Maxim,
Thank you very much for your report.
I'm not the code owner who has the last word, but I agree there is
a memory leak. It seems to me that insert_new_lump family is ok as
long as the calling functions care to free their lump buffers on
failure -- that's where the leak lives.
Which imho happens at few places on CVS:
+insert_new_lump_before
- save_ruri in rr/loose.c
- textops/append_hf_helper.c
- (on contrary, it is ok in maxfwd/add_maxfwd_header, rr functions
calling insert_RR, and msg_translator.c)
+insert_new_lump_after
- search_append_f in textops.c
- replace_f in textops.c
- (ok in build_req_buf...)
The memory leak is unlikely to occur, as it is triggered by lack of
private memory which (unlike shmem) hardly gets exhausted -- so
operation is fortunately not affected. It will be fixed in the next
release.
-Jiri
ps -- I swear on cscope :)
>Folks,
>
>I've noticed that there are multiple potential memory leaks in SER.
>The problem is that if a insert_new_lump*() function returns a NULL
>for some reason (currently the only condition is memory allocation
>error), it doesn't free the memory buffer passed to it and most of
>the code doesn't care to deallocate that buffer after NULL is returned.
>It could be easily fixed and probably needs to before the next version
>is released.
>
>Thanks!
>
>-Maxim
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Goal: To learn about ser in a "non-critical" and very small PBX-like
environment so as to be able to understand nuances of the system in a
production environment at a later date at various firms whose owners
who have expressed to me a high degree of interest in SIP call
routing for larger enterprise and CLEC implementations.
Sub-Goal: to make all calls into and out of my house routed via IP to
alternate destinations based on ser routing configuration. I have
subscribed to a long-distance plan via "iconnecthere.com", I have a
PRI gateway configured at a remote location (for calls into two area
codes only) and my plan is to have a Cisco 2610 with FXO card for
terminating my "local" phone line. My house phones are all Cisco
ATA-186 devices. Based on called number, my calls will be sent to
the iconnecthere.com SIP service, the Cisco PRI gateway, or the Cisco
2610 single-line gateway. Calls will also be routed appropriately
based on number called on an inbound basis from any of those three
gateway systems (PRI, 2610, or iconnecthere.com)
Progress: I have phone-to-phone calling working well, and I have
phone-to-PRI gateway calling working well in both directions. I have
not yet received the 2610, so I do not have the single-line analog
gateway running, but I don't expect any issues with that system, as I
understand the Cisco implementation of inbound/outbound VOIP sessions.
Problem: My "iconnecthere.com" account is a username/password
protected account which (of course) requires a UA at my side of the
connection. To forward calls from the various phones in the house, I
would need to have something re-write my username/password requests
on the fly when they are sent out to the iconnecthere.com SIP
proxies/gateways. My first assumption is that I'll need a (sigh)
B2BUA to act as a gateway, running (for convenience) on the same UNIX
platform as ser. After thinking about it for a while, I am
uncertain if that is required, but at this point I can't determine
what I need to do.
I would appreciate hints on:
a) Wether I need a B2BUA at all, and if not, what config options
should I be looking at in ser?
b) If I do need a B2BUA, what would you recommend? I'm using
OpenBSD as my platform, and (personally) I'd like to stay away from
Java for the moment.
Continuing discussion:
I could see this as a fairly useful toolkit trick for a small
business who wants to replace their phone switch with SER or ser-like
systems. If you've got an office with 10 people, it may make
economic sense to simply get "generic" accounts with a SIP long
distance gateway provider like iconnecthere.com (there are others)
and allocate each of those accounts to individuals in the
organization. This is not a solution for a large-scale operation for
the various reasons outlined in the
http://www.iptel.org/info/trends/#b2bua texts, but the number of
small-scale shops out there is very large and a simply understood
package (and simply billed) is what would be desired for many places
who still use under ~10 analog lines for outbound dialing from their
PBX. Any outbound calls to certain prefixes would always be pushed
through a specific account for LD calling.
Additionally, inbound calling through a similar service would have
to also come in via the same mechanism, with a REGISTER request being
sent by the B2BUA and all subsequent calls being routed through the
SER proxy.
PS: I'd appreciate any open-source hints on how to get an ATA-186
(v2.15) running behind NAT with ser on the "outside" of the NAT,
without statically configuring the "NAT" address on the ATA-186 every
time the outside address changes. Lots of keyword matches found on
Google, but very few clues to be scraped from the resulting documents
as to "how" do it from the server side.
JT
Greetings to all and happy new year
I'm in the process of installing ser 0.8.10 on Linux
(intel) and when I run Too much shared memory the
"ser"executable I receive the following error message
"Too much shared memory demanded: 33554432"
When I turn the debug option on I receive the
following:
0(4019) WARNING: hash function optimized for 1024
entries
0(4019) WARNING: use of 65536 entries may lead to
unflat distribution
0(4019) ERROR: shm_mem_init: could not attach shared
memory segment: Invalid argument
0(4019) could not initialize shared memory pool,
exiting...
