Hello,
I ran out into some problem when implementing Call Forwarding feature using SER.
My test setup:
192.168.25.101 - XLite ---> INVITE UA(sip:444)
192.168.25.163 - ser; rewrites to Address of record of Asterisk (sip:1111 instead of sip:444)
192.168.25.169 - PBX Asterisk
As I see from log files, SER sends INVITE 2 times,
then Asterisk sends 3 times 200 status etc . until BYE is sent (here everything correct....)
Here is my configuration script:
==================================
route{
. . .
record_route();
#--rewrite URI
if (method=="INVITE"){ #--call forwarding
setflag(1);
route(1); #--cfw
if(!isflagset(1)) {
break;
}
lookup("location")
...
};
route[1] {
log(1, "forwarding\n");
if (!if_activated("to")) {
#--continue regular call set up
break;
};
if (!lookup_dn()){
sl_send_reply("404","Not Found");
resetflag(1); #--continue!
break;
};
}
Here if_activated checks whether the feature was activated; lookup_dn() rewrites URI to new UA and call append_branch()
Any ideas what I am doing wrong?
if (!t_relay()) {
sl_reply_error();
};
}
===========
Any help will be highly appreciated!!!
Thanks, Tina,
software engineer
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Hello,
I ran out into some problem when implementing Call Forwarding feature using SER.
My test setup:
192.168.25.101 - XLite ---> INVITE UA(sip:444)
192.168.25.163 - ser; rewrites to Address of record of Asterisk (sip:1111 instead of sip:444)
192.168.25.169 - PBX Asterisk
As I see from log files, SER sends INVITE 2 times,
then Asterisk sends 3 times 200 status etc . until BYE is sent (here everything correct....)
Here is my configuration script:
==================================
route{
. . .
record_route();
#--rewrite URI
if (method=="INVITE"){ #--call forwarding
setflag(1);
route(1); #--cfw
if(!isflagset(1)) {
break;
}
lookup("location")
...
};
route[1] {
log(1, "forwarding\n");
if (!if_activated("to")) {
#--continue regular call set up
break;
};
if (!lookup_dn()){
sl_send_reply("404","Not Found");
resetflag(1); #--continue!
break;
};
}
Here if_activated checks whether the feature was activated; lookup_dn() rewrites URI to new UA and call append_branch()
Any ideas what I am doing wrong?
if (!t_relay()) {
sl_reply_error();
};
}
===========
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Hi there,
I have the following:
SER:
version: ser 0.9.0 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE
262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c
compiled on 16:46:51 Mar 29 2005 with gcc 2.96
CISCO GATEWAY:
IOS (tm) 5300 Software (C5300-IS-M), Version 12.2(26), RELEASE SOFTWARE
(fc2)
The situation is the following:
If I have to UAs, both behind same (linux-iptables) nat, I have a
two-way audio communication, with nice quality, with no problems at all.
If I place a call to the PSTN, trough my Gateway, from my UA, I have a
two-way audio communication, with nice quality, with no problems at all.
If I try to receive a call from the PSTN, trough my Gateway, I only have
a one-way audio communication, but only in the SIP->PSTN direction. What
could this be?.
I can send my ser.cfg and some dumps from communications, but I believe
the problem is quite clear. In fact, I followed the instructions from
onsip.org.
- some ngrep's ...
U ip_ua:5070 -> ip_ser_mediaproxy:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP ip_ser_mediaproxy;branch=z9hG4
U ip_ser_mediaproxy:5060 -> snom_proxy:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP snom_proxy:5060;branch=z9
U snom_proxy:5060 -> ip_ser_mediaproxy:5060
ACK sip:ip_ser_mediaproxy:5060 SIP/2.0..v: SIP/2.0/UDP 38.117.2
U ip_ser_mediaproxy:5060 -> ip_ua:5070
ACK sip:1991015@ip_ua:5070 SIP/2.0..Record-Route: <
The ACK arrives nicely to my UA (ip_ua:5070), but it keeps sending RTP
to my GATEWAY's IP and port.
