Hi all
I'm an Italian Student
and I'm working on CPL module.
I've seen on the IPTEL site a WEB page with the CPL composer.
Where could I download CPL composer software for the creation of the CPL scripts?
Thanks a lot
Paolo
HI,
i am using redhat with kernel 2.4.18-3 and grub loader 0.91.
while using SIP server, to get a detailed error log i changed some options in serv.conf file as given by server user guide.
-------------------------------------
ser logs -- with default settings few logs are printed to syslog facility which typically dumps them to /var/log/messages. To enable detailed logs dumped to stderr, apply the following configuration options: debug=8, log_stderror=yes, fork=no.
---------------------------------------
but after i restarted my machine it is getting hanged at boot time as it tries to start ser: while booting up somewhere at the end of booting process. and stops there. probably waiting for some response and because of that stopped system from booting normally.
any idea why it is so .
regards,
Geetha
--
********************************************************************
eRiva provides IT Solutions & Services to companies worldwide.
Our State of the Art Research & Development Centers provides
Innovative Solutions to global customers.
********************************************************************
--
Hello all
We have set up an environment that includes one jabberclient and one sip (msn) client.
Now, the problem.
Registration is no problem, everything is working fine and we se the "online" on both sides.
The thing now is that our sip-client can send one or several mesages to the jabbberclient but as soon as the jabberclient
send a respond, the connection is broken.
When trying to send a msg from sip to jabber the sip-client just says:
"The following message could not be delivered to all recipients:"
Etherreal says nothing is happening, the client doesn't even try to send something out, not even to DNS or Ser.
jabber to sip is no problem, and sip to jabber is no problem as long as you haven't send a msg from jabber to sip. (puh)
Do you have any idea what is happening here ?
Does it have to do with call-id's or the contact-field's ????
version: ser 0.8.11pre8-mem (i386/linux)
Thank's in advance
Regards
Anna
__________________________________________________________________
anna rahm
ICQ#: 176699319
Current ICQ status:
+ More ways to contact me
__________________________________________________________________
The ser-0.8.10 sources contain a debian sub-directory for building
debian packages. However the control file in this directory lists
incomplete and improper build dependancies for the package.
Currently the dependancies are listed as:
Build-Depends: debhelper (>> 3.0.0), libmysqlclient-dev, libexpat1-dev,
fakeroot
While they should be:
Build-Depends: debhelper (>> 3.0.0), libmysqlclient-dev, libexpat1-dev,
zlib1g-dev
Fakeroot is not a necessary package for building the Debian ser package.
It simply makes it easier to build the package as a non-root user.
--
Jamin W. Collins
Folks,
I am currently playing with t_on_negative feature trying to implement
overflow routing (i.e. if original destination returns an error, then
request should be adjusted somehow and redirected to some, possibly
different, destination). I've started with the following config:
route {
[...]
rewriteFromRoute();
if (method == "INVITE") {
addRecordRoute();
};
if (method == "INVITE) {
t_on_negative("1");
};
t_relay_to("address1", "port1");
}
reply_route[1] {
rewritehostport("address2:port2");
append_branch();
}
But quickly found that after transaction was redirected to
address2:port2, ACKs, BYEs, 200K and CANCELS are still being forwarded
to address1:port1, despite containing valid Route fields pointing to
address2:port2. Then I've modified it as follows to let ser use
information from that field to route ACKs, 200OKs and BYEs:
route {
[...]
rewriteFromRoute();
if (method == "INVITE") {
addRecordRoute();
};
if (method == "INVITE" || method == "CANCEL") {
# INVITEs and CANCELs
if (method == "INVITE) {
t_on_negative("1");
};
t_relay_to("address1", "port1");
} else {
# ACKs, 200OKs, BYEs
t_relay();
};
}
reply_route[1] {
rewritehostport("address2:port2");
append_branch();
}
For the most of the time everything works like a charm - if
address1:port1 is unreachable or replies with an error the request is
being redirected to the second destination, BUT if the initiating UA
tries to cancel transaction when transaction is already redirected but
before receiving final "200 OK" from the second destination, the
CANCEL request is forwarded to address1:port1, not to address2:port2
as it should be. I've tried to modify setup as follows, thinking that
maybe in the case of CANCEL explicit specification of proxy address
confuses ser, but no avail - in this case ser forwards the CANCEL
request to its own address and eventually it dies with Too Many Hops.
