Hello,
I made a ngrep and I noticed that NO "487 Request Cancelled" is reaching my
ser on CANCEL. That could be the problem.
I also tried to use the avpops module:
if (method == "CANCEL") {
setflag(1);
avp_write("cancel", "s:failover");
log(1, "-CANCEL PSTN-\n");
};
...and then with avp_check in the failover route. Without result. Could it
be that the variables in the avp don't reach another route?
Thanks for your help
Sebastian
----- Original Message -----
From: "Bayan William Towfiq" <william(a)telepacket.com>
To: "Sebastian Kühner" <skuehner(a)veraza.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Thursday, August 18, 2005 3:39 PM
Subject: Re: [Serusers] cancal
> Hi Sebastian,
> Sorry about that, I misread the code. I will test your code and dig up
> an example of how I do it in my failure routes to show you.
>
> William
>
> Sebastian Kühner wrote:
>
> >Hi William,
> >
> >Thanks for your help.
> >
> >In my failover route I have the following code:
> >
> >failure_route[2] {
> > if (t_check_status("408|500|503"))
> > {
> > log (1, "next gateway...\n");
> > if (!next_gw())
> > {
> > t_reply("503", "Service not available, no more gateways");
> > break;
> > }
> > log (1, "gateway changed...\n");
> > t_on_failure("2");
> > t_relay();
> > }
> >}
> >
> >So the status is already filtered. I tried your code but without
result...
> >
> >Does anybody know how I can stop the failover timer?
> >
> >Thanks!
> >
> >Sebastian
> >
> >
> >
> >----- Original Message -----
> >From: "Bayan William Towfiq" <william(a)telepacket.com>
> >To: "Sebastian Kühner" <skuehner(a)veraza.com>
> >Cc: <serusers(a)lists.iptel.org>
> >Sent: Thursday, August 18, 2005 1:28 AM
> >Subject: Re: [Serusers] cancal
> >
> >
> >
> >
> >>Hi Sebastian,
> >>If the message is a cancel you it will have status 487
> >>
> >>so you can just add
> >>
> >> if (t_check_status("487")) {
> >> break;
> >> };
> >>
> >>before the rest of the code in the failure route. You can mail me
> >>privately if you have any more questions about this issue.
> >>
> >>William
> >>
> >>Sebastian Kühner wrote:
> >>
> >>
> >>
> >>>Hello,
> >>>
> >>>I'm using ser to forward to PSTN Gatways.
> >>>
> >>>Before the t_relay I put the command:
> >>>
> >>>t_on_failure("2");
> >>>
> >>>So, after a timeout ser goes to:
> >>>
> >>>failure_route[2] {
> >>> log (1, "next gateway...\n");
> >>> if (t_check_status("408|500|503"))
> >>> {
> >>> if (!next_gw())
> >>> {
> >>> t_reply("503", "Service not available, no more gateways");
> >>> break;
> >>> }
> >>> log (1, "gateway changed...\n");
> >>> }
> >>> t_on_failure("2");
> >>> t_relay();
> >>>}
> >>>
> >>>This is working very good... if the user doesn't make a hangup.
> >>>
> >>>If caller hangs up, the CANCEL hits SER and the call is cleanly
> >>>
> >>>
> >terminated.
> >
> >
> >>>SER however continues to failure_route after timeout of the initial
> >>>INVITE... ser sends out a new INVITE message to another PSTN-Server...
> >>>
> >>>
> >and
> >
> >
> >>>the phone rings (with nobody in the line)
> >>>
> >>>Many thanks for your help!
> >>>
> >>>Sebastian
> >>>
> >>>
> >>>
> >>>_______________________________________________
> >>>Serusers mailing list
> >>>serusers(a)lists.iptel.org
> >>>http://lists.iptel.org/mailman/listinfo/serusers
> >>>
> >>>
> >>>
> >>>
> >>
> >>
> >>
> >
> >
> >
> >
>
>
>
Hi all,
I have ser-0.8.14 installed on my Linux with MySQL support.
I want to upgrade it to ser-0.9.3. What do you advise me to do? Use bin
or src?
Thanks,
Marc
hello
i m using ser-0.9.0 on 5060 and asterisk-1.0.9 on 5970
but problem is that i cannot send my ser back to
asterisk.
the problem is that it always goto route(4) second
else if is not properly checked how to check this
condition. i m adding 0 when asterisk is sending call
back.
if(uri=="^sip:00[1-9]+@.*") {
route(4);
break;
} else if (uri=="^sip:0[1-9]+@.*") {
strip(1);
route(5);
break;
} else {
route(4);
break;
}
__________________________________
Do you Yahoo!?
