Quick question for everyone:
I have dial strings coming in as "(770) 555-1212" and need them passed
on as "50507705551212" What is the cleanses and most efficient way of
doing this? (A link to a page showing basic config may not be enough as
I am new to SER)
Thank you in advance for your support.
Mark Baker
678-468-7061
Hi
It seems that he wants to do it dynamically...
>From: GR S <gr_sh2003(a)yahoo.com>
>To: Anil Bhati <abhati_voip(a)hotmail.com>
>CC: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] t_relay through c code
>Date: Wed, 16 Jun 2004 21:59:57 -0700 (PDT)
>
>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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Hi, can you send me a copy of the binary ngrep already patched for newline in SIP headers ?
The sip_router/utils/ngrep source is not compiling in my system.
Thanks
Ezequiel
I have the voicemail function with SEMS working ok but I haven't been
able to get the custom greetings to work. The sems.conf file looks pretty
straight forward but I haven't figured out why I can't get the custom
greetings to work. I recorded a greeting in wav format and placed it in
/usr/local/lib/sems/audio/ser.vci.net/510.wav My SIP ID is 510(a)ser.vci.net
I have also tried using /usr/local/lib/sems/audio/192.168.1.55/510.wav
because when the email is sent out the domain (in the email) is listed as
"192.168.1.55" instead of "ser.vci.net" I also tried
/usr/local/lib/sems/audio/192.168.1.55/510(a)ser.vci.net.wav and that doesn't
work either.
If it matters I actually used a symbolic link from
/usr/local/lib/sems/audio/192.168.1.55 (and
/usr/local/lib/sems/audio/ser.vci.net) to /var/greetings/ser.vci.net because
serweb puts the custom greetings there. The paths work though.
Most of the sems.cfg file is defaulted. I beleive the only thing I
changed was the smtp server address.
Bill Dunn
# optional parameter: announce_path=<path>
#
# - sets the path where announce files are searched for
# - the file to be played is determined the following way:
# <announce_path>/<domainname>/<username>.wav
# if this file is not available <announce_path>/<default_anounce> is used
announce_path=/usr/local/lib/sems/audio/
# parameter: default_announce=<filename>
#
# - sets the name of the default announce WAV file
default_announce=default_en.wav
As far as I know Asterisk is the only router that does transcoding
between call legs including both media and signaling. I have tested
only SIP to H323 (GSM to G711).
Cisco will support transcoding beginning with Q3 for 2621XM family
(between G729 and G711).
Regards
Adrian
Does anyone know of a media proxy that will transcode between g.711 and
iLBC?
Thanks,
Jamey
mawali(a)news.icns.com wrote:
> You must have noticed I have been asking this question for last 4 days.
>
Yeah! :)
> Lets work together.
ok!
> I do not force_rtp_proxy before t_relay_to_udp because it breaks a lot
> more things than just NAT'd clients. When I dont use force_rtp_proxy,
> everything works, I can talk from NAT'd cleints to non-NAT'd, but
> asterisk doenst work for clients through NAT. I have noticed that the
> rtpproxy is still not getting engaged.
>
>
>
I noticed that in a configuration like this
X-lite call 9916 -> on ser force_rtp_proxy(); and
t_relay_to_udp("xxx.xxx.xxx.xxx","5090"); -> asterisk at 5090
ASTERISK send voice to RTPPROXY but XLITE send directly voice to
ASTERISK instead RTPPROXY...
> So t_relay_to_udp is not what we need, it may be a B2BUA. I wish
> someone could give us a simple answer before we dive into it ourselves.
>
>
>
B2BUA? What is that?
> Regards
>
>
>
>
Thanks! Bye...
> On Thu, 1 Jul 2004, Aldo Armiento wrote:
>
>
>
>> Hi all!
>> I know the argument is very famouse, I think I have read all ser.cfg
>> on the net....
>> The problem is that when I call extension 9916 I don't hear asterisk
>> music-on-old sound...
>>
>>
>> The scenario:
>>
>> ser at port 5060
>> asterisk at port 5090
>> rtpproxy
>>
>> I would like that, when an UA (x-lite for example) call 9916 UA can
>> hear "music-on-hold" in asterisk.
>>
>>
>
>
>
>
Hi
I posted the below message a couple of days ago. I want all numbers
starting with 8 to go to a SIP PSTN gateway. The problem is that neither
forward() nor relay_to_udp() work through NAT. I want to be able to use
rtpproxy (or mediaProxy for that matter) for such connections. I see that
forward() and relay_to_udp() both bypass the nathelper rtpproxy. Anyone
knows the best way to ensure rtpproxy involvement in this scenario. If
rtpproxy is not involved, I cannot use the PSTN gateway from NAT'd
endpoints. There is gotta be a way.
/-------------------\
| SER (port 5060) + |
SIP UA <----- NAT ------> | rtpproxy |
| |
| SIP_PSTNGW |---->PSTN
| (* at port 7060) |
\___________________/
Previous post:
Hi
I am trying to send all my calls to a PSTN gateway (asterisk with digium
cards running on the same machine on 7060). I can use forward, but then I
cannot use the solution SER provides me by using NAT helper. Consider the
forwarding script:
      if (uri=~"^sip:[8].*@")
{
forward(209.7.34.58,7060);
    }
Here it works fine if I am calling from a public IP, but will not work if
I am calling from a NATed client since this stateless forwarding will
take SER out of the picture, and rtpproxy will not be used. I have also
tried using t_relay_to_udp, but is is doing the same, the below lines are
still not making SER to proxy the request.
if (uri=~"^sip:[8].*@")
{
t_relay_to_udp("209.7.34.58","7060");
break;
}
What would be the best way of doing it. I want SER to act as a proxy
(B2BUA?) between my caller and the PSTN asterisk server. I want to be able
to use nathelper for this scenario.
Regards
Hi, i am trying to upgrade to 0.8.13 and get Segmentation Fault when run SER.
All the CVS source compile fine and the configuration test with "-c" pass but SER dont start.
I have the core dump file, if any of the developers can help me please request it off the list.
[root@billing sip_router]# ./ser -f ./ser.cfg -E -D -d
Listening on
udp: 172.16.0.1 [172.16.0.1]:5060
tcp: 172.16.0.1 [172.16.0.1]:5060
Aliases:
tcp: labo:5060
tcp: labo.arcotel.net:5060
udp: labodomain:5060
udp: labo.labodomain.net:5060
*: labodomain.net:*
WARNING: no fork mode
acc - initializing
Segmentation fault
Thanks
Ezequiel Colombo