route {
[...]
rewriteFromRoute();
if (method == "INVITE") {
addRecordRoute();
};
if (method == "INVITE") {
# INVITEs
t_on_negative("1");
t_relay_to("address1", "port1");
} else {
# CANCELs, ACKs, 200OKs, BYEs
t_relay();
};
}
reply_route[1] {
rewritehostport("address2:port2");
append_branch();
}
I think that such behaviour arises from the fact that ser after
branching a transaction doesn't keep an address this transaction was
forwarded to. In my opinion, it needs to be corrected, so that after
receiving CANCEL from the UA that initiated transaction ser probably
should CANCEL *all* branches of this transaction. To do this it needs
to be able to tell exactly where to send CANCELs.
What do you think?
-Maxim
---------- Original Message ----------------------------------
From: "Geetha Shree" <geethas(a)erivasystems.com>
Reply-To: <geethas(a)erivasystems.com>
Date: Tue, 15 Apr 2003 01:15:24 -0700
hi
Thanks for the answer.
But again sorry ,i am geeting 408 request time out again.
I have changed the via-stack as was in the request
like below
>>SUBSCRIBE sip:geetha@192.168.1.9 SIP/2.0
>>Via: SIP/2.0/UDP 192.168.1.9:5060
>>Via: sip/2.0/UDP 192.168.1.22:5060
>>From: <sip:client1@192.168.1.9>;tag=1050301248710
>>To: <sip:client2@192.168.1.9>
>>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22
>>CSeq: 1 SUBSCRIBE
>>Contact: <sip:192.168.1.22:5060>
>>Expires: 1800
>>Content-Length: 0
>>
>>
>>SIP/2.0 200 OK
>>Via: SIP/2.0/UDP 192.168.1.9:5060
>>Via: SIP/2.0/UDP 192.168.1.22:5060
>>From: <sip:client1@192.168.1.9>;tag=1050301248710
>>To: <sip:client2@192.168.1.9>;tag=1050301221190
>>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22
>>CSeq: 1 SUBSCRIBE
>>Contact: <sip:192.168.1.21:5060>
>>Expires: 1800
>>Content-Length: 0
>the reply does not fit the request, since it has a different via stack.
>It's thus not recognized as a part of the transaction by proxy, forwarded
>statelessly and terminated with 408.
>
>-Jiri
>
>At 09:03 AM 4/15/2003, Geetha Shree wrote:
>>hi all,
>>
>>We are getting 408 request timeout for our SUBSCRIBE method inspite of other Useragent sending
>>a 200OK response for the SUBSCRIBE method.
>>
>>Both SUBSCRIBE and 200OK packets are getting proxied to respective user agents correctly.
>>But the client1 is receiving 408 request time out error and the client2 is receiving SUBSCRIBE methods often.
>>
>>The SUBSCRIBE packet and the 200OK packet are as shown below.
>>
>>SUBSCRIBE sip:geetha@192.168.1.9 SIP/2.0
>>Via: sip/2.0/UDP 192.168.1.22:5060
>>From: <sip:client1@192.168.1.9>;tag=1050301248710
>>To: <sip:client2@192.168.1.9>
>>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22
>>CSeq: 1 SUBSCRIBE
>>Contact: <sip:192.168.1.22:5060>
>>Expires: 1800
>>Content-Length: 0
>>
>>
>>SIP/2.0 200 OK
>>Via: SIP/2.0/UDP 192.168.1.9:5060
>>Via: SIP/2.0/UDP 192.168.1.22:5060;received=192.168.1.22
>>From: <sip:client1@192.168.1.9>;tag=1050301248710
>>To: <sip:client2@192.168.1.9>;tag=1050301221190
>>Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22
>>CSeq: 1 SUBSCRIBE
>>Contact: <sip:192.168.1.21:5060>
>>Expires: 1800
>>Content-Length: 0
>>
>>
>>
>>
>>Thanks
>>Geetha
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
>--
>Jiri Kuthan http://iptel.org/~jiri/
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
--
********************************************************************
eRiva provides IT Solutions & Services to companies worldwide.
