Hello.
I just downloaded the 0.9.0 SER version from CVS and compile it
without problem. I used and old, but working (with the 0.8.14 version)
ser.cfg file with the new SER version, when i start ser i see this ERROR:
0(0) DEBUG: init_mod: domain
0(0) domain - initializing
0(0) find_mod_export: <db_use_table> in module
/usr/local/etc/ser/domaintables not found
0(0) bind_dbmod: Module /usr/local/etc/ser/domaintables does not export
db_use_table function
0(0) ERROR: domain_db_bind: cannot bind to database module! Did you forget
to load a database module ?
0(0) init_mod(): Error while initializing module domain
ERROR: error while initializing modules
The ser.cfg snippet for the domain initialization is :
#module domain
modparam("domain", "db_url", "/usr/local/etc/ser/domaintables")
modparam("domain", "domain_table", "domain")
modparam("domain", "domain_col", "domain")
My /usr/local/etc/ser/domaintables path is ok....
I'm not so sure but is this a version problem or am i missing something?.
Thanks in advance.
Ricardo Martinez.-
Hi all.
We have two ser-0.8.99-dev24 servers configured behind a load balancer, but we are having
record-route problems whereby an INVITE gets sent to an Asterisk server and the "100 Trying" reply
from Asterisk gets sent to the other SER proxy because the load balancer chooses to send the 100
reply to the other ser proxy.
Below is an INVITE with a Record-Route and two Via headers. In this message the 68.84.225.30 IP is
the **virtual** public IP of the SER proxies.
Is the top VIA (ie, Via: SIP/2.0/UDP 68.84.226.30) the reason the "100 Trying" message gets sent
to the other SIP proxy? We believe that if this VIA had the physical IP of the ser proxy that sent
them message - then all would be fine.
We've played with record_route_preset("xxx.xxx.xxx.xxx") but that doesn't seem to help with
regards to changing the top via.
Can anyone give some words of wisdom?
Regards,
Paul
U 10.2.20.21:5060 -> 10.3.0.23:5060
INVITE sip:4075551212@10.3.0.23:5060;user=phone SIP/2.0.
Record-Route: <sip:10.2.20.21;ftag=4d0c0449837b0fce;lr=on>.
Via: SIP/2.0/UDP 68.84.226.30;branch=z9hG4bK7277.557b17d3.0.
Via: SIP/2.0/UDP
192.168.0.195:5109;rport=5109;received=68.84.242.67;branch=z9hG4bKc5b09d33c79bfc1e.
From: <sip:4075551212@sip.mycompany.com;user=phone>;tag=4d0c0449837b0fce.
To: <sip:4075551212@sip.mycompany.com;user=phone>.
Contact: <sip:34075551212@68.84.242.67:5109;user=phone>.
Supported: replaces.
Proxy-Authorization: DIGEST username="4075551212", realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@sip.mycompany.com;user=phone",nonce="41c1f8a5c69796cff47ad459a19632f0d31105ee",
response="5613a719bd6a2b3b7315522611e09653".
Call-ID: 71cfc2d5a0fbdae1(a)192.168.0.195.
CSeq: 12663 INVITE.
User-Agent: Grandstream BT100 1.0.5.16.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 411.
X-host: 10.2.20.21 .
P-hint: 3 && Record-Route.
.
v=0.
o=3212514234 1 8000 IN IP4 192.168.0.195.
s=SIP Call.
c=IN IP4 68.84.226.26.
t=0 0.
m=audio 35338 RTP/AVP 98 18 15 4 2 9 0 8 101.
a=sendrecv.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
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Hi,
I encounter the following error message when running ser 0.8.14 and
rtpproxy ( nathelper enable).