Too much shared memory demanded: 33554432
Any ideas?
Thanks
__________________________________________________
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Dear Sir/Madam,
We are developing a sip stack.
We wanted to know whether you can provide any sip server for testing our stack.
We also wanted to know the procedures invloved in it.
Thanks in advance
Geetha
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Dear Sirs,
I am having some strange problems when trying to use b2bua for
accounting and call duration limiting with SER proxy server. The idea
is simple: since the SER can route SIP messages depending on their
source address, we can force incoming SIP messages to be passed to
B2BUA for accounting purposes first, and after the same request
re-enters the proxy from B2BUA pass it to the final destination. Call
flow looks like the following:
-----
|UA2|
-----
^
|
|4
|
----- 3 -------
----- 1 | |<-----------| |
|UA1|---->|SER| 2 |B2BUA|
----- | |----------->| |
----- -------
For some reason, it doesn't work in such configuration. The problem is
that B2BUA's UAC keeps resending `200 OK' replies ingnoring ACKs it
receives from UA2 until timeout hits, after which it considers the
call dead, despite the fact that both UA1 and UA2 think that the call
is established. Maybe it has something to do with the fact that it
sends to and receives messages from the same host (SER), but I don't
think that this should be a problem, since those two call legs have
different call id's, so that b2bua should be able to distinguish
between them easily. Attached please find tcpdump logs of one such
session, here 192.168.1.1 is UA1 (originating), 192.168.0.9 is UA2,
192.168.1.100 is the host running both SER and B2BUA (the former uses
port 5060, while the latter - 5065). There are two files:
ser-b2bua.log is the log of udp exchange between SER and B2BUA and
ser-ua1ua2.log - log of exchange between SER and UA1/UA2.
Any ideas are appreciated.
Thanks!
-Maxim
On Mon, Jan 20, 2003 at 03:22:33PM +0100, Jiri Kuthan wrote:
> At 09:53 PM 1/13/2003, you wrote:
> >Attached please find the patch, which implements t_on_positive()
> >approach. Now my config looks like the following and everything now
> >works as expected:
>
> thanks for the patch -- I will try to integrate it as I move ahead
> with other changes to TM. Does your nathelper module work now?
Attached please find more fresh patch with several bugs smashed. It
is probably the last one on this topic, because it works reasonably
well and we now have moved to adjusting b2bua for our needs (mostly
radius auth/acc). In addition to nathelper module, this patch
implements rport support as specified in the correstponding IETF
draft.
Thanks!
-Maxim
Folks,
I've noticed that there are multiple potential memory leaks in SER.
The problem is that if a insert_new_lump*() function returns a NULL
for some reason (currently the only condition is memory allocation
error), it doesn't free the memory buffer passed to it and most of
the code doesn't care to deallocate that buffer after NULL is returned.
It could be easily fixed and probably needs to before the next version
is released.
Thanks!
-Maxim
I'm thinking that the 'forceful termination' should have options for
redirection, i.e. to an announcements (or other media) server, etc. Is this
in line with your thoughts?
-Benson
> -----Original Message-----
> From: Maxim Sobolev [mailto:sobomax@portaone.com]
> Sent: Thursday, January 16, 2003 3:37 PM
> To: serusers(a)lists.iptel.org; kapitan(a)portaone.com
> Subject: [Serusers] [RFC] ideas about new dialog module
>
>
> Hi,
>
> I am thinking about writing a new module for SER, which will track SIP
> dialogs and will serve as an abstraction layer for other modules, much
> like the tm module now. We need such module for 2 reasons:
>
> 1. Call accounting. Our billing engine is based on the assumption that
> a node provides accounting information for completed calls, not for
> individual transactions. It is easier for us to extend proxy with
> similar features than to modify billing engine to do transaction
> matching.
>
> 2. Debit card application. Currently, there is no way to use SER for
> debit card applications, where it is necessary to set the maximum
> duration of the call and terminate it forcefully if that duration is
> exceeded.
>
> The raw idea is as follows:
>
> - the module will register callbacks with tm.register_tmcb(), probably
> TMCB_REQUEST_IN and TMCB_REPLY_IN ones and will match INVITEs to BYEs
> keeping information about the state of ongoing sessions in the shared
> memory.
>
> - the module will provide interested modules with ability to register
> several callbacks, i.e. on dialog creation/teardown and yet another
> callback on dialog timeouts (more about that below).
>
> - the module will provide utility functions for forceful termination
> of any ongoing dialog.
>
> - when invoking dialog creation callback function, the module will
> give the function opportunity to install a timer on that dialog, so
> that if the dialog is still active after timer expires, then some
> action is performed. For example, in the debit card applications, in
> such case the accounting module can decide to forcefully terminate a
> dialog.
>
> What do you think?