Our State of the Art Research & Development Centers provides
Innovative Solutions to global customers.
********************************************************************
--
hi all,
We are getting 408 request timeout for our SUBSCRIBE method inspite of other Useragent sending
a 200OK response for the SUBSCRIBE method.
Both SUBSCRIBE and 200OK packets are getting proxied to respective user agents correctly.
But the client1 is receiving 408 request time out error and the client2 is receiving SUBSCRIBE methods often.
The SUBSCRIBE packet and the 200OK packet are as shown below.
SUBSCRIBE sip:geetha@192.168.1.9 SIP/2.0
Via: sip/2.0/UDP 192.168.1.22:5060
From: <sip:client1@192.168.1.9>;tag=1050301248710
To: <sip:client2@192.168.1.9>
Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22
CSeq: 1 SUBSCRIBE
Contact: <sip:192.168.1.22:5060>
Expires: 1800
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060
Via: SIP/2.0/UDP 192.168.1.22:5060;received=192.168.1.22
From: <sip:client1@192.168.1.9>;tag=1050301248710
To: <sip:client2@192.168.1.9>;tag=1050301221190
Call-Id: 237f8aec0b47b309dfe8af3d844f9e95(a)192.168.1.22
CSeq: 1 SUBSCRIBE
Contact: <sip:192.168.1.21:5060>
Expires: 1800
Content-Length: 0
Thanks
Geetha
FYI
We found some SIP widely used implementations don't like loose-routing
parameter (";lr") without any value (which is the currently documented
use of loose-routing) and break. We will probably introduce a workaround
option which will allow to use lr with value (e.g., ";lr=true").
In particular, we learned that Windows Messenger, rejects loose-record-routed
requests and replies with "400 BAD Request" to INVITES with ";lr" in it.
Cisco IOS strips all RR parameters without value away, including ";lr" from
Route header fields in subsequent requests.
-Jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hi all,
I downloaded the latest code from CVS and trying to
include Presence Agent(pa.so) in my ser.cfg file.
While trying to start the SER, i am geting
"Segmentation fault" error. I have included dependent
modules (tm,usrloc,jabber) before loading pa.so inside
ser.cfg. Is there any special parameters to be passed
for pa.so module through modparam ?
Also is there a client test program available to test
Presence module ?
Thanks,
Santosh
__________________________________________________
Do you Yahoo!?
Yahoo! Tax Center - File online, calculators, forms, and more
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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
it turned out that cisco ios sip implementation does not include the new
route uri parameters (ftag, lr) that cvs ser adds in its record route
entry in the requests (ack, bye, ...) that cisco sends within a dialog.
what i heard is that it may take until the end of the year until these
bugs get fixed.
thus there is even more reasons to push cisco to get their acts together
or find a better sip/pstn gw from some other vendor (so far i haven't
heard of any).
-- juha
Howdy,
I've been playing around with this SIP scenario
tool. Maybe I'm just the last one to find out.
But after ngrep'ing one too many packets this
is really refreshing. It is a freeware application
and it's output is either html or text. I've uploaded
a couple of files to my website so you can see the
call flow diagram capabilities.
The source (perl) code can be had at:
http://www1.cs.columbia.edu/sip/implementations.html
under sip scenario.
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
It has been kinda quiet on these lists.
---greg
Hallo
First I want to thank you for your quick response the last time.
