you
have to pass a compiled regexp to the function - see the fixup function
for replace_all().
Best regards,
Marian
Nicolas Fauvel wrote:
> May be but here is an amazing thing:
>
> when I use this funtion from routing script defined in ser.cfg, it works. I
> haven't modified textops sources...
> The process crashes during the execution of the regex() function.
>
> Any idea ?
>
> --- Marian Dumitru <marian.dumitru(a)voice-sistem.ro> wrote:
>
>>Hi Nicolas,
>>
>>Most probably you got a segmentation fault somewhere before due your
>>code changes. What you see is a side effect from the FIFO server process.
>>
>>Best regards,
>>Marian
>>
>>Nicolas Fauvel wrote:
>>
>>>Hi,
>>>
>>>I want to use the exported replace_all function in textops module, but as
>>
>>soon
>>
>>>as the function is called:
>>>
>>> 2(4848) ERROR: fifo_server fgets failed: Illegal seek
>>> 2(4848) ERROR: fifo_server fgets failed: Illegal seek
>>> 2(4848) ERROR: fifo_server fgets failed: Illegal seek
>>> 2(4848) ERROR: fifo_server fgets failed: Illegal seek
>>> 2(4848) INFO: signal 15 received
>>>
>>>What's happening ? I can't understand.
>>>
>>>Thanks,
>>>
>>>Nicolas
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos
>>
>>mails !
>>
>>>Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/
>>>
>>>_______________________________________________
>>>Serusers mailing list
>>>serusers(a)lists.iptel.org
>>>http://lists.iptel.org/mailman/listinfo/serusers
>>>
>>
>>--
>>Voice System
>>http://www.voice-system.ro
>>
>
>
>
>
>
>
>
> Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails !
> Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/
--
Voice System
http://www.voice-system.ro
I am involved in a new deployment of the famous SER and Asterisk tandem.
The goal is to use SER as a SIP proxy (what it is) and Asterisk as an IP-PBX
(connection to PSTN and voicemail).
We plan to make extensive use of WiFi phones.
I begin to have "some" understanding of the interplay between SER and
Asterisk, especially when it comes to handle
calls to/from the PSTN as well as transfers to voicemail.
On the other hand, I keep stumbling on the following question:
Besides handling voicemail and interconnection to PSTN, Asterisk offers many
of the traditional PBX features
(for example CallWaiting activated by hookflash or *64 for DoNotDisturb).
For SIP to SIP calls handled solely by SER (unless I am mistaken), how are
these features invoked (or activated)?
What kind of messaging goes on when hookflash or *xx is activated on the
phone?
What are the respective roles and responsibilities of SER and Asterisk in
handling such messaging?
Regards
Yves DeSerres (ydeserres(a)raris.net)
Raris Communications
Hi everybody I having a weird behavior with
Mediaproxy I'm using SER 0.9.1 and Mediaproxy Release 1.2.1
The Gateway is a Cisco AS5300 using release 12.2(15)T
I have a UA using behind a NAT, the PSTN gateway and the SER server both
are using public IP addresses.
The behaviors are the following:
1. When I tried to place a call from a NATed UA to the PSTN gateway I'm
not receiving the call progress tone or ring back tone at the NATed UA.
2. If I decide to wait until the call is completed (doesn't matter if a
heard death air as progress tone) I start hearing the calling
party as soon the call is completed and we can talk each other without
any troubles.
3. If I cancel the call and hang up the phone at the NATed UA before the
call is completed (during the call progress stage) the SER/Mediaproxy
don't cancel the call it still in progress until it is completed or
cancelled by the PSTN.
I'm attaching my ser configuration and the logging I'm getting when I
cancel the call before it's completed.
Please Help me to know how to fix this If I'm making some mistake with
mediaproxy or routing configuration at the ser configuration.
Thanks in advanced.
