I am using SER 0.9.3, MediaProxy 1.4.2 and sometimes the call quality gets pretty bad. Parts of the conversation drops and sometimes there will be a delay of 10 or 15 seconds. The SER server has a public IP and is not behind a firewall. The UAs are spread out at different locations behind different firewalls. Sometimes it works perfect though, with a 45min conversation having no problems whatsoever. Does anyone have any suggestions on what I might be able to do to track down/fix this?
-John
Hi!
I am new to SER, and have following question. How can I provide that:
1) Users can register only their sip addresses;
2) Users can use their username in sip From: field.
In my installation sip username and authentication usernames differ a
little. For example, sip username could be 112233, but auth username
is sip112233. So, I want to be sure that user that can successfully
autheticate as sip112233 will not use 345678 username, and always will
be using 112233 in From? I need something like check_from, but this
function requires identical sip and auth usernames.
Is this thing doable? If so, I will be glad to hear how I should do
this in openser.cfg or any hints/links that could point me in right
direction.
Thanks!
--
Timur Irmatov, JID: thor(a)jabber.org.au.
Hi!
Has anyone of you tried to use the database to make routing? I need to get
information of the database and then rewritehost with the result...
Thanks!
Sebastian
Hello,
Yes, the INVITE goes to PORT 5060.
Here is my ngrep:
U 2005/08/02 15:43:56.584366 xxx.xxx.xxx.xxx:51403 -> xxx.xxx.xxx.xxx:5060
INVITE sip:10099999999999@xxx.xxx.xxx.xxx:5060 SIP/2.0.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;x-route-tag="tgrp:OutTrunkGroup".
From: "anonymous" <sip:54@xxx.xxx.xxx.xxx>;tag=CEA94420-184A.
To: <sip:10099999999999@xxx.xxx.xxx.xxx>.
Date: Tue, 02 Aug 2005 18:43:56 GMT.
Call-ID: 375AFBE2-2BC11DA-B615D293-1A0569D1(a)xxx.xxx.xxx.xxx.
Supported: timer,100rel.
Min-SE: 1800.
Cisco-Guid: 927669769-45879770-3054686867-436562385.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO.
CSeq: 101 INVITE.
Max-Forwards: 6.
Remote-Party-ID:
<sip:54@xxx.xxx.xxx.xxx>;party=calling;screen=yes;privacy=full.
Timestamp: 1123008236.
Contact: <sip:54@xxx.xxx.xxx.xxx:5060>.
Expires: 180.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Length: 382.
Thanks!
Sebastian
----- Original Message -----
From: "Miguel Angel Villar" <mavilla(a)redvoiss.net>
To: "'Sebastian Kühner'" <skuehner(a)veraza.com>
Sent: Tuesday, August 02, 2005 7:38 PM
Subject: RE: [Serusers] UDP
Did you chaeck that the INVITE messages from the Cisco server are send to
UDP port 5060?.
MaV.-
> -----Mensaje original-----
> De: Sebastian Kühner [SMTP:skuehner@veraza.com]
> Enviado el: Martes, 02 de Agosto de 2005 04:11 p.m.
> Para: serusers(a)lists.iptel.org
> Asunto: [Serusers] UDP
>
> Hi!
>
> I'm getting INVITES from another Cisco server, but my ser isn't catching
> it.
> The difference between a "normal message" (from one of our phones) is,
> that
> a dump shows us "UDP" like protocol (Cisco)... and SIP from our phones.
>
> Do I have to respect different SIP signallings?? I made a log on
> "INVITES"... but ser isn't catching it :-(
>
> Thanks!
>
> Sebastian
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Hi everybody,
Looking to implement Codec Translator with Asterisk - loading codec modules API in Asterisk to support transcoding.
Has anyone experienced this issue, and does anyone have any suggestions or hint, simple scripts, whatever ?
Thanks in advance and kind Regards,
Frank
---------------------------------
Yahoo! Mail for Mobile
Take Yahoo! Mail with you! Check email on your mobile phone.
Hi Bogdan,
Thanx for the information about using allow_trusted. It works fine. I just wanted to clarify one thing, is using SQL more efficient or Radius is better. I wanted to know performance wise. What would be the overhead in both cases.
one more thing, whenever we add a new IP address to the trusted table openser needs to be restarted. Is there any alternative to this. Which type of calls would be affected when we restart openser.
please let me know some information about this.
Thanx a lot.
Jayesh
Thanks to Bogdan, who pointed out that my UAC was not behaving properly.