When trying to send message from jabber server to the SIP client i get the
following error from the SIP server, it looks like all the adresses are
correct and the messages sent from SIP is properly delivered.
1(24392) DEBUG: mk_proxy: doing DNS lookup...
1(24392) str2ip: WARNING: unexpected char s in [sip.storstark]
1(24392) str2ip6: WARNING: unexpected char s in [sip.storstark]
1(24392) get_record: lookup(_sip._udp.sip.storstark, 33) failed
1(24392) sip_resolvehost: not SRV record found for sip.storstark, trying
'normal' lookup...
1(24392) str2ip: WARNING: unexpected char s in [sip.storstark]
1(24392) str2ip6: WARNING: unexpected char s in [sip.storstark]
1(24392) ERROR: mk_proxy: could not resolve hostname: "sip.storstark"
1(24392) qm_free(0x80aad40, 0x80b7aec), called from proxy.c: mk_proxy(225)
1(24392) qm_free: freeing block alloc'ed from proxy.c: mk_proxy(208)
1(24392) ERROR: t_relay: bad host name in URI <sip:x@sip.storstark>
1(24392) ERROR: uri2sock: Can't create a dst proxy
1(24392) ERROR: t_uac_dlg: no socket found
Thanks in advance
Best regards
Magnus
_________________________________________________________________
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Hi,
How do I make the aliases field editable in serweb. The user should
have the option of editing his aliases.
Any help in this regard, greatly appreciated.
Regards,
Santosh M Hulkund
Hello,
The FCP module for SER seems to be approaching to a stable version (no
guarranties, though). The module is in charge of 2 functions:
- Translation of SIP messages from a private network to a public one
(using NAT).
- Open and close pinholes in a firewall to allow media sessions through.
The module implements the FCP side that communicates with a deamon called
fcpd that runs on the firewall itself (developed by iptel:
http://www.iptel.org/fcp/). It seems to be reasonably stable now with cvs
version of ser from 31/03/03.
I'm looking for anyone interested in trying it and provide any feedback
(bugs, new features or complains :)).
Greetings,
Jaime
At 09:34 PM 4/6/2003, Mike wrote:
>I apologize for the terminology, I'm still learning.
Don't worry -- actually there was some confusion in what I wrote you too.
ATA may do BYE/Also, but the standard way of call-transfer is using
REFER request.
>Maybe not an "issue" with SER, but I guess what I need is a way for SER to
>get this BYE message, see the Also: and initiate another INVITE to the
>correct station, handling it like a new call?
No. What you need is standardized call transfer support in your phone.
-Jiri
Hello
Does nethelper module exist as a patch, or it's now intergrated into
ther core of SER, can anyone clarify this?
I've just mentioned 'nethelper' module in documentation, but no source
code in tree.
--
Michael Vasilenko
I've successfully installed serweb and am able to log in. However, once I
do so, I get the following error:
Warning: fopen(/tmp/ser_fifo) [function.fopen]: failed to create stream:
Permission denied in /usr/local/apache/htdocs/serweb/functions.php on line
172
at the top of the screen, and the message "sorry -- cannot open fifo" in
red letters on the webpage above the user's information. I'm wondering if
ser or apache needs to be configured somewhere with the correct
permissions or ownership to make this work.
Thanks!
Meghan Byrne
Meghan Patricia Byrne
Georgia Institute of Technology, Atlanta Georgia, 30332
Email: gte705u(a)prism.gatech.edu
At 01:19 AM 4/4/2003, Mike wrote:
>I'm trying to get SER to work properly with the call forwarding feature of
>an ATA-186. The ATA-186 forwards calls (#90number#) by doing a BYE with
>an "also" header. SER doesn't seem to do anything with this at all, and
>the call just sits in limbo until everybody hangs up.
>
>Call forwarding on busy/no-answer works, but I think that's because the
>ATA is doing a refer instead.
