hi, there
I'm a newbie of SER, I wounder if SER can register as a SIP client?
In this case, SER working as a client can recieve/send calls from a remote SIP server.
just like Asterisk has done, adding a line in it's sip.conf:
register > user:secret:authuser@remoteSIPhost:port/extension
Thanks!
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Hello,
I have installed and configure SER 0.9.6 in our office.
<SIP Phone> <----><SER><----> <SIP Phone>
Interoffice SIP phone to SIP phone calls are ok.
I want to connect to other SIP Provider who is conncted to PSTN, so that
from my office SIP phone I can call PSTN numbers
<SIP Phone> <----><SER><----> <SIP Phone>
|
|
<INTERNET>
|
|
<SIP Provider>
|
|
|
<PSTN>
Hope you guys have understood the scenario.
Regards
Sarfaraz
Hi,
I can not enable ipv6 on ser.
I have been set the following configuration on ser.cfg.
listen=2001:238:202:0:230:1bff:feb8:32b9
listen=140.113.131.41
However, I found the following information on /var/log/message:
Mar 9 20:10:09 TIPRA-41 ser: WARNING: fix_socket_list: could not rev.
resolve 2001:238:202:0:230:1BFF:FEB8:32B9
Mar 9 20:10:09 TIPRA-41 ser: WARNING: fix_socket_list: could not rev.
resolve 2001:238:202:0:230:1BFF:FEB8:32B9
I can't receive any 200 OK ,when I had been sent REGISTER using my
UA(ipv6 enabled).
The following is my "ifconfig" information:
TIPRA-41:/home/rhliou # ifconfig
eth1 Link encap:Ethernet HWaddr 00:30:1B:B8:32:B9
inet addr:140.113.131.41 Bcast:140.113.131.127 Mask:255.255.255.128
inet6 addr: 2001:238:202:0:230:1bff:feb8:32b9/64 Scope:Global
inet6 addr: fe80::230:1bff:feb8:32b9/64 Scope:Link
UP BROADCAST NOTRAILERS RUNNING MULTICAST MTU:1500 Metric:1
RX packets:1785125 errors:0 dropped:0 overruns:0 frame:0
TX packets:20517 errors:0 dropped:0 overruns:0 carrier:0
collisions:759 txqueuelen:1000
RX bytes:158891946 (151.5 Mb) TX bytes:3155181 (3.0 Mb)
Interrupt:169 Base address:0xc000
lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:568 errors:0 dropped:0 overruns:0 frame:0
TX packets:568 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:269972 (263.6 Kb) TX bytes:269972 (263.6 Kb)
Thanks in advance
Best regards,
Ren-Huang Liou
Hi Friends,
I would like to know your experience on using any of the softphones on your way,, can you suggest me to have a try with your likely ones,,
am running ser release v_0_9_3
Thank You,
Se.
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Dear all,
is there a function or some method to find out whether a
specific user is registered or not? I'd like to use it in
ser.cfg.
Something like 'serctl ul show'
Thank you in advance,
Best Regards,
Balazs Godor
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After talking with clona in #ser on Freenode, I've learned that CVS-HEAD
has some great XML-RPC implementation. clona also mentioned that a new
release might not be far away.
Would anyone mind commenting on how much the current XML-RPC implementation
might change between now and the next release? I'd love to use it but am
concerned about developing the necessary clients to interact with it now,
and having to redo them when the next release comes out.
Thanks guys!
-- Nick
e: nick.hoffman(a)altcall.com
p: +61 7 5591 3588
f: +61 7 5591 6588
If you receive this email by mistake, please notify us and do not make any
use of the email. We do not waive any privilege, confidentiality or
copyright associated with it.
Hi,
just wonder if it would be interesting to have path also included in
fifo ul_show_contact?
br hw
--
Helge Waastad
Senior Konsulent
Systemavdelingen
Smartnet
I have a box with a bunch of ip addedes assigned to eth0:xx. I want rtpproxy
to bind to a particular address and send RTP packets from that address. I
see an option -l addr1 but can't figure out if it's the address rtpproxy
will expect to receive packets or send from. The rtprpoxy --help isn't
exactly helpful. Can someone englighten me?
Hello everyone,
I am trying to debug why my rtpproxy isn't working. I have the following
setup, on my LAN.
softphone (192.168.1.100) -> openser/rtpproxy (192.168.1.10) -> asterisk (
192.168.1.12)
The rtpproxy is running and I see commands flying thru it.
the following route works
if(method=="INVITE") {
if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {
forward(192.168.1.12,5060);
};
}
when I replace it with this route
if(method=="INVITE") {
if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {
forward(192.168.1.12,5060);
};
force_rport();
force_rtp_proxy();
}
I get dead air while asterisk logs show that my test message is playing. How
should I proceed to debug this?
ScriptHead