Read only the mail you want - Yahoo! Mail SpamGuard.
http://promotions.yahoo.com/new_mail
Hi
I have alittle problem, I am looking at doing call transfers from
inbound PSTN to IP phones, who can then transfer across their
department, BUT my pstn gateway is currently not supporting REFER. This
is what seems to be causing the problem (tks Steve), now if this is the
case, can I
a) get transfer to work without this
b) use asteriisk somehow to take the call inbound into asterisk , and
then do transfer
Iqbal
I need to load in memory a large table (currently about 10.000 entry).
During the sql query (MySQL) the following error happens and SER quit!
The error is due to the failure of a pkg_malloc().
I tried to increase the PKG_MEM_POOL_SIZE and SHM_MEM_SIZE but nothing
changed.
What I have to do to succed the large query?
SER is 0.9.3 under OpenBSD 3.7 amd64.
Thanks.
Aug 19 17:56:09 eowyn ./ser[27075]: convert_row(): No memory left
Aug 19 17:56:09 eowyn ./ser[27075]: convert_rows(): Error while
converting row #7014
Aug 19 17:56:09 eowyn ./ser[27075]: convert_result(): Error while
converting rows
Aug 19 17:56:09 eowyn ./ser[27075]: store_result(): Error while
converting result
--
___________________________________________________
__
|- giannici(a)neomedia.it
|ederico Giannici http://www.neomedia.it
___________________________________________________
Hi!
Here is a ngrep of my cancel message of a call that wasn't established
(cancel on ring, not while talking):
U 2005/08/19 09:45:46.747447 xxx.xxx.xxx.xxx:1024 -> xxx.xxx.xxx.xxx:5060
CANCEL sip:0054261xxxxxxxx@pbx2.test.com:5060 SIP/2.0.
Content-Length: 0.
Call-ID: 2E863532-7483-4585-B9FA-C6EC3340203B(a)192.168.1.101.
Max-Forwards: 70.
From: "Administrator"<sip:44441@pbx2.test.com:5060>;tag=257139028539.
CSeq: 1 CANCEL.
To: <sip:0054261xxxxxxx@pbx2.test.com:5060>.
Via: SIP/2.0/UDP
xxx.xxx.xxx.xxx:1024;rport;branch=z9hG4bKc0a801650131c9b14305d51d00002639000
000b4.
User-Agent: SJLabs-SJphone/1.30.252.
.
#
U 2005/08/19 09:45:46.997466 xxx.xxx.xxx.xxx:5060 -> xxx.xxx.xxx.xxx:1024
SIP/2.0 200 ok -- no more pending branches.
Call-ID: 2E863532-7483-4585-B9FA-C6EC3340203B(a)192.168.1.101.
From: "Administrator"<sip:44441@pbx2.test.com:5060>;tag=257139028539.
CSeq: 1 CANCEL.
To:
<sip:0054261xxxxxxx@pbx2.test.com:5060>;tag=2fb8a6135db5d855a493c61ec9633675
-1e47.
Via: SIP/2.0/UDP
xxx.xxx.xxx.xxx:1024;rport=1024;branch=z9hG4bKc0a801650131c9b14305d51d000026
39000000b4.
Server: Sip EXpress router (0.9.3 (i386/linux)).
Content-Length: 0.
Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=31972
req_src_ip=xxx.xxx.xxx.xxx req_src_port=1024 in_uri=sip:00542
.
Shouldn't that be 486 Request Terminated instead of 200 ok?
Thanks!
and I thought I was privileged :-)
Iqbal
Juan wrote:
>No. You are not the only to receive them. As I don't think I am the only one
>receiving the mails from Lazy.charlie.
>I would like to see both blocked.
>Regards
>
>Juan
>
>-----Original Message-----
>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
>Behalf Of Iqbal
>Sent: Friday, August 19, 2005 11:22 AM
>To: fedora(a)voice.dyndns.tv
>Cc: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] User-Agent: Asterisk PBX.
>
>am I the only one who gets these messages, if so, although I dont mind being
>part of the privileged few...I would prefer t if I didnt get these :-), if
>everyone is getting them, mailing list admin, can we delete/remove/ban them
>
>Iqbal
>
>fedora(a)voice.dyndns.tv wrote:
>
>
>
>>#
>>U 2005/08/20 00:13:30.111494 221.186.150.67:5060 -> 221.186.150.68:5060
>>BYE sip:8000@221.186.150.68:5060 SIP/2.0.
>>Max-Forwards: 10.
>>Record-Route: <sip:221.186.150.67;ftag=as1cbee789;lr=on>.
>>Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0.
>>Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062.
>>From: <sip:500@voice.dyndns.tv>;tag=as1cbee789.
>>To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543.
>>Contact: <sip:500@221.186.150.67:5062>.
>>Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68.
>>CSeq: 102 BYE.
>>User-Agent: Asterisk PBX.
>>Content-Length: 0.
>>Route: <sip:8000@221.186.150.68:5060>.
>>.
>>
>>#
>>U 2005/08/20 00:13:30.147446 221.186.150.68:5060 -> 221.186.150.67:5060
>>SIP/2.0 200 Ok.