>
>Any clue where I should be looking for this? Is SER supposed to support
>this BYE/also method, and I have something misconfigured, or is there
>additional work that needs to be done to support this?
I'm not sure what exactly happens from your narration. The terminology is
little a bit new to me -- forwarding means imho sending a 3xx reply to an
INVITE and not BYE requests. A message dump would help me to understand.
Anyway, I don't think that is an issue related to SER -- it is about
end-device features.
-Jiri
hi all,
We wanted to know whether the SER server will just Proxy the Notify method or else it gives the 200OK response also.
When we sent the Notify method, it just proxied to the contact ip address but did not send the 200OK response to us.
Thanks
Geetha
I have been testing the last changes about the "rport-received" tags, and I
have found the next bug:
when using UDP transport, if the IP address and port in the "Via" field
differ from the IP address and port of the incoming packet, the "received"
and "port" tags added to the Via of the response are correct, but the packet
is sent back to the "received" IP address BUT to an incorrect port ( this
should be the "rport" port ), that generates an ICMP "Destination
unrecheable port".
I have been capturing these packets and I think I have found exactly where
is the problem. For example in the next sequence:
1)REQUEST PACKET: ( REGISTER for example )
Source: 192.168.1.2:1667 Destination: 192.168.1.3:5060
"Via: SIP/2.0/UDP 192.168.1.100:6969;rport"
2)RESPONSE PACKET:
Source: 192.168.1.3:5060 Destination: 192.168.1.3:33542 <- HERE IS THE
PROBLEM, this should be "rport"
"Via: SIP/2.0/UDP 192.168.1.100:6969;rport=1667;received=192.168.1.2"<-but
this is correct
3)ICMP PACKET:
Source 192.168.1.2 Destination: 192.168.1.3 Type:3 "Destination unrecheable"
Code:3 "Port unrecheable"
I've been checking other pairs of ports sent and received like 1667-33542,
1507-58117, 1509-58629, 1511-59141 and the second is the network address
order of the first ( 1667 = ntohs(33542) ) .
I think that you forget to make a htons() of the port received in host
format before sending the packet back to network.
Please, let me know when this issue is solved in the CVS.
Best regards and congratulations for your excellent work.
Sergio.
fyi (in response to a frequently asked question "how to get snom's
buggy RR-ing working along with PRACK).
I just tried to create a workaournd to the problem (which is PRACK
does not put Routes learned in 183). It's as easy as mangling SIP
messages not to advertise their support for "100rel":
if (method=="INVITE" && search("User-Agent: snom")) {
#replace("PRACK, ", "");
replace("100rel, ", "");
};
-jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hello,
please send me the whole log.
Jan.
On 04-04 03:14, 2 2 wrote:
> Thanks Jan, for your quick response !! However I have
> uncommented the mysql.so and it is getting loaded
> before auth.so. Any other pointer ??
>
> Thanks,
> Santosh
>
> --- Jan Janak <jan(a)iptel.org> wrote:
> > You forgot to uncomment line containing loadmodule
> > ".../mysql.so"
> >
> > Jan.
> >
> > On 04-04 02:32, 2 2 wrote:
> > > Hi all,
> > >
> > > I am new to the SER and trying to install on my
> > linux
> > > box with persistent Data storage option. I have
> > > downloaded ser-0.8.10_linux_i386.tar and trying to
> > > configure it.
> > >
> > > I have installed MYSQL and now trying to start
> > SER. I
> > > am geting following error messages.
> > >
> > > Apr 4 14:50:59 incq066a /etc/init.d/ser[9866]:
> > > mod_init(): Unable to bind database module
> > > Apr 4 14:50:59 incq066a /etc/init.d/ser[9866]:
> > > init_modules(): Error while initializing module
> > auth
> > >
> > > I have followed all the steps from:
> > >
> > >
> >
> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/INSTALL…
> > >
> > > Where am I going wrong ? Any pointer will be
> > helpful.
