Hello all,
Im using ser, with esteras softphone, and a couple of cisco ata186's. I
have to say I really like ser, as it is small fast, and reliable without
being over complicated.
I would like to know what other ser users and developers use for pstn
gateways? I have a openline4 card from voictronix, but there is no
software support for sip (When I bought it I thought I would use h.323).
I have looked at cisco's pstn gateway modules, but was turned off them
because of price, and lack of clear straight forward information to name
but two reasons.
Ideally what I would like to have is a solution that would support at
least 4 pstn lines (expandable to maybe 8), that would run on FreeBSD
(or at worst linux), works well with sip & ser, with high quality audio,
and minimal delays when dialing in out.
TIA,
-Emil
hello.
i've been trying to register to ser with couple of UAs like x10, estara and snom100.
the response is following:
RECEIVE << ser_ip_address:5060
SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP my_ip_address:5060
From: <sip:krav@sip.sipuser.com>
To: <sip:krav@sip.sipuser.com>;tag=53d3b67acb7f1aa5eed7d0549c34647b.dea0
Call-ID: 77FCCA606A8111D7B7A2000021F0A5BA(a)sip.sipuser.com
CSeq: 1788 REGISTER
Server: Sip EXpress router (0.8.10 (i386/freebsd))
Content-Length: 0
Warning: 392 212.58.142.4:5060 "Noisy feedback tells: pid=93155 req_src_ip=ser_ip_address in_uri=sip:sip.sipuser.comout_uri=sip:sip.sipuser.com via_cnt==12"
mysql has been installed properly, and i can add subscribers using serctl......
i guess there's something wrong with script, which is default, i've just uncommented everything regarding auth and mysql.
and following is what i get in ser box logs:
May 12 09:08:57 sip ./ser[93190]: ERROR: parse_cseq: bad cseq
May 12 09:08:57 sip ./ser[93190]: ERROR: get_hdr_field: bad cseq
May 12 09:08:57 sip ./ser[93190]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed
any suggestions please?
thanks.
Hi All,
I need to implement an used based dialplans module on SER, so that depending
on the user information stored in the database, a certain prefix should be
inserted in the called number. Has anyone had a similar issue?
Thanks
Fernando
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Howdy,
I am trying to determine why a call won't connect when dialed using an
alias. I'm running SIP Express Router v0.8.10 on FreeBSD 4.8-RELEASE.
Below is the message the client (Xten X-Lite) is receiving and
apparently discarding.
It appears as though the call is being routed but I'm puzzled by these
message headers.
RECEIVE << 192.168.1.11:5060
INVITE sip:pyuska@192.168.1.25:5060 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKad62.cf628db7.0
Via: SIP/2.0/UDP 192.168.1.15:5060
From: yuska <sip:pyuska-ppc@shitworks.com>;tag=585911808
To: <sip:5315@shitworks.com>
Contact: <sip:pyuska-ppc@192.168.1.15:5060>
Call-ID: 06E1E114-22A1-8B9D-2891-72E17E9928EA(a)192.168.1.15
CSeq: 12625 INVITE
Content-Type: application/sdp
Content-Length: 283
Do I need to configure the SER server re-write the To: address or is the
client misbehaving by not responding to the Contact: ?
Any light you could shed on this would be appreciated.
Phil
At 12:14 PM 4/30/2003, radan(a)nasty.gda.pl wrote:
>Hello all !
>I'm new user of ser :)
>
>I'd like have a information about registered users in my syslog
>
> save("location");
> log(3,"REGISTER zarejestrowany uzytkownik $USER");
> sl_send_reply("200", "ok"); ^^^^^^
> break;
> };
>
>
>is some variable to use ?
currently not. We are planning it for the future.
-Jiri
Hello all !
I'm new user of ser :)
I'd like have a information about registered users in my syslog
save("location");
log(3,"REGISTER zarejestrowany uzytkownik $USER");
sl_send_reply("200", "ok"); ^^^^^^
break;
};
is some variable to use ?
