Hello
is someone using serweb with OpenSER 1.2.0 and postgres?
I want to know if it is possible.
I tried the version in
http://voip-info.org/wiki/view/OpenSER+1.1.x+and+SerWEB+0.9.4
but it isn't working and this error happens:
Function aggregate_methods() doesn't exists. Try install Classkit
extension. http://pecl.php.net/package/classkit
I already tried classkit and the problem remains.
Is someone using serweb + OpenSER 1.2.0 + postgres?
Thanks
regards
Joao Pereira
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______________________________________________
João Gomes Pereira
FCCN
Av. do Brasil, nº 101
1700-066 Lisboa
tel: +351 218 440 100 - fax: +351 218 472 167
email|SIP: joao.pereira(a)fccn.pt
http://www.fccn.pt
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Hi,
I am using openser1.2
If the presence module receives a PUBLISH:
- without E-Tag, it gets stored and if needed, a NOTIFY is generated
... works for me.
- with an E-tag, for which this openser has not seen the first PUB
(the one without E-tag) ... it gets discarded/ignored?
How is the e-tag used in processing publishes and then generating notifies?
I ask this because in my (peculiar) setup with two openser's sending
publish to each other, depending on the order I start the elements (2
phones, 2 opensers), I don't get always a working behavior:
- if the opensers are running before the phones start ... no problem
... I guess as they see the PUB without e-tag, it all goes fine
- if the boot order is : openser1 ... then phone 1 ... sometime after
(enough to miss the first PUB without e-tag) ... openser2 ... then
phone2 ... Phone 2 does not receive NOTIFYs of status changes from
phone1, eventhough the PUBlish messages do get to openser2 ... but
somehow, they get "discarded" ...
Regards,
Cesc
Hi all,
I'm occured in this problem already seen here:
http://www.openser.org/pipermail/users/2006-March/003540.html
My problem occured between two different UA and only in case of a
Re-INVITE for a T.38 fax communication.
Im' using openser 1.2.0.
I'm doing some translation with From for a correct CLI screening.
I'm also checked that the RR parameters are correctly mirrored.
find attached here the debug.log and a network trace.
thank you
:tele
Hi
I need to use the SER proxy over NAT.
For that, I only need that all SIP communcation between SER and other Sip
Server will always work on a single port - altough the SIP client that
connects SER use different port for REGISTER and INVITE
Is it possible?
Thank you
Saar Carmi
--
Saar Carmi
Mobile: 054-7782417
Email: saarcarmi(a)gmail.com
Hi all,
I would like to encourage you to test and use new module search page.
It was placed here http://www.iptel.org/ser/doc/search
Additionally some links to this page were added by greger in
appropriate places across the iptel.org portal.
Note that search-page is constructed using AJAX "technology" and used
frameworks limit the amount of browsers that may correctly display
this page.
Supported browsers are:
- IE >= 6;
- Opera >= 9;
- FF >= 1.6;
Should you experience any problems or if you have any comments or
suggestions, please let me know.
If you are unable to use it (because of browser) and you would like to
have a search page that support even lynx and have similar
functionalities like described in this functional specification:
http://www.iptel.org/ser_documentation_dynamic_search_page_functional_speci…,
write to me too. If there will be huge demand for less interactive
but more universal and better supported version of that search page I
will consider creating it.
Best regards,
Marcin :)
Hello Users,
I want to know , is OpenSER supports Highly expensive IBM machines for the
VoIP services.
the Machines like
IBM System p™ servers
IBM IntelliStation(r) POWER™
IBM BladeCenter(r)
IBM System i5™, eServer™ i5 and eServer™ iSeries™ systems
These are Unix
I need to install Fedora Core 6 or 5 or SUSE linux into them,
Please help me ........................
--
Thanks and Regards
Ravi Prakash Sunkara
ravi.sunkara(a)hyperion-tech.com
M:+91 9985077535
www.hyperion-tech.com
Client and Parent company :- www.august-networks.com
Hi everybody,
OpenSER 1.2.0 has now the capability to do DNS based failover, according
to RFC3263 (http://www.ietf.org/rfc/rfc3263.txt).
The SIP resolver was enhanced to to be able to save and resume later the
DNS queries in order to get all possible IP destinations. The resolving
process is step-by-step done (get next IP only on demand) to minimize
the total number of DNS queries. So having this support does not imply,
in normal processing, more load on the DNS server. Additional queries
are done only when needed (after failure detected).
The scanning for new IP destinations is done by the SIP resolver on all
DNS levels: NAPTR, SRV, A.
Both core and TM are using this new feature.
In core, the stateless forwarding can do only DNS-based failover at
transport level (if no egress interface found or send operation
failed due whatever reason).
In TM, the DNS-based failover is extended to transaction level. If the
transaction completes with 503 or 408 with no reply, automatically, a
new branch will be fork if any destination IP can be found by the DNS
resolver.
Read more here http://www.openser.org/docs/modules/1.2.x/tm.html#AEN103
For controlling this feature use:
- newly added core parameter "disable_dns_failover" - use to
generally disable the DNS-based failover. By default is false.
- in TM, t_relay() take a new flag for turning off the DNS-based
failover. This setting is per transaction. By default, the failover is done.
Any feedback is appreciated.
regards,
Bogdan
I cannot seem to get avp_db_query to work with dbtext. My query looks
like this..
avp_db_query("SELECT uuid FROM subscriber WHERE username = '$rU'",
"$avp(s:callee_uuid)");
I receive the following error: avpops:db_query_avp: error - cannot do the query
Are there any other options for doing a query on a dbtext table from
within the config?
Thanks,
Daryl
Hello,
I use opsnser 1.2 with nathelper and I see that not all c= in the SDP are
changed.
with some user agent it's ok but for some else, this is a problem.
any solutions ? flag ?
Laurent
exemple: SDP in the 200 ok modified with nathelper (openser -> UA )
v=0.o=CiscoSystemsSIP-GW-UserAgent 5635 2443 IN IP4 xxx.xxx.127.1.s=SIP
Call.c=IN IP4 xxx.xxx.127.1.t=0 0.m=audio 43594 RTP/AVP
8 106.c=IN IP4 yyy.yyy.14.234.a=rtpmap:8 PCMA/8000.a=rtpmap:106
telephone-event/8000.a=fmtp:106 0-15.a=ptime:20.a=nortpproxy:yes..