> > > Thanks in advance,
> > > Santosh
> > >
> > >
> > >
> > > __________________________________________________
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> > forms, and more
> > > http://tax.yahoo.com
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
> > ATTACHMENT part 2 application/pgp-signature
>
>
>
> __________________________________________________
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> http://tax.yahoo.com
Hi all,
I am new to the SER and trying to install on my linux
box with persistent Data storage option. I have
downloaded ser-0.8.10_linux_i386.tar and trying to
configure it.
I have installed MYSQL and now trying to start SER. I
am geting following error messages.
Apr 4 14:50:59 incq066a /etc/init.d/ser[9866]:
mod_init(): Unable to bind database module
Apr 4 14:50:59 incq066a /etc/init.d/ser[9866]:
init_modules(): Error while initializing module auth
I have followed all the steps from:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/INSTALL…
Where am I going wrong ? Any pointer will be helpful.
Thanks in advance,
Santosh
__________________________________________________
Do you Yahoo!?
Yahoo! Tax Center - File online, calculators, forms, and more
http://tax.yahoo.com
I edited the /etc/ld.so.conf and added these two lines
/usr/lib
/usr/lib/mysql
Thanks for the reply, It has been a great help to us. Thanks once again.
But Still after this i am getting the following error
[root@afi init.d]# ser start
Listening on
127.0.0.1 [127.0.0.1]::5060
10.1.3.131 [10.1.3.131]::5060
Aliases: localhost:5060 localhost.localdomain:5060 afi:5060
[root@afi init.d]# 0(5164) connect_db(): Access denied for user: 'ser@localhost' (Using password: YES)
0(5164) db_init(): Error while trying to connect database
0(5164) mod_init(): Error while connecting database
0(5164) init_modules(): Error while initializing module usrloc
ERROR: error while initializing modules
The Password that has been assigned to ser user is ser
from the prompt if i try to connect to mysql with the following command i am able to connect
mysql -user -pser ser
do you have mysql client library installed ?
> >
> > Jan.
> >
> > On 01-04 17:54, vijay wrote:
> > > hi,
> > > got the following error while restarting the ser server after changing
> the
> > > ser.cfg file for loading the mysql module.
> > >
> > > [root@afi init.d]# ser start
> > > 0(4478) ERROR: load_module: could not open module
> > > <//usr/lib/ser/modules/mysql.so>: libmysqlclient.so.10: cannot open
> shared
> > > object file: No such file or directory
> > > 0(4478) parse error (22,13-44): failed to load module
> > > ERROR: bad config file (1 errors)
> > >
> > > what may be the cause. Can any body help me.
> > >
> > > thanx
> > > vijay
Hello
I'm trying to implement the following scheme
<User LAN, ATA186, PC, etc> --- < Linksys router with dynamic public IP > -- shdsl ---
--- ISP --- .... --- VoIP provider network ( SER, PSTN gateway, etc )
It will be nice, of anyone could give me an advice how to implement
the possibility of calls to and from VoIP devices INSIDE User network on
private IPs to the VoIP provider network on fixed static IPs.
Maybe I misunderstood something, but it's not clear to me, how the
incoming call to user will be possible in this scenario.
Linksys router = NAT + DHCP + 1-8 port hub
Thanks.
--
Michael Vasilenko
I'm trying to get SER to work properly with the call forwarding feature of
an ATA-186. The ATA-186 forwards calls (#90number#) by doing a BYE with
an "also" header. SER doesn't seem to do anything with this at all, and
the call just sits in limbo until everybody hangs up.
Call forwarding on busy/no-answer works, but I think that's because the
ATA is doing a refer instead.
Any clue where I should be looking for this? Is SER supposed to support
this BYE/also method, and I have something misconfigured, or is there
additional work that needs to be done to support this?
Thanks!
- Mike
Hi,
we are suffering the same problem. The serctl script was originally written
for Linux and is a little incompatible with other systems. Even if you
change the shell from sh to bash there are still some utilities like "tail"
which has a diffirent syntax than from Linux.
