hi
i have installed ser-0.8.10-2.rpm and ser-mysql-0.8.10-2.rpm . and i can run them successfully .the SIP telephone can get thought after i set modparam ("usrloc", "db_mode", 0) ,but when 1 set modparam ("usrloc", "db_mode", 2), i cannot make it . how do i configure that files?
thank you varymuch !
gg
gdut200104024532(a)163.com
2004-08-17
My SER proxy is sending REGISTER tokens downstream to the gateway, after it
receives a REGISTER from UA.
My gateway device is a Cisco AS53XX and does not support REGISTERs. Not
that fork has anythin to do with it but I have fork=no in my ser.cfg file.
Any ideas what it may be trying to accomplish here?
MS
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Hello,
I have troubles with SER routing. It's complaining about a syntax error
on these lines:
if (uri=~"teleware.fi") {
if (method=="REGISTER") {
save("location");
break;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
} else if (uri=~"^tel:") {
forward("1.2.3.4", "5060");
} else if (uri=~"^sip:[0-9]") {
forward("1.2.3.4", "5060");
} else if (uri=~"^sip:+[0-9]") {
forward("1.2.3.4", "5060");
};
The problem is on one of the else if -lines. Are the nested ifs causing
trouble?
Is the config otherwise OK, the scenario is this: For requests to our
domain, forward to (registered) users, for calls to PSTNs, forward to
1.2.3.4 and for calls to other SIP addresses, forward to correct SIP
proxies.
--
TONI HEINONEN
TELEWARE OY
Mob. +358 40 836 1815 / Tel. +358 (9) 3434 9110
Laajalahdentie 23, FIN-00330 Helsinki, Finland
toni(a)teleware.fi / www.teleware.fi
Hello all,
I would like to do some changes to the uri based upon the username.
I have SIP endpoints with numbered usernames expressing thier location
e.g. username 0361234567
I want to make translations to international format but also support local without having to dial 036
International en national are ok, see below.
# it is Internationaly
if (uri=~"^sip:00")
{
strip(2);
};
# It is NL
if (uri=~"^sip:0")
{
strip(1);
prefix("31");
};
Local I am having problems with. I would like to use length but the fucntion does not seem to exist. In addition I run into the problem on how to add "3136" in this case but something else if username is different ?
if (uri=~"^sip:[1-9][1-9][1-9][1-9][1-9][1-9][1-9]@")
{
prefix("some nice trick based on username");
};
any ideas ?
thanks,
Arne.
Hi All,
I am trying to build ser-0.8.14 from source.
I downloaded the file ser-0.8.14_src.tar.gz and after
unzipping, I used following commands to build:
make prefix=/ all
But following errors occured, and no compilation at
all. I am sure, I am missing something in its
compilation. Any help would be much appreciated.
[root@test1 ser-0.8.14]# make prefix=/ all
Makefile.rules:81: action.d: No such file or directory
Makefile.rules:81: crc.d: No such file or directory
Makefile.rules:81: data_lump.d: No such file or
directory
Makefile.rules:81: data_lump_rpl.d: No such file or
directory
Makefile.rules:81: dprint.d: No such file or directory
Makefile.rules:81: dset.d: No such file or directory
Makefile.rules:81: error.d: No such file or directory
Makefile.rules:81: fifo_server.d: No such file or
directory
Makefile.rules:81: flags.d: No such file or directory
Makefile.rules:81: forward.d: No such file or
directory
Makefile.rules:81: hash_func.d: No such file or
directory
Makefile.rules:81: ip_addr.d: No such file or
directory
Makefile.rules:81: main.d: No such file or directory
Makefile.rules:81: md5.d: No such file or directory
Makefile.rules:81: md5utils.d: No such file or
directory
Makefile.rules:81: modparam.d: No such file or
directory
Makefile.rules:81: msg_translator.d: No such file or
directory
Makefile.rules:81: pass_fd.d: No such file or
directory
Makefile.rules:81: proxy.d: No such file or directory
Makefile.rules:81: re.d: No such file or directory
Makefile.rules:81: receive.d: No such file or
directory
Makefile.rules:81: resolve.d: No such file or
directory
Makefile.rules:81: route.d: No such file or directory
Makefile.rules:81: route_struct.d: No such file or
directory
Makefile.rules:81: script_cb.d: No such file or
directory
Makefile.rules:81: sr_module.d: No such file or
directory
Makefile.rules:81: stats.d: No such file or directory
Makefile.rules:81: tcp_main.d: No such file or
directory
Makefile.rules:81: tcp_read.d: No such file or
directory
Makefile.rules:81: timer.d: No such file or directory
Makefile.rules:81: udp_server.d: No such file or
directory
Makefile.rules:81: usr_avp.d: No such file or
directory
Makefile.rules:81: mem/f_malloc.d: No such file or
directory
Makefile.rules:81: mem/mem.d: No such file or
directory
Makefile.rules:81: mem/memtest.d: No such file or
directory
Makefile.rules:81: mem/q_malloc.d: No such file or
directory
Makefile.rules:81: mem/shm_mem.d: No such file or
directory
Makefile.rules:81: mem/vq_malloc.d: No such file or
directory
Regards,
Jawad
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Hello,
Has anyone experiences with the zyxel Sip phone?