>
> -Maxim
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Hi,
I am thinking about writing a new module for SER, which will track SIP
dialogs and will serve as an abstraction layer for other modules, much
like the tm module now. We need such module for 2 reasons:
1. Call accounting. Our billing engine is based on the assumption that
a node provides accounting information for completed calls, not for
individual transactions. It is easier for us to extend proxy with
similar features than to modify billing engine to do transaction
matching.
2. Debit card application. Currently, there is no way to use SER for
debit card applications, where it is necessary to set the maximum
duration of the call and terminate it forcefully if that duration is
exceeded.
The raw idea is as follows:
- the module will register callbacks with tm.register_tmcb(), probably
TMCB_REQUEST_IN and TMCB_REPLY_IN ones and will match INVITEs to BYEs
keeping information about the state of ongoing sessions in the shared
memory.
- the module will provide interested modules with ability to register
several callbacks, i.e. on dialog creation/teardown and yet another
callback on dialog timeouts (more about that below).
- the module will provide utility functions for forceful termination
of any ongoing dialog.
- when invoking dialog creation callback function, the module will
give the function opportunity to install a timer on that dialog, so
that if the dialog is still active after timer expires, then some
action is performed. For example, in the debit card applications, in
such case the accounting module can decide to forcefully terminate a
dialog.
What do you think?
-Maxim
hi,
I have a problem with radius_auth, i use raidus_auth module as www-authentication ,but something is wrong, the free-radius server need the PW_PASSWORD attribute for authentication,
while the ser's radius_auth module doesn't send the PW_PASSWORD attribute, i can't find the password from the REGISTER message, there is a NONCE like
nonce="3e1947180000000040c0e6240c7f8145a64eb654ce9a14e0" and encrypted with MD5.
Do i need get the user's password and how to get user's the password? please help.
thanks.
Dear Sirs,
I am having some strange problems when trying to use b2bua for
accounting and call duration limiting with SER proxy server. The idea
is simple: since the SER can route SIP messages depending on their
source address, we can force incoming SIP messages to be passed to
B2BUA for accounting purposes first, and after the same request
re-enters the proxy from B2BUA pass it to the final destination. Call
flow looks like the following:
-----
|UA2|
-----
^
|
|4
|
----- 3 -------
----- 1 | |<-----------| |
|UA1|---->|SER| 2 |B2BUA|
----- | |----------->| |
----- -------
For some reason, it doesn't work in such configuration. The problem is
that B2BUA's UAC keeps resending `200 OK' replies ingnoring ACKs it
receives from UA2 until timeout hits, after which it considers the
call dead, despite the fact that both UA1 and UA2 think that the call
is established. Maybe it has something to do with the fact that it
sends to and receives messages from the same host (SER), but I don't
think that this should be a problem, since those two call legs have
different call id's, so that b2bua should be able to distinguish
between them easily. Attached please find tcpdump logs of one such
session, here 192.168.1.1 is UA1 (originating), 192.168.0.9 is UA2,
192.168.1.100 is the host running both SER and B2BUA (the former uses
port 5060, while the latter - 5065). There are two files:
ser-b2bua.log is the log of udp exchange between SER and B2BUA and
ser-ua1ua2.log - log of exchange between SER and UA1/UA2.
Any ideas are appreciated.
Thanks!
-Maxim
Hello,
your messages about CISCO ATA reminds me that I have also one (a preproduction
unit originally made by Komodo), so I upgraded firmware to 2.15 and tried it
behind a symmetric NAT. It works, I had to enable portforwarding for signalling
and media and setup properly outer address of the NAT box in the phone.
Once you finish your NAThelper module, I would like to give it a try and see
if it works without modifications of the NAT box. So I can help you to test the
code if you want.
regards, Jan.
Folks,
While playing with SER I found that I can trigger repeatable crash when
doing REGISTER multiple times. Quick glance at the code in question
revealed that indeed, when constructing reply to REGISTER message,
SER uses fixed-lengh buffer to put all non-expired contacts for that
user and doesn't bother to check for overflow. The bug could be easily
exploited by a complete stranger on servers that don't perform
authentification of REGISTER requests, and by an user with a valid
credintals on server that do authentification. Mounting attack leads
to denial of service.
Attached please find fake REGISTER message, which if sent to open
server kills it (nc -u my.sip.server 5060 < register.killser),
and patch to fix the problem.
-Maxim
just in case someone would like to show up there. first come, first served.
-jiri
>Hello!
>
>Internet Telephony Conference & Expo is right around the corner! As a speaker and participant in the conference program, TMC would like to offer you (2) conference passes free of charge. These passes can be given to colleagues, clients, or guests that you would like to invite to hear you speak and to the show. These passes would be full conference passes and would allow that person to go to the entire event, not just to your session.
>
>If you would like to take advantage of this offer all you need to do is e-mail the following
>information for each person:
>
>Name
>Title
>Company name
>Address
>Phone
>Fax
>E-mail
>
>Put "Internet Telephony Miami Speaker Conference Program" in the subject line and you are done! The pass will be sent to them in the mail.