- tcpdumping RTP on the UA (private IP:192.168.1.185.15090)(behind NAT)
20:36:37.596851 192.168.1.185.15090 > IP_CISCO.20482: udp 32
- show call active voice brief at my cisco:
2D2B : 104269100hs.1 +273 pid:1991334 Originate 1991015 active dur
00:00:36 tx:162/3202 rx:1790/35800 IP ip_ser_mediaproxy:35234 rtt:0ms
pl:33170/140ms lost:0/0/0 delay:69/69/70ms g729r8
- session.py (from media proxy) (look at the ports, its fine and logic
according to the tcpdumping of RTP at my LAN).
Caller Via Called Status Duration
Codec Type Traffic
------------------------------------------------------------------------
---------------------------
IP_CISCO:20482 - ip_ser_mediaproxy:35234 - ?.?.?.?:? inactive 1'04"
G729 Audio 11.29k/0/0
I would appreciate some help ( or even IDEAS )
Best regards,
Lucas
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27/03/2005
Dear All,
We had found that the config files provided didn't work as expected, so
we now have tested the mediaproxy version and modified it to ensure that
is works. We have also updated the rtpproxy file to reflect these
changes.
So please download the new version 03a which we hope now works ! !
www.ONsip.org
Sorry for the inconvenience but this is why we are a community working
together.
Thanks
Paul, Simon, Greger
Hi All,
I have found that while using consume_credentials() in my ser config,
when I make a call to a one to many alias, only the first URI in the
alias will get sent a invite. Removing the consume_credentials() from my
config allows parallel forking to work again, but with the
Proxy-Authorization: header field still present.
Is this a known issue?
Thanks,
-Jev
Hey guys,
This might be the wrong list for this... but I'm trying to find a large
scale multi-user conferencing system which supports silence suppression.
Asterisk works well enough in terms of supporting a large number of
users being able to connect to the same "room" simultaneously, but there
are a lot of posts about it not supporting silence suppression... which
from empirical evidence appears to be true!
Does anyone know if SEMS properly supports silence suppression (i.e. in
both directions... if no-one in the conference is saying anything (as
unlikely as that might be.. then there's no data sent to UAs) ?
On a slightly different note, having not looked into the code yet, does
anyone have any ideas how easy it is to implement new codecs in SEMS?
Once again, if I've posted to the wrong list, sorry about that.... I'm
sure someone will redirect me..
Many thanks for all the work and useful help I've got from this list
over the last year or so.
Cheers!
Dave
Ozan,
Please reply to the list and not me directly. It would help if you provide
the exact configuration and sip log/message dump of you successful and
unsuccessful attempts. Then people can look and see if there is a problem in
SER or just your routing implementation.
Thanks
Alex
_____
From: Ozan Blotter [mailto:cosmocid@ispro.net.tr]
Sent: Wednesday, March 30, 2005 12:21 PM
To: avishnev(a)optonline.net
Subject: PSTN logic in SER
Estemeed Alex Vishnev,
Now there's no error while starting SER, i've removed "{", it routes call
but how. I use Zyxel Prestige P2002 ATA's as VoIP Devices.
It has some usual properties like (This is original):
SIP Number : 8334843600
SIP Local Port : 5060
SIP Server Address : 212.154.104.198
SIP Server Port : 5060
REGISTER Server Address : 212.154.32.154
REGISTER Server Port : 5060
SIP Service Domain : 212.154.32.154
If i do above configuration in my ATA's and for example i call back my
cellular as 905353490056, SER routes call to PSTN Gateway since its prefix
is 90. But this time when i call 8334454556 for example, there's no
activity, it doesn't ring.
I do not apply auth. on SER because all important information is kept on
other SIP Server (Sysmaster's VoiceMaster) like billing,customer information
etc. We're planning to give free SIP2SIP calls between our customers
beginning with prefix 833 or 834.
If i change all information above to SER (all 212.154.32.154 to
212.154.104.198) i can call other customers but this time when i try to call
my cellular again as example it does not work, i think SER doesn't pass
register details to other SIP Server which is acting as PSTN Guard.