>>Via: SIP/2.0/UDP 221.186.150.67;branch=z9hG4bK1e0d.6051d181.0.
>>Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062.
>>From: <sip:500@voice.dyndns.tv>;tag=as1cbee789.
>>To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543.
>>Contact: <sip:8000@221.186.150.68:5060>.
>>Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68.
>>CSeq: 102 BYE.
>>Server: X-Lite release 1103m.
>>Content-Length: 0.
>>.
>>
>>#
>>U 2005/08/20 00:13:30.149017 221.186.150.67:5060 -> 221.186.150.67:5062
>>SIP/2.0 200 Ok.
>>Via: SIP/2.0/UDP 221.186.150.67:5062;branch=z9hG4bK2482746e;rport=5062.
>>From: <sip:500@voice.dyndns.tv>;tag=as1cbee789.
>>To: 8000 <sip:8000@voice.dyndns.tv>;tag=4080605543.
>>Contact: <sip:8000@221.186.150.68:5060>.
>>Call-ID: D8497E35-CB5C-4537-9650-A25AA13CED13(a)221.186.150.68.
>>CSeq: 102 BYE.
>>Server: X-Lite release 1103m.
>>Content-Length: 0.
>>
>>_______________________________________________
>>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
>
>
>.
>
>
>
Hi Ashutosh,
Thanks for your response.
The readme file of mediaproxy talks about load balancing the RTP streams across multiple proxies.
Does the signaling also follow the same path as the media (for the same call).
Also, The readme file of the mediaproxy says that:
"Once it has determined which domain to use to obtain the list of available
proxy servers, it will make a DNS lookup for the following SRV records:
_mediaproxy._tcp.selected.domain.name"
Does this mean that the DNS lookup serves as the routing mechanism for the call?
Also, the document says:
"However it should be fairly easy to implement another dispatcher that uses a
different mean to determine the proxy server (like using a database mapping),
or even to distribute the traffic by other means than SIP domain name."
I guess that this means that I could use a 3rd party routing engine to route the calls.
Is that correct? If so, is there an interface in the dispatcher that I could use to make the
dispatcher talk to my routing DB and not do a DNS lookup.
Thanks,
Vikrant
Dear List,
At below config. i setted up OpenSER on Public IP and PSTN Gateway is again
on another Public IP. I use Zyxel Prestige Series 660-HW ADSL modems on both
client side and note that they have SIP ALG, also i've completed port
forwarding in each modem, so SIP2SIP call happens (voice traffic goes from
UA to UA) without need of anything extra as STUN or RTP RELAY. But since
those are very away from each other if a UA needs to call PSTN the voice
traffic flows from UA to PSTN and that lowers QoS :(
I would like to activate MediaProxy again but just for one case which is
PSTN, so whenever a UA needs to call PSTN the voice traffic flows like :
UA (ATA) >>> OpenSER + MediaProxy >>> PSTN
How can i setup this like above without touching SIP2SIP calls between
UA's...
or adding the lines beginning with # these to my existing config is enough ?
Thanks,
Ozan Blotter
debug=3
fork=yes
log_stderror=no
check_via=no
listen=212.XXX.104.XXX # This is OpenSER's Public IP
port=5060
children=4
dns=no
rev_dns=no
fifo="/tmp/openser_fifo"
# fifo_db_url="mysql://openser:openserrw@localhost/openser"
loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
loadmodule "/usr/local/lib/openser/modules/registrar.so"
# loadmodule "/usr/local/lib/openser/modules/mediaproxy.so"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
# modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock")
#
modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients")
#
modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients")
route {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message Overflow");
break;
};
if (method!="REGISTER") {
record_route();
};
if (loose_route()) {
route(1);
break;
};
if (uri!=myself) {
route(1);
break;
};
if (uri==myself) {
if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(1);
break;
};
if (uri=~"^sip:0[0-9]*@*") {
rewritehost("195.XXX.122.XXX"); # This is PSTN Gateways's
Public IP
# use_media_proxy();
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(1);
};
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
if (!save("location")) {
sl_reply_error();
};
}
SER does not handle the media, so I don't think this possible. Codec
negotiation is handled by the gateways.
- Daryl
On 8/19/05, Daryl Sanders <daryl.sanders(a)gmail.com> wrote:
> SER does not handle the media, so I don't think this possible. Codec
> negotiation is handled by the gateways.
>
> - Daryl
>
>
> On 8/18/05, Frank Kostin <frankostin(a)yahoo.com> wrote:
> > Hi everybody,
> >
> > Is it possible routing by Codec type with SER and how would you do that ?
> >
> > Thanks in advance,
> > Frank
> >
> > ________________________________
> > Do you Yahoo!?
> > Read only the mail you want - Yahoo! Mail SpamGuard.
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> >
>