Hi,
I got Pingtel, SipTone hardphones and Windows Messenger softphone to
work just fine with the SER 0.8.9 server on the Internet via a Intertex
IX66 firewall.
I tweaked the SIP phone configurations to use the SER 0.8.10 I installed
at home, from the binary RPM obtained from
ftp://ftp.berlios.de/pub/ser/0.8.10/packages/redhat/7.x. I ran into
some very strange behaviors with SER and the firewall. I disconnected
the firewall connection to the Internet and the SIP INVITE transactions
stopped involving the firewall. This was evident from analyzing
Ethereal captures. (It was if the firewall was in a promiscuous mode of
operation an acted upon packets that did not apply to it.)
However, I still could not get INVITE transactions to generate 2xx
responses for two local phone and SER proxy configured without
authentication. So, I tried compiling the source code on my Redhat 7.3 box.
> gcc -v
Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs
gcc version 2.96 20000731 (Red Hat Linux 7.3 2.96-110)
> make config <- for SER
.
.
.
Old gcc detected (2.9x), use gcc >= 3.1 for better results
make: *** No rule to make target `config'. Stop.
My question is should I compile SER with the GCC 2.96 compiler or pay
attention to the make config message and update GCC? (Some of the RPM
binaries differ from what I compiled with the default RH 7.3 GCC 2.96
compiler.)
Sincerely,
Scott Holben (skaht(a)iptel.org)
Hi all,
I just installed MySql server on my linux box and
trying to start SER. I am geting an error message
while staring SER:
"connect_db(): Cann't connect to local MySQL server
through socket '/var/lib/mysql/mysql.sock' (2)"
My MySql serevr is runing and I can connect to SER
database from my machine. However the socket is
created at /tmp/mysql.sock. Now my question is, from
where did SER pickup the socket path while starting ?
How can I change it to /tmp/mysql.sock ?
Any pointer is appreciated.
Thanks and Regards,
Santosh
__________________________________________________
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We're looking to bring on full-time programmers/engineers to work on SIP-based VoIP platform. Useful skill sets are: -Experience with Linux/Solaris-Experience programming with C/C++ -Experience with MySQL/PostgreSQL/RADIUS-Experience Administering Linux/Unix/Apache-Experience with PHP or similar scripting languages-Experience testing and deploying Soft/Hard UAs-Experience with wholesale termination/origination networks using Cisco Voice Gateways-And of course, interest in and experience using SIP. Some H.323 experience may be useful as well Location of work is US West Coast and permission to work in the US is unfortunately a must. Please reply if you're interested to siptelco(a)yahoo.com with a resume/experience/skills.
---------------------------------
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I've discovered a bug in SER 0.8.10 which can segfault the server with a
dereference of NULL, one time in RAND_MAX.
In the function branch_builder() in msg_translator.c, if both the parameter
'label' and the parameter 'char_v' are 0, SER will crash. This is because
the code assumes that if label is 0, char_v is non-NULL, and so will attempt
to call memcpy() with char_v as the source.
When branch_builder is invoked by the tm module, however, the label
parameter comes from a random value assigned by h_table.c. This value is
generated by rand(). As such, its value can legitimately be 0, which will
happen, on average, one time in RAND_MAX.
On Linux, RAND_MAX is 2^31, so this crash is very unlikely. However, on
Solaris (where I'm doing some testing), RAND_MAX is 2^15, so this crash is
reasonably common for a server under heavy load. However, this is a "valid"
crash in either case; this isn't just a portability issue.
(Note that RAND_MAX == 2^15 being less than TABLE_ENTRIES == 2^16 can also
cause problems, according to a comment in h_table.c, though I believe only
ones of efficiency, not correctness.)
The patch below works around the problem in the simplest possible way,
though it isn't a correct fix. I suspect the proper solution would be a) to
reverse the logic of branch_builder() to test char_v for NULL, rather than
label for non-0; and b) to check with the preprocessor if RAND_MAX is less
than TABLE_ENTRIES, and if so, use random() rather than rand() in
modules/tm/h_table.c.
