sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[Serusers] Cisco phones port range
by Joao Pereira 18 Nov '05

18 Nov '05

Hi Im using Cisco IP 7940 (with SIP firmware) and I want to force him to put the media stream in some specific port. To do it I put this in the Cisco configuration file: start_media_port: 8000 end_media_port: 9000 but the Cisco IP phone boots and doesnt accept these ports, and assumes the defaults (16384-32766). Even when I put these ports directly in the phone configuration, he doesnt accept them. How can I change the RTP ports in the Cisco IP phone? ( Like in Xlite we do: "System Settings"-> "Network" -> "Listen RTP port" ) Thanks Joao Pereira

1 0

[Serusers] SER download site continuously "busy"
by Davey, Douglas A (Doug) 18 Nov '05

18 Nov '05

I have been unable to download SER from ftp://ftp.berlios.de/pub/ser/ <ftp://ftp.berlios.de/pub/ser/> for several days. Several colleagues have reported the same problem. The ftp server appears to be "busy" at all times even though it is never near its maximum of 95 users. I have downloaded SER in the past for several projects without problems, so this seems to be a relatively recent change. Are there mirror sites from which to download a SER installation package? I found source packages on http://developer.berlios.de/projects/ser/ <http://developer.berlios.de/projects/ser/> . Are pre-built executable SER packages no longer distributed? Thanks. Doug Davey

1 0

Re: [Users] RE: [Serusers] Re: [Serdev] Inaccurate Radius Accounting
by 'Jan Janak' 18 Nov '05

18 Nov '05

On 17-11-2005 10:21, Lenir wrote: > In this case the radius proxy wont work, because you never can anticipate > who hangs up the call, thus radius wont know who hung up the call...Besides > all other voice applications/hardware (SIP and H323) that use radius do not > behave like that, the Called-Station-ID ALWAYS remains the same, as with the > Calling-Station-ID. Could you name those SIP applications that behave the way you describe ? Jan.

2 3

[Users] new group feature
by Bogdan-Andrei Iancu 18 Nov '05

18 Nov '05

Hi, I have implemented a new group feature. So far we have group support with fixed matching: user X belongs to group G, etc.. This proves to be inefficient when you have huge block of users belonging to small numbers of groups. So I did some regular expression based groups: if user@domain matched a RE -> belongs to group G very usefull if try to provisions routing logic based on groups. I developed this new functionality inside the group module since there is about same functionality and it's too small to be a separate module. any pro or against comments / suggestions before committing? regards, Bogdan

2 3

[Users] Re: How does openser use the nathelper
by Klaus Darilion 18 Nov '05

18 Nov '05

Hi messa! I suggest you to take a look on the getting_started document from onsip.org. It uses mediaproxy module and the mediaproxy (just another flavour of nathelper+rtpproxy). You can't integrate only nathelper to vocal. Use openser+nathelper as replacement for vocal or together with vocal. btw: you could resolve the problem by using clients which support ICE. regards klaus PS: please no direct email. Send your emails always to the list. messa(a)innsof.com wrote: > Hi Klaus > > I want you to help me. I want to integrate nathelper in my vocal > system to > fix the problem of NAT traversal because the STUN solution is not good for > me. With stun, i cannot make in the same moment a call to another client > in a remote network and to a client in the same network. ANd NAthelper do > it good for openser. So i want to know, the architecture of openser. How > call the nathelper module. I want to have the architecture of this > nathelper module. > > Thanks in advance! > > Best regards > > Serge > >

1 0

[Users] Re: How does openser use the nathelper
by Klaus Darilion 18 Nov '05

18 Nov '05

messa(a)innsof.com wrote: > Hi Klaus > > I want you to help me. I want to integrate nathelper in my vocal > system to > fix the problem of NAT traversal because the STUN solution is not good for > me. With stun, i cannot make in the same moment a call to another client > in a remote network and to a client in the same network. ANd NAthelper do > it good for openser. So i want to know, the architecture of openser. How > call the nathelper module. I want to have the architecture of this > nathelper module. > > Thanks in advance! > > Best regards > > Serge > >