Regards
Alberto Cruz
Mar 20 00:10:02 matrix proxydispatcher[1379]: command request call-80BF3413-4297-D911-0217-21(a)172.31.254.240 172.31.254.240:10286:audio 200.67.33.247 172.31.254.240 remote 65.208.39.219 remote Quintum/1.0.0 info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:
Mar 20 00:10:02 matrix proxydispatcher[1379]: will use default mediaproxy for this call.
Mar 20 00:10:02 matrix mediaproxy[1376]: command request call-80BF3413-4297-D911-0217-21(a)172.31.254.240 172.31.254.240:10286:audio 200.67.33.247 172.31.254.240 remote 65.208.39.219 remote Quintum/1.0.0 info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:,dispatcher
Mar 20 00:10:02 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: started. listening on 65.208.39.215:35150
Mar 20 00:10:02 matrix mediaproxy[1376]: command execution time: 9.23 ms
Mar 20 00:10:02 matrix proxydispatcher[1379]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '65.208.39.215 35150'
Mar 20 00:10:02 matrix proxydispatcher[1379]: command execution time: 13.13 ms
Mar 20 00:10:06 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73
Mar 20 00:10:06 matrix mediaproxy[1376]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73,dispatcher
Mar 20 00:10:06 matrix mediaproxy[1376]: command execution time: 1.63 ms
Mar 20 00:10:06 matrix proxydispatcher[1379]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '65.208.39.215 35150'
Mar 20 00:10:06 matrix proxydispatcher[1379]: command execution time: 4.88 ms
Mar 20 00:10:06 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: caller signed in from 200.67.33.247:49125 (RTP) (will return to 200.67.33.247:49125)
Mar 20 00:10:06 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: called signed in from 65.208.39.219:18342 (RTP) (will return to 65.208.39.219:18342)
Mar 20 00:10:08 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: caller signed in from 200.67.33.247:49126 (RTCP) (will return to 200.67.33.247:49126)
Mar 20 00:10:11 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: called signed in from 65.208.39.219:18343 (RTCP) (will return to 65.208.39.219:18343)
Mar 20 00:10:13 matrix proxydispatcher[1379]: command delete call-80BF3413-4297-D911-0217-21(a)172.31.254.240 info=
Mar 20 00:10:13 matrix mediaproxy[1376]: command delete call-80BF3413-4297-D911-0217-21(a)172.31.254.240 info=dispatcher
Mar 20 00:10:13 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: 29/44/73 packets, 1380/5320/6700 bytes (caller/called/relayed)
Mar 20 00:10:13 matrix mediaproxy[1376]: session call-80BF3413-4297-D911-0217-21(a)172.31.254.240: ended.
Mar 20 00:10:13 matrix mediaproxy[1376]: command execution time: 1.43 ms
Mar 20 00:10:13 matrix proxydispatcher[1379]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: ''
Mar 20 00:10:13 matrix proxydispatcher[1379]: command execution time: 4.29 ms
Mar 20 00:10:45 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73
Mar 20 00:10:45 matrix proxydispatcher[1379]: warning: trying to lookup session with non-existent id: 'call-80BF3413-4297-D911-0217-21(a)172.31.254.240'
Mar 20 00:10:45 matrix proxydispatcher[1379]: command execution time: 0.91 ms
Mar 20 00:10:45 matrix /usr/local/sbin/ser[11723]: error: use_media_proxy(): empty response from mediaproxy
Mar 20 00:10:45 matrix /usr/local/sbin/ser[11723]: ERROR: on_reply processing failed
Mar 20 00:10:46 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73
Mar 20 00:10:46 matrix proxydispatcher[1379]: warning: trying to lookup session with non-existent id: 'call-80BF3413-4297-D911-0217-21(a)172.31.254.240'
Mar 20 00:10:46 matrix proxydispatcher[1379]: command execution time: 1.08 ms
Mar 20 00:10:46 matrix /usr/local/sbin/ser[11725]: error: use_media_proxy(): empty response from mediaproxy
Mar 20 00:10:46 matrix /usr/local/sbin/ser[11725]: ERROR: on_reply processing failed