How can i arrange this ?
Thanks again and sorry for taking your time.
Ozan Blotter
Hello,
I am trying to activate call forwarding using ser. When subscriber sends INVITE which starts from some prefix (say *22), I strip the prefix and write activation information to mysql.
I use ser 08-14 ran into the following problem:
=====ser.cfg:
if (uri =~ "^sip:\*22") {
log(1, "*22\n");
strip(3);
route(1);
...
route[1] {
if (!exec_msg('
echo "SIP_HF_MAX_FORWARDS ${SIP_HF_MAX_FORWARDS}">>/tmp/cfwdebug.log;
echo "SIP_HF_VIA ${SIP_HF_VIA}">>/tmp/cfwdebug.log;
echo "SIP_HF_CSEQ ${SIP_HF_CSEQ}">>/tmp/cfwdebug.log;
echo "SIP_HF_FROM ${SIP_HF_FROM}">>/tmp/cfwdebug.log;
echo "SIP_ORUI ${SIP_ORUI}">>/tmp/cfwdebug.log;
echo "SIP_HF_CONTENT_LENGTH ${SIP_HF_CONTENT_LENGTH}">>/tmp/cfwdebug.log;
echo "SIP_TID ${SIP_TID}">>/tmp/cfwdebug.log;
echo "SIP_DID ${SIP_DID}">>/tmp/cfwdebug.log;
echo "SIP_RURI ${SIP_RURI}">>/tmp/cfwdebug.log;
echo "SIP_HF_TO ${SIP_HF_TO}">>/tmp/cfwdebug.log;
echo "SIP_OUSER ${SIP_OUSER}">>/tmp/cfwdebug.log;
echo "SIP_HF_CALLID ${SIP_HF_CALLID}">>/tmp/cfwdebug.log;
echo "SIP_SRCIP ${SIP_SRCIP}">>/tmp/cfwdebug.log;
echo "SIP_HF_CONTENT_TYPE ${SIP_HF_CONTENT_TYPE}">>/tmp/cfwdebug.log;
echo "SIP_HF_CONTACT ${SIP_HF_CONTACT}">>/tmp/cfwdebug.log')) {
log(1, "script-echo fail\n");
}
}
now when I am looking to /tmp/cfwdebug.log, it contains:
SIP_HF_MAX_FORWARDS 69
SIP_HF_VIA SIP/2.0/UDP 192.168.25.101:5060;rport;branch=z9hG4bK5410DAEAA0744FA0B2BD64F59D465EF8
SIP_HF_CSEQ 1591 INVITE
SIP_HF_FROM matuka <sip:tuka@192.168.25.163>;tag=1660681736
SIP_ORUI sip:*226566548@192.168.25.163
SIP_HF_CONTENT_LENGTH 275
SIP_TID 3c8495de485acf4427dcfc7c5a0c0e8a
SIP_DID
SIP_RURI sip:6566548@192.168.25.163
SIP_HF_TO <sip:*226566548@192.168.25.163>
SIP_OUSER *226566548
SIP_HF_CALLID 50748A60-1D39-470B-B9CC-84CCC32B67BD(a)192.168.25.101
SIP_SRCIP 192.168.25.101
SIP_HF_CONTENT_TYPE application/sdp
SIP_HF_CONTACT <sip:tuka@192.168.25.101:5060>
Any help highly aprreciated!
Tina,
software engineer
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Can someone point me toward documentation about activating
the flatstore module?
Thanks,Steve
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
Hi -
Trying to get a handle on how to get presence working in SER/pa module.
- Does some kind soul have an ngrep output of a typical
subsribe-ok-notify-ok exchange with a UA and SER? Even better would be
some additional notifies with updates? I think mine is broken (no
'status' section in the notify) and not sure if/why......
- Is there supposed to be tables in mysql related to this presentities?
(assuming that mysql is fully engaged), I don't seem to have any nor can
I find their creation in a db init script.
Thanks in advance.
