Does anybody knows how I can configure Asterisk to work properly My* box is
sending wrong messages and instead of sending the release message for
invalid # it sends an ALERTING message saying that the call is going on.
Thanks
hello ,
thanks for your responses , iam desperatley searching
for the free billing software for the reliable cvs
ser-0.8.12 and freeradius-0.9.3 .
i tried trabas but it s supporting only h323
attributies not sip.
and if i go for ispbs i doubt it does not support
radius and rtpproxy.
i am expecting some leads from you.
so kindly suggest me good free voip billing system for
ser.
with regards
rama kanth
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Hi all,
I got a DB error when I tried to login into serweb(both admin and user_interface).
The message showing on IE is "DB Error: no such table, file: /var/www/novsky.com/serweb/html/data_layer.php:405".
I checked mysql's log. It query DB ser, is:
select priv_name, priv_value from admin_privileges where username = 'admin' and domain = 'iptel.org' and (priv_name = 'change_privileges' or priv_name = 'is_admin');
That table "admin_privileges" is not exist.
In DB ser, there are these tables:
+-----------------------+
| Tables_in_ser |
+-----------------------+
| acc |
| active_sessions |
| aliases |
| config |
| domain |
| event |
| grp |
| location |
| missed_calls |
| pending |
| phonebook |
| preferences |
| reserved |
| server_monitoring |
| server_monitoring_agg |
| silo |
| subscriber |
| uri |
| version |
+-----------------------+
And these tables is created by ser_mysql.sh.
How can I do?
Best regards,
Jimmy
Are there any config settings for controlling echo in SER ?
I have a Grandstream HT-486 connecting to SER then to Asterisk. There is a
bad echo when calling from the SIP side, asterisk has several methods of
canceling echo, but in this case have no effect.
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We are looking for some help in setting up a voice over IP system similar to Vonage. If you are experienced with SER and interested, contact me directly at:
778-893-9348
rhulays(a)netfone.ca
Best of luck.
Anil,
Still i dont think you need to t_relay calls through your code as there are a number of options
available with SER which you can use to (re)direct the calls to the desired gateways. See the
sections for handling URIs, and setting hosts in the SER Admin's Guide:
http://www.iptel.org/ser/doc/seruser/seruser.html
Best Regards,
--- Anil Bhati <abhati_voip(a)hotmail.com> wrote:
>
> Hi All
>
> actually i m facing t_relay if i wants to relay a call in more than 3 pstn
> gw that time i have to check which prefix is coming and where is call going
<SNIP>
> Abhati
>
=====
Girish Gopinath <gr_sh2003(a)yahoo.com>
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I have searched the archive for an answer to this question, and I did not find it. If I missed, I apologize. Please point me in the right direction.
I am having a problem with the pa module of ser (0.8.12). Using ethereal, I see that in response to a SUBSCRIBE, ser will sometimes send an OK and then a NOTIFY. However, other times ser sends the NOTIFY and then the OK. For a single UA sending multiple SUBSCRIBE's to ser, about half of the OK's are sent first and about half of the NOTIFY's are sent first.
This is a problem because I am running the UA behind a SIP-aware firewall which expects the OK first. If the OK is not received, the NOTIFY will not be allowed back through the firewall.
Looking at RFC 3265, 3.1.4.1:
"This SUBSCRIBE request will be confirmed with a final response. 200-class responses indicate that the subscription has been accepted, and that a NOTIFY will be sent immediately."
Looking at RFC 3265, 3.1.6.2:
"Upon successfully accepting or refreshing a subscription, notifiers MUST send a NOTIFY message immediately to communicate the current resource state to the subscriber. This NOTIFY message is sent on the same dialog as created by the SUBSCRIBE response.
<snip>
Note that a NOTIFY message is always sent immediately after any 200-class response to a SUBSCRIBE request, regardless of whether the subscription has already been authorized."
My interpretation is that the RFC requires the OK to be sent first. Is that a valid interpretation?
Is there an easy fix for this (perhaps in ser.cfg) or will this require a code change?
Thanks,
Jay
As far as we dig into it, none of the following:
freeradius
radiusclient
radius-cistron
radius-livingston
> Implement 15 Failed
Following RFCs should not lead us falling over the cliff. Maybe 15
should be traded with something else understood by the radius servers
found in the real-world?
Adrian
Begin forwarded message:
> From: Juha Heinanen <jh(a)tutpro.com>
> Date: June 16, 2004 8:26:49 PM CEST
> To: Adrian Georgescu <ag(a)ag-projects.com>
> Cc: iancu(a)fokus.fraunhofer.de, serusers(a)lists.iptel.org
> Subject: 15 is a not a recognized RADIUS Packet
>
> Adrian Georgescu writes:
>
>> Do you know by any chance what does it mean?
>
> here are all official radius acct-status-types:
>
> Values for RADIUS Attribute 40, Acct-Status-Type [RFC 2866]:
>
> 1 Start [RFC 2866]
> 2 Stop [RFC 2866]
> 3 Interim-Update [RFC 2866]
> 4-6 (unassigned)
> 7 Accounting-On [RFC 2866]
> 8 Accounting-Off [RFC 2866]
> 9 Tunnel-Start [RFC 2867]
> 10 Tunnel-Stop [RFC 2867]
> 11 Tunnel-Reject [RFC 2867]
> 12 Tunnel-Link-Start [RFC 2867]
> 13 Tunnel-Link-Stop [RFC 2867]
> 14 Tunnel-Link-Reject [RFC 2867]
> 15 Failed [RFC 2866]
>
> you can find definition of 15 from the above mentioned rfc, but if a
> fails before starting, then 15 is the right way to report it.
>
> -- juha
HI ALL;
vonage system offers voicemail service in 2 forms:
1- email with a .wav file attachment
2-the user can access his voicemail box by phone.
As I know the first one is exutable by ser+sems.
My question is, with ser+sems can we dial a number and then recive our voicemail via phone.
thank you in advance
mohammad