Mar 20 00:10:47 matrix proxydispatcher[1379]: command lookup call-80BF3413-4297-D911-0217-21(a)172.31.254.240 65.208.39.219:18342:audio 65.208.39.219 172.31.254.240 remote 65.208.39.215 unknown Cisco-SIPGateway/IOS-12.x info=from:8412@172.31.254.240,to:018183324166@65.208.39.215,fromtag:ac1ffef0-20,totag:711504-73
Mar 20 00:10:47 matrix proxydispatcher[1379]: warning: trying to lookup session with non-existent id: 'call-80BF3413-4297-D911-0217-21(a)172.31.254.240'
Mar 20 00:10:47 matrix proxydispatcher[1379]: command execution time: 1.09 ms
Mar 20 00:10:47 matrix /usr/local/sbin/ser[11723]: error: use_media_proxy(): empty response from mediaproxy
Mar 20 00:10:47 matrix /usr/local/sbin/ser[11723]: ERROR: on_reply processing failed
#
# $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $
#
#
#
# ----------- 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
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=65.208.39.215
port=5060
alias=65.208.39.215
alias=sip.telereunion.com.mx
children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# ------------------ 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/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.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"
# ----------------- 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)
modparam("domain", "db_mode", 1)
modparam("auth_db|usrloc|domain|group", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("group", "use_domain", 0)
# -- 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")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("mediaproxy", "natping_interval", 60)
modparam("registrar", "nat_flag", 2)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
if (method!="ACK") {
sl_send_reply("483","Too Many Hops");
};
break;
};
if (msg:len >= 2048 ) {
if (method!="ACK") {
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
if (!method=="REGISTER") record_route();
if (method=="REGISTER") {
if (uri==myself || is_from_local()) {
# Mark as NAT'ed
if (client_nat_test("3")) {
setflag(2);
force_rport();
fix_contact();
};
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
break;
} else if (!check_to()) {
sl_send_reply("403", "Username!=To not allowed");
break;
};
if (!save("location")) {
sl_reply_error();
};
} else {
append_hf("P-hint: outbound alias\r\n");
sl_send_reply("403", "This domain is not served here");
};
break;
};
if (method=="INVITE") {
if (!(is_from_local() || uri==myself || is_uri_host_local())) {
sl_send_reply("403", "Relaying is forbidden");
break;
};
t_on_failure("1");
} else if (method == "BYE" || method == "CANCEL") {
end_media_session();
};
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
# The following lines are added due media proxy
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
# end media session for BYE and CANCEL is done above
# before entering the loose route. no need to call it here
t_relay();
break;
};
if (client_nat_test("3") && !search("^Record-Route:")) {
# Mark as NAT'ed
force_rport();
fix_contact();
};
### Begin PSTN evaluation
if (method=="INVITE") {
if (uri=~"sip:01[1-9][0-9]+@.*") {
if (!is_user_in("From", "ld")) {
sl_send_reply("403", "LD permissions needed");
break;
};
rewritehostport("65.208.39.219:5060");
} else if (uri=~"sip:00[1-9][0-9]+@.*") {
if (!is_user_in("From", "int")) {
sl_send_reply("403", "International permissions needed");
break;
};
rewritehost("65.208.39.219");
};
t_on_reply("1");
};
### End PSTN evaluation
if (is_uri_host_local() || uri==myself) { # join with next if?
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "User not found");
break;
};
};
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
failure_route[1] {
end_media_session();
}
onreply_route[1] {
if (status=~"(180)|(183)|(2[0-9][0-9])") {
if (client_nat_test("1")) {
fix_contact();
};
use_media_proxy();
};
}
Hello.
When i start the STUN server from iptel (mystun) with this command line :
% ./server.exe -D
as a deamon.
How can i check which is my bind address and my alternative bind address?
Thanks!
Ricardo.-