1 0

[Serusers] Contact field in SIP HF between asterisk + ser
by harry gaillac 18 Nov '05

18 Nov '05

Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be advice. One box --------------------------- | ---------------- | | | asterisk pbx | | | ---------------- | | || | | ---------- ---------- | | SER |====|NAT box |==== private network | ---------- ---------- --------------------------- Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Contact field in SIP HF between asterisk + ser
by harry gaillac 18 Nov '05

18 Nov '05

1 0

[Serusers] RTPproxy how to
by crazy 18 Nov '05

18 Nov '05

hi all i donot know howto set the port of RTPproxy, my server public ip such as 202.97.123.123, i want to set RTPproxy use the 5080-5082 ports,what can i do for it. client can use 202.97.123.123:5080-5082 as a outbound proxy!

2 1

[Users] Re: [Serdev] How to gaurd PSTN gateway with openser + NAT
by Lokesh Kumar 18 Nov '05

18 Nov '05

Hello Nils, Attached is the mentioned files you asked me to send. As you said, after disabling gnutls in configure, I can compile it. It works fine . Thanks for your kind help. One important question I would like to ask you. Suppose I am having 100 SIP clients, all of them are behind their respective 100 NAT's, The NAT devices are assigned public IP dynamically. What besides openser do I need on my SIP server to traverse those NAT's. Assume NAT's does not support STUN. Do I need to put STUN or Mediaproxy/RTPProxy server on my SIP server. If not , then how could i can manage such network. Thanks a lot for your kind help Best Regards Lokesh ----- Original Message ----- From: "Nils Ohlmeier" <lists(a)ohlmeier.org> To: "Lokesh Kumar" <lokesh(a)interacesso.pt> Sent: Thursday, November 17, 2005 10:14 PM Subject: Re: [Serdev] How to gaurd PSTN gateway with openser + NAT > Hello Lokesh, > > could you do me a favor and send me config.h and config.log from the directory > where you tried to compile 0.9.2? > > After you send me the files you could try the following command > './configure --disable-gnutls' and afterwards 'make'. I think that should help > to compile sipsak-0.9.2 on your system. > > Thanks > Nils > > On Friday 18 November 2005 19:54, you wrote: > > Hello Nils, > > > > When I do make in the source directory of sipsak-0.9.2, it gives me error > > like this. > > > > /usr/lib/libgnutls-openssl.so: undefined reference to > > 'asn1_delete_structue' /usr/lib/libgnutls-openssl.so: undefined reference > > to 'asn1_create_element' /usr/lib/libgnutls-openssl.so: undefined reference > > to 'asn1_der_decoding' collect2: ld returned 1 exit status > > make[1]: ***[spsak] Error 1 > > make[1]: Leaving directory '/usr/src/sipsak-0.9.2' > > make: *** [all] Error 2 > > > > But there is no such problem with 0.9.0 version. > > > > Regards > > > > Lokesh > > > > > > ----- Original Message ----- > > From: "Nils Ohlmeier" <lists(a)ohlmeier.org> > > To: <serdev(a)iptel.org> > > Cc: "Lokesh Kumar" <lokesh(a)interacesso.pt>; <deepak.dhiman(a)uf4.net>; > > <serusers(a)iptel.org> > > Sent: Thursday, November 17, 2005 6:05 PM > > Subject: Re: [Serdev] How to gaurd PSTN gateway with openser + NAT > > > > > Hello Lokesh, > > > > > > On Friday 18 November 2005 01:27, Lokesh Kumar wrote: > > > > I installed openser, it is working fine. sipsak-0.9.0 is also working > > > > fine, > > > > > > previously i was using 0.9.2 version of sipsak that is having problem. > > > > > > I would really appreciate it if you could describe we what problems you > > > > had > > > > > with sipsak 0.9.2. Then I could try to fix it, if it is a bug in sipsak. > > > > > > Thanks > > > Nils

2 1

← Newer
1
...
25
26
27
28
29
30
31
...
58
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users November 2005