Greetings:
Our only (currently available) method to browse the sgml docs
is to use Panorama for Win16 (free version) or Novell Dynatext and
Sun Answerbook2. I've grabbed a great variety of docbook.dtd files
besides the ones included with the above tools; they do not include
various elements such as "section id". A search of the oasis site
came up negative except for some e-mail traffic.
Is there something I can get from iptel.org?
Regards,
Michael Grigoni
Cybertheque Museum
Hi All,
Is it possible for SER to forward SIP traffic to another SER? I have a
setup below.
Where all SER clients will point to SER-A and SER-A will just forward all
request to SER-B. I know how to forward INVITES to SER-B. But is it is
possible to forward REGISTER, and all other METHODS?
SER-A --> SER-B
TIA
hi,
i have met the same problem with latest version as Peter has met. i loaded the latest version today.
In ser.cfg, when I wrote: modparam("usrloc", "db_mode", 1), the user can not registr at all.
when I wrote: modparam("usrloc", "db_mode", 2)
the user can register but for a moment the debug information shows as the fellow:
----------------------------------------------------------------
entering route[2] - user not online
val2str(): Destination buffer too short
print_values(): Error while converting value to string
submit_query(): Column count doesn't match value count at row 1
insert_row(): Error while submitting query
db_insert_ucontact(): Error while inserting contact
wb_timer(): Error while inserting contact into database
----------------------------------------------------------------
And the Location table of ser database have no records at all.
After a few minutes SER server is down. the debug information shows as the fellow:
----------------------------------------------------------------
val2str(): Destination buffer too short
val2str(): Destination buffer too short
child process 10639 exited by a signal 11
core was generated
INFO: terminating due to SIGCHLD
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
INFO: signal 15 received
-----------------------------------------------------------------------
where is wrong?
Hi All,
Could you help me whit "auth_module"?
I compiled "radiusclient-0.4.6" and "ser-0.8.14" with auth_radius module.
But when i insert module "auth_radius" in ser.cfg file I get this
message "Starting SER : cat: /var/run/ser.pid: No such file or
directory"
Could you advice me, pls.
10x in advance!
Hello all,
I want make a call which would be forked with multiple destionation, and each of forked call has a priority number(spcified in SIP RFC, the q field in SIP header.).
To do this, I first use the serctl command to add user with specified q number like this:
% hsucc- sudo serctl ul show 54760
<sip:054760@140.113.214.205>;q=1.00;expires=1003738934
<sip:54760@140.113.214.205:5060>;q=0.00;expires=3597
But while testing, ser didn't seem to follow the q header field. Could anyone can tell me if I had anything worng about this?
P.S. My ser is version 0.8.14.
Any advice would be great......
--
Best regards,
D2 mailto:leonheart.csie90@nctu.edu.tw
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my
final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using for
calling to/from PSTN is requiring me to register at their server and
authorize outgoing calls, which is something SER won't do. So I got the idea
to use asterisk between the PSTN carrier and SER for the authorization,
since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN
-----| |-----------| |-----------| |------
--------- --------- ---------
SER Asterisk Carrier
<-- auth stuff -->
<-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
Hi,
Not sure if this is simply a dumb thing to do or the configuration is
wrong... .8.14 SER is running mostly fine.
1) SER gets an invite to 123(a)domain.com
2) lookup() resolves this to 456(a)domain.com
(serctl ul show 123 returns sip:456@domain.com)
3) t_relay is called and eventually get a 408 and the URI indicates
123(a)domain.com
Logging has shown that the start of route logic is never called with
456(a)domain.com. It seems like the t_relay() is internally sitting on the
location URI. Is this intentional, to stop loops from occurring? I realize
that 'aliases' would better serve this but would like to understand what is
occurring under the covers.
Thanks in advance.
Jim
Hi all!
Is it possible to handle the body of a SIP message (or the complete
message) to an external script? I tried the exec module without luck.
regards,
klaus
Environment:
ser 0.8.14-3 on a AMD64 box
Two ethernet interfaces with SER listening only on ONE
I have a db_mode of 2 for usrloc.
What I do not understand is the once contact
information is written to the location db, it does not
get flushed at all. I still have "old" contact info in
the location db for the UA's. In the log file, I see
the recurring message:
"Keeping binding '2221111', sip:2221111@X.X.X.X" for
replication
The message is pretty self-explanatory but I have not
configured any replication. Can someone please explain
what is happening and how to fix this.
Thx in advance
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