Dec 17 12:55:16 officehk123 /usr/local/sbin/ser[7930]: ERROR:
extract_body: message body has lenght zero
Dec 17 12:55:16 officehk123 /usr/local/sbin/ser[7930]: ERROR:
force_rtp_proxy2: can't extract body from the message
Dec 17 12:55:16 officehk123 /usr/local/sbin/ser[7930]: ERROR: on_reply
processing failed
Dec 17 13:01:30 officehk123 /usr/local/sbin/ser[7930]: ERROR:
extract_body: message body has lenght zero
Dec 17 13:01:30 officehk123 /usr/local/sbin/ser[7930]: ERROR:
force_rtp_proxy2: can't extract body from the message
Dec 17 13:01:30 officehk123 /usr/local/sbin/ser[7930]: ERROR: on_reply
processing failed
Dec 17 13:02:31 officehk123 /usr/local/sbin/ser[7949]: ERROR:
open_reply_pipe: open error (/tmp/ser_receiver_8025): No such file or
directory
Dec 17 13:02:31 officehk123 /usr/local/sbin/ser[7949]: ERROR:
fifo_reply: no reply pipe /tmp/ser_receiver_8025
Is this a serious problem ? Do I need to solve it?
Hope someone can answer me pls.
Thomas
Hi list,
I have a doubt about SER forwarding to GW.
Using the rule below, everything works the way it should work:
if(uri=~"^sip:55.*"){
strip(2);
prefix("55");
rewritehost("1.2.3.4");
} else {
sl_send_reply("404", "Not Found");
break;
};
But if I change for the rule below, my SIP client receives a 503 (Service Unavailable) message:
if(uri=~"^sip:055.*"){
strip(3);
prefix("55");
rewritehost("1.2.3.4");
} else {
sl_send_reply("404", "Not Found");
break;
};
Nbody can help me?
Thanks,
Vitor Brasileiro.
Hi list,
I'm using the accounting feature on SERWEB and everything works fine, but I'd like to improve this feature calculating the cost of each call. I'm planing to create another table, for example, costs, and show the call length times the cost. Do u have this modification already?
Vitor Brasileiro.
Hi All,
I have defined extension 2000 - 8000 on my SIP server.
All of them can make calls to PSTN, but my boss wants now, to block all 2XXX
extensions to call PSTN but they should still be able to call other extensions
from 2000 - 8000.
I was thinking of adding something like this line,
if ((uri=~"^(sip:)?001[0-9]*@([a-z]+\.)?mydomain\.com") ||
(uri=~"^(sip:)?001[0-9]*@123\.45\.67\.89"))
but the condition will be if calls originated from 2XXX going to pstn, which
would look like 00XXXXXXXXX, it's more like this line
if (((uri=~"^(sip:)?001[0-9]*@([a-z]+\.)?mydomain\.com") ||
(uri=~"^(sip:)?001[0-9]*@123\.45\.67\.89")) &&
(((uri=~"^(sip:)?001[0-9]*@([a-z]+\.)?mydomain\.com") ||
(uri=~"^(sip:)?001[0-9]*@123\.45\.67\.89")))
but here I compare the destination uri, where should I compare the source?
hope you understand my question, as I'm having a hard time explaining in english.
Regards
Ron
Hi Guys,
I'm trying to test call conferencing with SER and
SEMS. With my present configuration, voicemail
function is working nicely. Can I use the xlite client
to test the call conferencing? Do i have to configure
SER and SEMS for this to work?
Any help is highly appreciated.
Regards,
Lakmal
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That script is not called from ser.cfg. It is completely independent from
SER, other than it uses the ser mysql database to get information. Just
change that DB connection parameters in the top of the script and run it
from the command line.
Note - It does require that you have the perl DBI and MySQL modules
installed.
Darren
> -----Original Message-----
> From: Bruno Lopes F. Cabral [mailto:bruno@openline.com.br]
> Sent: Thursday, December 16, 2004 12:27 PM
> To: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Does SER do that?
>
> Err... do you mind to share the ser.cfg snippet that use
> your script? I'm at early stages of writing something
> like your script and perhaps I could use it directly or
> as base to my work
>
> thanks for sharing it
>
> Cheers
> !3runo
>
> Darren Nay wrote:
> > Sure, I am happy to share it.