Having solved that problem in sipp, I now have a new issue. The UAC
is opening a TCP connection to the proxy, but the proxy then forwards
the INVITE to the UAS over UDP. I'm using the stock openser.cfg file.
Any suggestions?
Thanks,
-Erich
--
Erich M. Nahum IBM T.J. Watson Research Center
Research Staff Member P.O. Box 704
nahum(a)watson.ibm.com Yorktown Heights NY 10598
Hi all,
I'm playing around serial forking with avpops module. My goal is to have one PSTN number mapped to N voip number; so, when the first voip number is busy, OPENSER catch 486 reply and then try with the second AOR and so on. But here raise the problem: the OPENSER forward the reply upstream with a code 407 Proxy Auth Required and the PSTN GW send his ACK and all stop here.
Below ther's a snippet of mi openser.cfg:
modparam("avpops", "avp_aliases", "serial_fork=i:665")
[..skipping..]
#if RURI is to a voip user:
avp_db_load("$ruri", "$serial_fork");
avp_print();
avp_pushto("$ruri","$serial_fork");
lookup("aliases");
if lookup("location") {
t_on_reply("1");
t_on_failure("2");
t_relay();
break;
};
onreply_route[1] {
if (t_check_status("486")) {
t_on_failure("2");
break;
};
}
failure_route[2] {
if (t_check_status("486")) {
# delete the first element of the list (if any) and pass to second from list
avp_delete("$serial_fork");
if (avp_pushto("$ruri", "$serial_fork")) {
append_branch();
avp_delete("$serial_fork");
t_on_failure("2");
t_relay();
}
}
}
My DB entries are:
+------+------------+------------------------+-----------+------+---------------------------------------+---------------------+
| uuid | username | domain | attribute | type | value | modified |
+------+------------+------------------------+-----------+------+---------------------------------------+---------------------+
| | 0662293703 | sipexp.mydomain.org | 665 | 2 | sip:0662293701@sipexp.mydomain.org | 2005-08-02 15:03:27 |
| | 0662293703 | sipexp.mydomain.org | 665 | 2 | sip:0662293702@sipexp.mydomain.org | 2005-08-02 15:03:33 |
+------+------------+------------------------+-----------+------+---------------------------------------+---------------------+
Do you have a suggestion to resolve this problem? Why OpenSer send upstream 407 after received 486 Busy Here and not try to send a new INVITE to voip user with new R-URI?
Many thanx,
Verbal
Jim,
Your prepaid system concept is a little different from mine. For mine,
the SER is just a middleman. I route all my calls to Cisco CallManager
which will then route accordingly. So I just get all my accounting info
from the Callmanager CDR. I then run ASP scripts to check the users
balances and if it ever finds a low-in-credit user, it will go into the
mysql db by odbc and insert the user into a grp (acl is preconfigured to
block this grp from accessing the PSTN).
For your case, probably the only thing u can do is to match 2 things,
the dest. Country code and the different local charges for each country.
Do comment about my set up.
Regards,
Sam
-----Original Message-----
From: Sanjim Sanjim [mailto:sanjim007@gmail.com]
Sent: Wednesday, August 03, 2005 5:33 PM
To: Sam Lee
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] is SER Dependable for prepaid System ? Anyone
using itfor prepaid system?
Sure Sam, I am currently developing a script which will check the credit
balance of the user from mysql and accordingly rewrite the uri.
If the balance is low... it will forward to SEMS announcement.
Like that when SER receives a BYE, the Script should check the
destination and total mins and calculate the amount and reduce it from
the user credits.
Currently the difficult part is the destination country code. I have
stored all the codes in the mysql prefix table. But each country have
sub codes for states. For example UK proper is only .02 cents, but
mobile is .40 cents We have to differentiate the code properly.
If you have any other suggestions kindly let me know.
Thanks,
JiM
On 8/3/05, Sam Lee <sam.lee(a)super.net.sg> wrote:
> I have been working for quite some time on deploying SER as a prepaid
> system. Currently it is not done yet as there's still quite a number
> of things to be done.
> Probably we can share a few ideas down the road.
>
> Regards,
> Sam
>
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
> On Behalf Of Sanjim Sanjim
> Sent: Tuesday, August 02, 2005 7:02 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] is SER Dependable for prepaid System ? Anyone
> using itfor prepaid system?
>
> is SER Dependable for prepaid System ? Anyone using it for prepaid
> system?
>
> I very much like the SER performance in the test environment. Only
> drawback(maybe advantage if we provide unlimited plan) ) is stateless
> operation.
>
> But I want to know anyone using it in realtime prepaid business?
>
>
>
> Thanks,
> JiM
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