I need to input the data of the SIP-server, e.g. of the iptel
SER-server.
Does anyone have the common data of server names, ports etc. in a small
description.
Thanks in advance,
Jaap Geelhoed
Mike Tkachuk wrote:
>
>
>
>>>
>>>
>>>
>
>It will go directly if you don't use any 'rtpproxy with nathelper' or 'mediaproxy'.
>
>
>
Hi Mike,
Are these x-lite or ser settings couldn't find them anywhere...I use
almost the default ser.conf...there is also no difference in quality
between gsm and the other codecs.
Rgds
Dimitri
What codecs are you using? By default xlite will use G711 which is a
64kbps + overheads codec. If you have a congested line this may well be
unsustainable. As far as I know, Skype uses a much lower rate codec by
default. Try switching to something like GSM in xlite and rerun the
tests.
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Dimitri Georganas
Sent: 19 August 2004 06:38
To: serusers(a)lists.iptel.org
Subject: [Serusers] newbie question
I did a test last week:
setup: 2 computers on separate locations, on both installed: Skype and
XTen's free sip phone: x-lite, 1 SERr server on a third location
Linksys WRT54g router on one location, Cisco SOHO78 on the others. No
QoS.
test: voice connection with skype between the two computers: perfect
quality
test2 voice connection with x-lite between the two computers: After
authorizing both computers on the third server,
the established connection showed terrible quality...long silence during
phonecalls.
First I thought that the actual x-klite/sip voice traffic is routed
through the server and since the server was heavily loaded and the
router didn't have QoS, the bad quality was explainable. Later I heard
that the actual voice traffic is not routed through the ser server, but
goes directly from one computer through the other.
Now if the latter is true, how can one explain the quality difference
between skype and x-lite/sip ?
Thanks in advance for your answers.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
I did a test last week:
setup: 2 computers on separate locations, on both installed: Skype and
XTen's free sip phone: x-lite, 1 SERr server on a third location
Linksys WRT54g router on one location, Cisco SOHO78 on the others. No QoS.
test: voice connection with skype between the two computers: perfect
quality
test2 voice connection with x-lite between the two computers: After
authorizing both computers on the third server,
the established connection showed terrible quality...long silence during
phonecalls.
First I thought that the actual x-klite/sip voice traffic is routed
through the server and since the server was heavily loaded and the
router didn't have QoS, the bad quality was explainable. Later I heard
that the actual voice traffic is not routed through the ser server, but
goes directly from one computer through the other.
Now if the latter is true, how can one explain the quality difference
between skype and x-lite/sip ?
Thanks in advance for your answers.
Dear All,
I am getting trouble at the time of recording the voice mail for calle when i am using PHS network for my sip client. My sip cleint support GSM 06.10 and it is working perfetally on LAN, but when I am using PHS card to make a SIP call and if calle is unavialable them my call is route to SEMS and SEMS send a greeting message to me on the behalf of callee. I am able to listen this greeting message in PHS network but as soon as greeting message finish I suppose to record my message for calle but instead I get the bye packet from SER proxy and my call terminated without any recording, it seems that some problem with GSM file format. but I could not identified what is the actual problem...Can any body suggest me someting how to solve this problem...
Thanks in advance
Koyama
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