>
>As a speaker, you will already have your speaker pass prepared. Please pick up your pass at the event registration.
>
>PLEASE NOTE - this program is not to replace an existing conference pass. No monies that might have been paid will be refunded.
Yes, that was my problem! Thank you very mutch!
Are you involved in the SHIP (http://voip.sh.cvut.cz/about.shtml)
project? (because of your emailaddress). I read that you are using Vocal
as voice mail server. Do you have an installation guide how to setup
such a system with vocal and ser?
regards,
Klaus
> -----Original Message-----
> From: Jan Janak [mailto:J.Janak@sh.cvut.cz]
> Sent: Friday, January 17, 2003 1:14 PM
> To: Klaus Darilion
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Password problem with ser and mysql
>
>
> Hello,
>
> auth is not the only module that uses database. If you have
> enabled support for database in usrloc (by modparam("usrloc",
> "db_mode", 1) or modparam("usrloc", "db_mode", 2) then you
> have to change usrloc's password as well. Try to use the
> following: modparam("usrloc", "db_url",
> "sql://ser:klaus@localhost/ser")
>
> regards, Jan.cd
>
> On 17-01 11:56, Klaus Darilion wrote:
> > Hello Jan!
> >
> > Comments inline.
> > > default username and password for auth module is
> > > serro:47serro11, if you are able to login as ser:heslo, then
> > > username and password can be changed.
> >
> > > If you change your password in mysql, you must change it for
> > > ser@localhost For example: grant ALL on ser.* to
> > > ser@localhost identified by 'klaus';
> >
> > I changed it for localhost, but it won't work.
> >
> > > If you use something like grant ALL on ser.* to ser
> > > identified by 'klaus'; then the server will be unable to
> > > login to the database.
> > >
> > > Try also the following:
> > > mysql -h localhost -u ser -p ser
> >
> > I changed the password of ser@localhost to 'klaus'. I can
> connect to
> > mysql with mysql -h localhost -u ser -p ser
> > and password 'klaus'. I changed the password in ser.cfg to 'klaus'
> > modparam("auth", "db_url", "sql://ser:klaus@localhost/ser")
> > , but still the ser server can't start:
> > : connect_db(): Access denied for user: 'ser@localhost' (Using
> > password: YES)
> >
> >
> > I still think there must be a bug somewhere, so that ser tries to
> > connect with the 'heslo' password.
> >
> > regards,
> > Klaus
> >
> > > This will prompt for password, if you are unable to login
> > > with your password, ser will be unable to login as well
> > > (probably because you changed password for ser and not for
> > > ser@localhost)
> > >
> > > regards, Jan.
> > >
> > > On 16-01 18:34, Klaus Darilion wrote:
> > > > Hello!
> > > >
> > > > I have sucessfully installed the ser server (0.8.10 from
> > > rpms) with an
> > > > mysql database. When I use the standard password for the user
> > > > "ser"
> > > > everything works fine. But if I change the password from
> > > "heslo" to a
> > > > new one, for example "klaus" instead of "heslo" (of
> course in the
> > > > mysql-database und in the config file) the ser server
> can not start
> > > > up.
> > > >
> > > > Following are some different configurations and my suggestions
> > > > what
> > > > could be the problem.
> > > >
> > > > ser.cfg: modparam("auth", "db_url",
> > > > "sql://ser:heslo@localhost/ser")
> > > > mySQL-root-password: XXXXX
> > > > mySQL-ser-password: heslo
> > > > ---> works fine
> > > >
> > > > ser.cfg: modparam("auth", "db_url",
> > > > "sql://ser:klaus@localhost/ser")
> > > > mySQL-root-password: XXXXX
> > > > mySQL-ser-password: klaus
> > > > ---> doesn't work: connect_db(): Access denied for user:
> > > > ---> 'ser@localhost'
> > > > (Using password: YES)
> > > > should work, so I tried another user
> > > >
> > > > ser.cfg: modparam("auth", "db_url",
> > > "sql://root:XXXXX@localhost/ser")
> > > > mySQL-root-password: XXXXX
> > > > mySQL-ser-password: heslo
> > > > ---> works fine
> > > >
> > > > ser.cfg: modparam("auth", "db_url",
> > > "sql://root:XXXXX@localhost/ser")
> > > > mySQL-root-password: XXXXX
> > > > mySQL-ser-password: klaus
> > > > ---> doesn't work: connect_db(): Access denied for user:
> > > > ---> 'ser@localhost'
> > > > (Using password: YES)
> > > > very strange, because I told ser to connect as root. Is ser
> > > using the
> > > > default user/password instead of the configured one?