I am porting this script to solaris for our systems where no bash is
available. If this is also useful for you I can send you later.
Regards
Yang
On Apr 02, 2003 at 08:26, Steve Blair <blairs(a)isc.upenn.edu> wrote:
>
> Hello:
>
> I'm just getting started with my implementation of
> SER on FreeBSD 4.7-RELEASE. I've read the
> documentation, installed Apache and mySQL and
> would like to add users for my domain.
>
> I've tried adding an administrative user using serctl
> however this script fails for reason I cannot explain.
> Here is what I did:
>
>
> serctl add user1 password1 email1(a)mydomain.com
> read: Illegal option -s
>
> read: Illegal option -s
> Try changing the first line of the script form #!/bin/sh to #!/bin/bash
> (the read in sh does not support the -s option).
> BTW: this is fixed on CVS for the new version (but don't try the CVS
> code until next week, we're commiting a lot of changes right now).
>
> Andrei
hi all,
i have installed SER (ver 0.8.10) with MySql support.
i have a problem. in ser.cfg when i set the "fork" parameter to "no", SER
is not coming up at all. if the default "yes" is used, the server comes up
just fine.
pls help. i want to disable forking somehow.
rgds,
sunithi
since many people mentioned that they are using cisco 5x00 series
sip/pstn gateways, it would be nice if also others than me would push
cisco to implement digest authentication in its ios software. currently
5x00 namely is the only sip UA that i'm aware of that doesn't support
digest (or any other) authentication of sip requests.
another annoying thing with 5x00 is that i cannot configure the host
part of its from uri. it always uses as the host part the numeric ip
address of the interface via which it is sending out the sip request.
-- juha
Hi Tomas,
we have an AS 5200 instead of AS 5300. Can we use it as SIP2PSTN gateway too?
Which IOS are you using? Must the IOS contain the MCM?
Best Regards
Yang
hi,
I note that there is a example of click_to_dial in the ser/examples/web_im directory, but it seems not to be implemented fully. I want to know how to implement click_to_dial in ser,does that need RTP stack and other things?
thanks.
        Joe_chen
        chenhb(a)sict.ac.cn
          2003-04-03
It is possibly a .net only option. The Messenger help seems
to indicate that it should work for any communications service,
but the help/document files are pretty thin.
I guess I need to see if I can find another SIP capable IM
interface.
Dan
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Wednesday, April 02, 2003 12:32 PM
To: Dan Austin; serusers(a)lists.iptel.org
Subject: RE: [Serusers] Re: windows messenger
At 10:05 PM 4/2/2003, Dan Austin wrote:
>Calls to the PSTN work for me as well, but I've run into a small snag
>using the IM functions.
>
>Messenger claims I can added multiple parties to the IM session, but
>the menu option is not available. Am I running into a Messenger issue,
>or a SER limitation?
Maybe it is just a .net option? It seems unlikely to me SER is guilty.
-jiri
Hi,
Installed ser (0.8.10 src) and mysql db. Tested the
register using some diffrent UAC. one of UAC got the
problem. When the UAC sent the register msg to ser,
ser generated more than 1M msg in log file (using log
level 8). A few seconds later, UAC got "513 msg too
big". I used the defualt max_len and the msg is kind
samll. see the attached register msg.
Thanks,
Alan
-----------------------------------------------
REGISTER sip:191.191.29.58:25060 SIP/2.0
Via: SIP/2.0/UDP 191.191.29.62:5060
From:
3001<sip:3001@iptel.org:5060;user=phone>;tag=27b74-12ec
To: 3001<sip:3001@iptel.org:5060;user=phone>
Call-ID:
a8cce363000000000000000000000000(a)191.191.29.62
CSeq: 101 REGISTER
Max-Forwards: 70
Expires: 900
Contact: <sip:3001@191.191.29.62:5060;user=phone>
Supported: timer
Content-Length: 0
=====
__________________________________________________
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