> >
> > Keep in mind. This script assumes that you are using a phone number
> (DID)
> > username scheme. So if your usernames aren't in a standard DID format
> of
> > 1XXXXXXXXXX or +1XXXXXXXXXX then you might have to tweak it some to make
> it
> > work correctly for you.
> >
> > Give it a try and give me some feedback. Thanks!
> >
> > Darren Nay
> >
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Sure, I am happy to share it.
Keep in mind. This script assumes that you are using a phone number (DID)
username scheme. So if your usernames aren't in a standard DID format of
1XXXXXXXXXX or +1XXXXXXXXXX then you might have to tweak it some to make it
work correctly for you.
Give it a try and give me some feedback. Thanks!
Darren Nay
> -----Original Message-----
> From: Matt Schulte [mailto:mschulte@netlogic.net]
> Sent: Thursday, December 16, 2004 11:58 AM
> To: Darren Nay; Alberto Martínez; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Does SER do that?
>
> Nice! Care to share the app? :-) heehee
>
> -----Original Message-----
> From: Darren Nay [mailto:dnay@IONOSPHERE.net]
> Sent: Thursday, December 16, 2004 10:39 AM
> To: Matt Schulte; Alberto Martínez; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Does SER do that?
>
>
> Yes, this can be done to some degree.
>
> I've written a perl script that queries the "acc" table and keeps track of
> current calls. The only issue is that occasionally if a call is not
> broken down correctly (ie. SER did not receive the BYE) then you may have
> some calls reported to stay up when actually they are not.
>
> However, the best way to keep track of concurrent calls is to use
> mediaproxy. It always knows exactly how many calls are up because it
> carries the audio for each call. Much more accurate than using sers "acc"
> table.
>
> That being said, we still use the "acc" table method because we prefer
> that the audio streams not ride our network. It saves cost on bandwidth
> and we've found that the voice quality is better directly from IAD to
> PSTN/IAD and not relayed through an RTP proxy.
>
> Darren
>
>
> > -----Original Message-----
> > From: Matt Schulte [mailto:mschulte@netlogic.net]
> > Sent: Thursday, December 16, 2004 11:38 AM
> > To: Alberto Martínez; serusers(a)lists.iptel.org
> > Subject: RE: [Serusers] Does SER do that?
> >
> > Yes, yes, and maybe.
> >
> > > I would like to know if SER can notify by anyway to other
> > > application
> > of the start and end of
> > > VoIP transmissions in order to maintain a control in real-time of
> > > calls
> > made by the users.
> >
> > This one maybe a little more tricky, I'm looking for such an app
> > myself. I suppose this could be done via the acc module.
> >
> > -----Original Message-----
> > From: Alberto Martínez [mailto:amartinez@astrasoft.es]
> > Sent: Thursday, December 16, 2004 9:21 AM
> > To: serusers(a)lists.iptel.org
> > Subject: [Serusers] Does SER do that?
> >
> >
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: MD5
> >
> > Hello,
> >
> > I am new in SER. I would like to ask you if SER is able to do
> > somethings I am looking for. I would like it to check if the users who
> > try to connect with it by SIP are correct, checking the login info
> > and, if they are, redirect the connection to the VoIP provider. If the
> > user who is trying to connect is not set up in SER it must be
> > rejected.
> >
> > I would like to know if SER can notify by anyway to other application
> > of the start and end of VoIP transmissions in order to maintain a
> > control in real-time of calls made by the users.
> >
> > Thank you.