Andrzej Radke
sip:radan@task.gda.pl
Hi,
I am currently trying to extend ser's parser to allow it recognize
User-Agent headers, but stuck into the magic HASH_TABLE_SIZE value.
Is there any algorithm to calculate it when adding support for a
new header type? Maybe there is some documentation, apart from
comments in the code itself, on extending parser, or at least on
its inner details? If so, it would be nice to obtain it if possible.
Thanks!
-Maxim
Folks,
Attached please find a patch, which extends usrloc/registrar modules
to save values from User-Agent field in REGISTER messages into the
database. It would be nice to have it included into the next release.
Please disregard hackish detection of User-Agent fields, I am currently
working on extending parser to recognize that type of field. I'll
post updated patch when finished.
Also, there is a new unrelated function which can be used to
retrieve full list of all currently registered users in all domains.
I am using this function in new version of my nathelper modules to
do periodical UDP "pinging" of all registered users to keep their NAT
bindings alive. I would like to have it reviewed and included into
the next release as well.
Thanks!
-Maxim
At 03:33 PM 4/22/2003, Alejandro Olchik wrote:
>I need some help to implement load balance
>when sending invites to an external sip
>proxy server.
>
>I would like to be able to balance load
>between to IPs (10.0.0.1 and 10.0.0.2) and
>use the backup IP when the primary one
>fails.
>
>Below is the routing code I have:
>
>
> if (!lookup("location")) {
> rewritehost("10.0.0.1");
> if (!t_relay_to("10.0.0.1","5060")) {
> sl_send_reply("404", "Not Found");
> };
> break;
> };
>
>How can I add this behaviour?
That depends on the distribution scheme you would like to use.
For example, you can implement a distribution function that
splits requests based on some downstream weights.
e.g.,
modparam("distrib", "weights",
# ...
# rewrite host-part with one of values using the distribution
# 20-30-50
distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%)
t_on_negative("1")
t_relay_to();
# ...
reply_route[1] {
# look at destinations that failed, and update their probabilities
# according to some strategy; for example, temporary 0% weight --
# note that this would take shared memory for keeping the weights
# and a timer
update_weights();
# if coupled to TM, it can look at previous attempts to eliminate
# retrying to a previously failed destination
distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%)
}
-Jiri
On your pots dial-peer set the codec to g711alaw, or configure your
SIP clients to use that codec.
Dan
-----Original Message-----
From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de]
Sent: Friday, April 25, 2003 7:55 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] problem with cisco 2600 to pstn
Hi all,
I am expericing problem with the cisco 2600, which should function as
the sip2pstn gateway. If I try to complete a call from a sip phone to
pstn, the router says:
------------------------------------------------------------------------
00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03
00:15:49: Bearer Capability i = 0x8090A2
^^^^^^
00:15:49: Channel ID i = 0x83
00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN,
may
have in-band info
00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown,
Type:Unknown
00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private,
Type:Subscriber(local)
00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83
00:15:49: Channel ID i = 0x89
00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate
now
available
00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83
00:15:49: Cause i = 0x80C1 - Bearer capability not implemented
^^^^^^^^^^^^^^^^^^^^^^^^^^
00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate
now
available
------------------------------------------------------------------------
----
------------
Please notice the line "Bearer Capability i = 0x8090A2", the digits
"80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps
and "A2" is for G.711 u-law.
So if I call the router from a normal telephone, the debugging looks as
follows:
------------------------------------------------------------------------
----
----
01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01
01:01:58: Bearer Capability i = 0x8090A3
^^^
01:01:58: Channel ID i = 0x89
01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN,
Type:Unknown
01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN,
Type:Subscriber(local)
01:01:58: High Layer Compat i = 0x9181
01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at
64 Kb/s
------------------------------------------------------------------------
----
---------
whereat the bearer capability is "0x8090A3". It means that the
ISDN-switch of German Telekom uses G.711 a-law.
I am afraid that is the reason why the sip-call doesn't go through. But
I can't find any way to configure this.
Has anybody in this mailinglist the same experience?
Any hints would be very appreciated.
Thanks
yang
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