> > > >
> > > > ser.cfg: modparam("auth", "db_url",
> > > "sql://root:YYYYY@localhost/ser")
> > > > mySQL-root-password: XXXXX
> > > > mySQL-ser-password: heslo
> > > > ---> doesn't work: connect_db(): Access denied for user:
> > > > 'root@localhost' (Using password: YES)
> > > > of course it doesn't work, wrong password. So ser cares
> about the
> > > > settings in ser.cfg
> > > >
> > > > So my suggestion is that ser connects several times to the
> > > > database
> > > > whereas one time it uses the configured user/password and
> > > another time
> > > > it uses the default user/password - maybe a bug in the auth
> > > > module?
> > > >
> > > > Or does somebody of you changed the password successfuly?
> > > >
> > > > It would be nice if you can help me.
> > > >
> > > > Thanks,
> > > > Klaus
> > > >
> > > > My system is:
> > > > Linux version 2.4.18-14 (bhcompile(a)astest.test.redhat.com)
> > > (gcc version
> > > > 3.2 20020903 (Red Hat Linux 8.0 3.2-7)) #1 Wed Sep 4
> > > 12:13:11 EDT 2002
> > > > MySQL 3.23.52
> > > > ser-0.8.10-2.i386.rpm
> > > > ser-mysql-0.8.10-2.i386.rpm
> > > > _______________________________________________
> > > > 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
> >
>
At 01:59 PM 1/17/2003, Klaus Darilion wrote:
>Yes, that was my problem! Thank you very mutch!
>
>Are you involved in the SHIP (http://voip.sh.cvut.cz/about.shtml)
>project? (because of your emailaddress).
Jan is.
>I read that you are using Vocal
>as voice mail server. Do you have an installation guide how to setup
>such a system with vocal and ser?
I tried to deploy the vocal voicemail a while ago and it did not
meet me expectations. We are working on our own now.
-Jiri
Hello Jan!
Comments inline.
> default username and password for auth module is
> serro:47serro11, if you are able to login as ser:heslo, then
> username and password can be changed.
> If you change your password in mysql, you must change it for
> ser@localhost For example: grant ALL on ser.* to
> ser@localhost identified by 'klaus';
I changed it for localhost, but it won't work.
> If you use something like grant ALL on ser.* to ser
> identified by 'klaus'; then the server will be unable to
> login to the database.
>
> Try also the following:
> mysql -h localhost -u ser -p ser
I changed the password of ser@localhost to 'klaus'. I can connect to
mysql with
mysql -h localhost -u ser -p ser
and password 'klaus'. I changed the password in ser.cfg to 'klaus'
modparam("auth", "db_url", "sql://ser:klaus@localhost/ser")
, but still the ser server can't start:
: connect_db(): Access denied for user: 'ser@localhost' (Using
password: YES)
I still think there must be a bug somewhere, so that ser tries to
connect with the 'heslo' password.
regards,
Klaus
> This will prompt for password, if you are unable to login
> with your password, ser will be unable to login as well
> (probably because you changed password for ser and not for
> ser@localhost)
>
> regards, Jan.
>
> On 16-01 18:34, Klaus Darilion wrote:
> > Hello!
> >
> > I have sucessfully installed the ser server (0.8.10 from
> rpms) with an
> > mysql database. When I use the standard password for the user "ser"
> > everything works fine. But if I change the password from
> "heslo" to a
> > new one, for example "klaus" instead of "heslo" (of course in the
> > mysql-database und in the config file) the ser server can not start
> > up.
> >
> > Following are some different configurations and my suggestions what
> > could be the problem.
> >
> > ser.cfg: modparam("auth", "db_url", "sql://ser:heslo@localhost/ser")
> > mySQL-root-password: XXXXX
> > mySQL-ser-password: heslo
> > ---> works fine
> >
> > ser.cfg: modparam("auth", "db_url", "sql://ser:klaus@localhost/ser")
> > mySQL-root-password: XXXXX
> > mySQL-ser-password: klaus
> > ---> doesn't work: connect_db(): Access denied for user:
> > ---> 'ser@localhost'
> > (Using password: YES)
> > should work, so I tried another user
> >
> > ser.cfg: modparam("auth", "db_url",
> "sql://root:XXXXX@localhost/ser")
> > mySQL-root-password: XXXXX
> > mySQL-ser-password: heslo
> > ---> works fine
> >
> > ser.cfg: modparam("auth", "db_url",
> "sql://root:XXXXX@localhost/ser")
> > mySQL-root-password: XXXXX
> > mySQL-ser-password: klaus
> > ---> doesn't work: connect_db(): Access denied for user:
> > ---> 'ser@localhost'
> > (Using password: YES)
> > very strange, because I told ser to connect as root. Is ser
> using the
> > default user/password instead of the configured one?
> >
> > ser.cfg: modparam("auth", "db_url",
> "sql://root:YYYYY@localhost/ser")
> > mySQL-root-password: XXXXX
> > mySQL-ser-password: heslo
> > ---> doesn't work: connect_db(): Access denied for user:
> > 'root@localhost' (Using password: YES)
> > of course it doesn't work, wrong password. So ser cares about the
> > settings in ser.cfg
> >
> > So my suggestion is that ser connects several times to the database
> > whereas one time it uses the configured user/password and
> another time
> > it uses the default user/password - maybe a bug in the auth module?