> >
> > Best regards,
> > Alberto
> >
> > -----BEGIN PGP SIGNATURE-----
> > Version: 2.6
> >
> > iQEVAwUAQcGn+HHoJ4bX5QlXAQEyCAf/T1bEaVlXWW+krVKlFl5yxJLLv3uFH9q0
> > qddg/6+YDx1lXnoigqxTWtbpvR0uMnVnFRqueifpWltzkawcgHYtVRNmHV1+bpDn
> > awrgE0UFb2zlaUsFf+INqaaFuXGrztgCA0jcwh4gvhAYzQX8L/32g9EsRrZNNpie
> > WK8uYxp7N9Pw6MqkQwvSrCrbuIt+umP+tbYJcza83d5+Bb/yNXn8ePY1ztnWYfZ+
> > 5vp8jimNO/93T9/k26zs1hdGEtW68tCIbMeWu37FFmPbRthlGQM/a5Ku76ZfhjjO
> > Lmv9fi3spkH+uOfgjlX6YFLdgEW1bxk7bEL0yPf7hB+2Awgqy3QIfw==
> > =tOcm
> > -----END PGP SIGNATURE-----
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Nice! Care to share the app? :-) heehee
-----Original Message-----
From: Darren Nay [mailto:dnay@IONOSPHERE.net]
Sent: Thursday, December 16, 2004 10:39 AM
To: Matt Schulte; Alberto Martínez; serusers(a)lists.iptel.org
Subject: RE: [Serusers] Does SER do that?
Yes, this can be done to some degree.
I've written a perl script that queries the "acc" table and keeps track of current calls. The only issue is that occasionally if a call is not broken down correctly (ie. SER did not receive the BYE) then you may have some calls reported to stay up when actually they are not.
However, the best way to keep track of concurrent calls is to use mediaproxy. It always knows exactly how many calls are up because it carries the audio for each call. Much more accurate than using sers "acc" table.
That being said, we still use the "acc" table method because we prefer that the audio streams not ride our network. It saves cost on bandwidth and we've found that the voice quality is better directly from IAD to PSTN/IAD and not relayed through an RTP proxy.
Darren
> -----Original Message-----
> From: Matt Schulte [mailto:mschulte@netlogic.net]
> Sent: Thursday, December 16, 2004 11:38 AM
> To: Alberto Martínez; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Does SER do that?
>
> Yes, yes, and maybe.
>
> > I would like to know if SER can notify by anyway to other
> > application
> of the start and end of
> > VoIP transmissions in order to maintain a control in real-time of
> > calls
> made by the users.
>
> This one maybe a little more tricky, I'm looking for such an app
> myself. I suppose this could be done via the acc module.
>
> -----Original Message-----
> From: Alberto Martínez [mailto:amartinez@astrasoft.es]
> Sent: Thursday, December 16, 2004 9:21 AM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Does SER do that?
>
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: MD5
>
> Hello,
>
> I am new in SER. I would like to ask you if SER is able to do
> somethings I am looking for. I would like it to check if the users who
> try to connect with it by SIP are correct, checking the login info
> and, if they are, redirect the connection to the VoIP provider. If the
> user who is trying to connect is not set up in SER it must be
> rejected.
>
> I would like to know if SER can notify by anyway to other application
> of the start and end of VoIP transmissions in order to maintain a
> control in real-time of calls made by the users.
>
> Thank you.
>
> Best regards,
> Alberto
>
> -----BEGIN PGP SIGNATURE-----
> Version: 2.6
>
> iQEVAwUAQcGn+HHoJ4bX5QlXAQEyCAf/T1bEaVlXWW+krVKlFl5yxJLLv3uFH9q0
> qddg/6+YDx1lXnoigqxTWtbpvR0uMnVnFRqueifpWltzkawcgHYtVRNmHV1+bpDn
> awrgE0UFb2zlaUsFf+INqaaFuXGrztgCA0jcwh4gvhAYzQX8L/32g9EsRrZNNpie
> WK8uYxp7N9Pw6MqkQwvSrCrbuIt+umP+tbYJcza83d5+Bb/yNXn8ePY1ztnWYfZ+
> 5vp8jimNO/93T9/k26zs1hdGEtW68tCIbMeWu37FFmPbRthlGQM/a5Ku76ZfhjjO
> Lmv9fi3spkH+uOfgjlX6YFLdgEW1bxk7bEL0yPf7hB+2Awgqy3QIfw==
> =tOcm
> -----END PGP SIGNATURE-----
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