--
Jonathan Lennox
lennox(a)cs.columbia.edu
--- ser-0.8.10.orig/msg_translator.c Mon Oct 21 15:21:50 2002
+++ ser-0.8.10/msg_translator.c Wed Apr 9 15:31:47 2003
@@ -813,6 +813,12 @@
begin++; size--;
} else return 0;
+ if (!label && !char_v) {
+ LOG(L_ERR, "ERROR: branch_builder: both label and char_v "
+ "are 0\n");
+ return 0;
+ }
+
/* label is set -- use it ... */
if (label) {
if (int2reverse_hex( &begin, &size, label )==-1)
Hello!
I'm interested to implement the possibility of call between
two NATted ATAs using SER 0.8.10 and Maxim's nethelper module.
I'm using config described at
http://lists.iptel.org/pipermail/serusers/2003-January/000165.html
Signalling works just fine, but no media stream.
My test setup:
ATA1 --- NAT1 --- SER --- NAT2 --- ATA2
cut from tcpdump of call from ATA1 to ATA2 on private side of NAT1:
213.186.192.26 is NAT2
23:52:05.707635 213.186.192.26 > 172.20.0.205: icmp: 213.186.192.26 udp
port 16384 unreachable [tos 0xc0]
23:52:05.727178 172.20.0.205.10000 > 213.186.192.26.16384: udp 32 [tos
0xa0]
23:52:05.727677 213.186.192.26 > 172.20.0.205: icmp: 213.186.192.26 udp
port 16384 unreachable [tos 0xc0]
23:52:05.747216 172.20.0.205.10000 > 213.186.192.26.16384: udp 32 [tos
0xa0]
23:52:05.747734 213.186.192.26 > 172.20.0.205: icmp: 213.186.192.26 udp
port 16384 unreachable [tos 0xc0]
ignore
(we are just introducing automated spam filters as the "click-to-kill-spam"
feature is currently killing our free time)
--
Jiri Kuthan http://iptel.org/~jiri/
hi,Jiri
In my last click to dial letter, i said that it seems not to be implemented fully. this example does not work. The reason is use of REFER for third-party call-control has not been standardized . I thank after the call session is
established , then it will need establish RTP channel to transport voice datas from web UA(caller) to another UA(callee),
so i said it need RTP stack to establish RTP channel.
you replyed: "It is in there and you don't need RTP." what's meaning? In files of click_to_dial and ser, i can't find any codes about establishing RTP channel ,you can tell me where or why don't need RTP?
Hi,
I am now trying to compile the stable release ser-0.8.10 on a solaris mashine where I have installed gcc 3.2.2 and gmake 3.77. I have also installed bison and flex on that mashine.
While compiling I get the following errors:
bash-2.03# make all
Makefile.rules:81: lex.yy.d: No such file or directory
Makefile.rules:81: cfg.tab.d: No such file or directory
yacc -d -b cfg cfg.y
cfg.y:154.17: warning: stray `,' treated as white space
cfg.y:155.22: warning: stray `,' treated as white space
cfg.y:155.31: warning: stray `,' treated as white space
cfg.y:155.36: warning: stray `,' treated as white space
cfg.y:155.44: warning: stray `,' treated as white space
cfg.y:156.20: warning: stray `,' treated as white space
cfg.y:156.26: warning: stray `,' treated as white space
cfg.y:586.9: syntax error, unexpected "|"
make: *** [cfg.tab.c] Error 1
What is missing? I can use this compiler well for compiling mysql on the same mashine.
Thanks,
Yang
> -----Original Message-----
> From: Greg Fausak [mailto:greg@august.net]
> Sent: Thursday, April 10, 2003 6:26 PM
> To: serusers(a)lists.iptel.org
> Cc: sip(a)august.net
> Subject: [Serusers] SIP Scenario Tool
...
> The example (real world debug)
> callflow can be viewed
> at my development website: http://stage.august.net/sip1_index.html
> and
> http://stage.august.net/sip1.html
>
> You use tcpdump (or ethereal, or whatever) to grab the
> Output, like on linux:
> tcpdump -s 0 -i eth0 'port 5060' -w /var/log/sip1.dump
Hello!
How do you capture from different networks? Is it possible to input
several dump-files into the tool, one captured at the caller site and
one captured at the callee site?
regards,
Klaus