> >
> > Or does somebody of you changed the password successfuly?
> >
> > It would be nice if you can help me.
> >
> > Thanks,
> > Klaus
> >
> > My system is:
> > Linux version 2.4.18-14 (bhcompile(a)astest.test.redhat.com)
> (gcc version
> > 3.2 20020903 (Red Hat Linux 8.0 3.2-7)) #1 Wed Sep 4
> 12:13:11 EDT 2002
> > MySQL 3.23.52
> > ser-0.8.10-2.i386.rpm
> > ser-mysql-0.8.10-2.i386.rpm
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
Hello!
I have sucessfully installed the ser server (0.8.10 from rpms) with an
mysql database. When I use the standard password for the user "ser"
everything works fine. But if I change the password from "heslo" to a
new one, for example "klaus" instead of "heslo" (of course in the
mysql-database und in the config file) the ser server can not start up.
Following are some different configurations and my suggestions what
could be the problem.
ser.cfg: modparam("auth", "db_url", "sql://ser:heslo@localhost/ser")
mySQL-root-password: XXXXX
mySQL-ser-password: heslo
---> works fine
ser.cfg: modparam("auth", "db_url", "sql://ser:klaus@localhost/ser")
mySQL-root-password: XXXXX
mySQL-ser-password: klaus
---> doesn't work: connect_db(): Access denied for user: 'ser@localhost'
(Using password: YES)
should work, so I tried another user
ser.cfg: modparam("auth", "db_url", "sql://root:XXXXX@localhost/ser")
mySQL-root-password: XXXXX
mySQL-ser-password: heslo
---> works fine
ser.cfg: modparam("auth", "db_url", "sql://root:XXXXX@localhost/ser")
mySQL-root-password: XXXXX
mySQL-ser-password: klaus
---> doesn't work: connect_db(): Access denied for user: 'ser@localhost'
(Using password: YES)
very strange, because I told ser to connect as root. Is ser using the
default user/password instead of the configured one?
ser.cfg: modparam("auth", "db_url", "sql://root:YYYYY@localhost/ser")
mySQL-root-password: XXXXX
mySQL-ser-password: heslo
---> doesn't work: connect_db(): Access denied for user:
'root@localhost' (Using password: YES)
of course it doesn't work, wrong password. So ser cares about the
settings in ser.cfg
So my suggestion is that ser connects several times to the database
whereas one time it uses the configured user/password and another time
it uses the default user/password - maybe a bug in the auth module?
Or does somebody of you changed the password successfuly?
It would be nice if you can help me.
Thanks,
Klaus
My system is:
Linux version 2.4.18-14 (bhcompile(a)astest.test.redhat.com) (gcc version
3.2 20020903 (Red Hat Linux 8.0 3.2-7)) #1 Wed Sep 4 12:13:11 EDT 2002
MySQL 3.23.52
ser-0.8.10-2.i386.rpm
ser-mysql-0.8.10-2.i386.rpm
Hi,
I have installed ser and was amazed how easy it was to install (we have
previously experimented with sipd, which finally runs ...).
I have run through the archives and also looked at all the
documentation, but some things still don't work.
I use kphone 2.11 as client, but have tried sipc on a solaris box to the
same extend. I am running kphone and ser 0.8.10 on redhat 8. MySQL is
3.23.52.
I have set the domain/realm things to cs.stir.ac.uk and restarted ser.
I have then set up a user using "serctl add srm test srm(a)cs.stir.ac.uk".
Now, on serweb in user_interface/index I cannot login with this user.
In kphone, I cannot register (it works ok on the columbia server):
----kphone says------------------------------------------
SipTransaction: Retransmit 1 (4000)
SipClient: Sending: 15:09:37.098
--------------------------------
REGISTER sip:cs.stir.ac.uk SIP/2.0
Via: SIP/2.0/UDP 139.153.254.196:5062
CSeq: 488 REGISTER
To: "Stephan" <sip:srm@cs.stir.ac.uk>
Expires: 900
From: "Stephan" <sip:srm@cs.stir.ac.uk>
Call-ID: 1902284369(a)139.153.254.196
Content-Length: 0
User-Agent: KPhone/2.11
Event: registration
Allow-Events: presence
Contact: "root"
<sip:root@139.153.254.196:5062;transport=udp>;methods="INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
SipClient: Receiving message...
SipClient: Received: 15:09:39.724
---------------------------------
SIP/2.0 478 Unresolveable destination (478/TM)
Via: SIP/2.0/UDP 139.153.254.196:5062
CSeq: 488 REGISTER
To: "Stephan"
<sip:srm@cs.stir.ac.uk>;tag=0a96755a1d32d0360b3e454593e5079b-9b73
From: "Stephan" <sip:srm@cs.stir.ac.uk>
Call-ID: 1902284369(a)139.153.254.196
Server: Sip EXpress router (0.8.10 (i386/linux))
Content-Length: 0
Warning: 392 139.153.254.196:5060 "Noisy feedback tells: pid=5138
req_src_ip=139.153.254.196 in_uri=sip:cs.stir.ac.uk
out_uri=sip:cs.stir.ac.uk via_cnt==1"
--- end kphone -----------------------------------------------------
ser does answer, the Replied locally counter in serctl moni reflects
this ok.
what is wrong?
do you need any other files?
In case you need more detail, where exactly is the server log??
by the way, I have tried this with auth turned on and off in ser.cfg.
and without persistence. always the same ...
thanks
S
--
Dr Stephan Reiff-Marganiec
Research Fellow
Department of Computing Science; University of Stirling
email: srm(a)cs.stir.ac.uk tel: 01786 46 7448
--
The University of Stirling is a university established in Scotland by
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person), you may not disclose, copy or deliver this message to anyone
and any action taken or omitted to be taken in reliance on it, is
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message and kindly notify the sender by reply email. Please advise
immediately if you or your employer do not consent to Internet email
for messages of this kind. Opinions, conclusions and other
information in this message that do not relate to the official
business of the University of Stirling shall be understood as neither
given nor endorsed by it.
hi all,
i've downloaded the ser package from you and it is
working fine with msn.
but as u said i tried for authentication... it is not
working properly...
i've done -
a.uncommented the lines required for
authentication...
b.changed iptel.org to localhost in www_* (2
places)
c.created mysql tables
d.did'nt made any changes in mysql tables
e.tried to authenticate using admin/heslo and
signinname as admin@<myipaddress>
f.restarted SER :( again asking for username/pass
pl. give ur comments
thanks in advance..
best regards
sunil
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I have a Linux box at home acting as a masquerading/NAT gateway for a few
Windows PCs, and have installed SER on there in order to use MS Messenger to
talk to people outside.
SER appears to be working in that I can get Messenger up on two PCs, connect
to SER and set up a voice connection between the two PCs. However, I cannot
connect to people offsite.
Relevant IPChains entries are
target prot opt source destination ports
ACCEPT udp ----l- anywhere anywhere any ->
5060
ACCEPT udp ------ anywhere anywhere any ->
7070:7080
I have made no changes to the default SIP configuration; it is working as
installed by the rpm package ser-0.8.10-1.i386.rpm. A browse through the
mailing list archive and through the admin guide doesn't show anything
obvious. No errors are reported to /etc/messages or /etc/syslog and serctl
moni does not show anything that looks relevant.
Does anyone have any suggestions?
--
Dr. Craig Graham, Software Engineer
Advanced Analysis and Integration Limited, UK. http://www.aail.co.uk/
Folks,
My nathelper module will need to tweak body of the messages, therefore
it will be third module after im and sms that needs body-related
functions. IMO it would probably make a lot of sense to move those
functions into core parser, so that modules will not have to import im
module just to use one or two utility functions.
Please let me know what do you think.
Thanks!
-Maxim
We have a group of people who run their own SIP servers (some are
SER, some are not) and we are going to be sharing a PSTN gateway (Cisco
box).
I need to rewrite the identities for the users of our gateway into
something that can be presented to the PSTN gateway for billing
purposes. We don't want to do that mapping on the gateway itself, for
various reasons.
How do people handle this situation? Is it wise to put all functionality
for user handling and incoming call handling and forwarding to another
place in the same SER server?
--Michael
Hi there,
In a REGISTER case there is a HeaderField AUTHORIZATION. The usual
encryption algorithm is "MD5".
Is it possible to REGISTER without any encryption?
Please give a statement if I understood everything right or correct me:
Digest Username => not encrypted;
realm =>not encrypted;
URI =>not encrypted;
Nonce => encrypted => is Password?
Response => encrypted => what´s that?
Thanks!
P.S.: I sniffered the network traffic when I registerd,established a
VoIP-Connection and ended my call. Etherreal, the snifferprogramm, does
understand SIP and SDP so I could easy protocol and understand what was
going on.
Joe_chen wrote:
>hi,bogdan
> Merry christmas!
> Thank you for giving me the jcpl server last time. i have run the jcpl server with ser , but i want to extend or improve some functions,so i need the source code of jcpl server. Can you give me the jcpl server's source codes? if you can ,please send it to my mailbox.
>
First, sorry for the delay, but I was on vacation! Merry Christmas a A
Happy New Year to you also!
It's possible for us to send you the sources, but, as I already told
you, this java version of the intrpreter will not be maintain or develop
any more. We started a C version that will be ready maximum in 4 month -
an interpreter as a modul of SER.
If you still want the java sources ( under GPL), let me know, and I will
send them!
Regards,
Bogdan
PS: for the future, please CC all your emails to "serusers(a)lists.iptel.org"!
>Thanks!
> Regards,Joe_chen.
>
>
>
>
hi,
i have compliled and loaded the sms module, then i linked my mobile phone (motorola v998) with my computer on com1 port. when i run ser in debug mode ,the output informations are:
..
0(1515) DEBUG:modem_process: openning modem
0(1515) INFO:initmodem: init modem Motorola on /dev/ttyS0.
0(1515) INFO:initmodem: Checking if Modem is registered to the network
0(1515) NOTICE:initmodem: Waiting 2 sec. before retrying
0(1515) NOTICE:initmodem: Waiting 2 sec. before retrying
0(1515) NOTICE:initmodem: Waiting 2 sec. before retrying
0(1515) NOTICE:initmodem: Waiting 2 sec. before retrying
...
the trouble is just like my modem device(motorola v998) doesn't work with my computer. do i need install the modem device drivers ? but i can't find the drivers, and my ser server is running on linux OS,if there are drivers for linux ?
the second problem is how i can send sms message to ser, is there another UA supported sending sms?
thanks!
reguards,Joe_chen
Hello,
I'm trying to set up radius accounting and has already compiled radius_acc.so, but I'm not shure how to configure it on ser.cfg.
What parameters can I use? Is there any document or config example with radius accounting?
Thanks and regards,
Claudio
Hello
I am using SER in a test environment as a proxy server.
I would like it to add additional fields in the SDP of an INVITE message
for some
test needs. (e.g. add 'a=sendonly').
Is it possible ?
If not possible, do you think it would be difficult to add a function
addSdp(<string>) that would append a given string to the SDP of the
current message and alter the 'content_length' header field ?
I do not see in the 'sip_msg' structure any pointer to the body part of the
message ; only pointers to header fields.
Thanks for you help
--
David Rio
Alcatel CIT - Rennes
ASD France
33 2 99 87 47 18
Using multiple domains in one server is pretty hard.
The schema even fights with me. I can't have both "graff" in the
"isc.org" domain and "graff" in the "flame.org" domain, for instance,
at least with the same user name. I think the unique field there
should be a <user, domain> tuple. Additionally, all tables should
probably grow a domain column, and lookup() would take that as an
argument (defaulting to the host-portion of the URI, probably)
That also makes things easier on routing rules, and on aliasing, and
on user interfaces, at least IMHO.
Are there any plans to make this sort of thing much easier? If not, I
may decide to hack on it a bit and see what I come up with.
Of course, I'd also like to replace the scripting language with
embedded Perl, but... :)
--Michael
Hello,
I wanted to configure ser so that only registered user are allowed to make calls. I have set up digest authentication but still unregistered users can make calls. I wonder if I'm missing something on my config file (as attached below).
Could someone please tell me what am I doing wrong, or provide me some configuration example on solving this issue?
Thanks and regards,
Claudio
###########
ser.conf
###########
route{
# Do strict routing if pre-loaded route headers present
rewriteFromRoute();
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("<my_ip>", "subscriber")) {
www_challenge("<my_ip>", "0");
break;
};
if (method=="INVITE" & !check_from()) {
sl_send_reply("403","Forbidden");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now
if (!t_relay()) {
sl_reply_error();
};
}
##########
Hi,
I downloaded SER yesterday and I am trying to
run it as a UAS.
I am using the sample configuration uas.cfg.
I modified it so that on receiving an INVITE
I send a 200 OK. Now the problem is when I
use the t_reply command to send the 200 OK,
it doesn't add any SDP part to it.
Is there anyway an SDP can be added to the
200 OK response to the INVITE.
Appreciate any help
Viji
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Is anybody aware of any SIP ALG developed or being developed for linux/freebsd?
Jaime
PS: If nobody is looking at the FCP client side for SER, I would like to
contribute in my spare time. Please let me know where should I start and may be
I could give it a try.
Jiri Kuthan <jiri(a)iptel.org> on 08/01/2003 21:39:20
To: Jaime GILL/EN/HTLUK@HTLUK
serusers(a)lists.iptel.org
cc:
Subject: Re: [Serusers] FCP support in SER
This piece of work is stalling for some reasons.
First, the gentlemen who was supposed to integrate an FCP
client silently stepped away from this assignment. FCP
server on linux is working.
The other problem with FCP is there are no real standards.
I was expecting the Midcom WG in the IETF to come up with one.
Instead, it spent two years with doing things whose use
I very strongly doubt.
If there is any volunteer on this mailing list who would
wish to complete or create the FCP client module, we will be
glad to provide guidance.
Users willing to traverse NATs may consider STUN/UPnP or ALGs
as an alternative solution. Users willing to traverse firewalls
may need to use an ALG or VPN technology. Unfortunately, NAT/FWs
break too many things and there is no one-size-fits-it-all
solution addressing all scenarios.
-Jiri
At 06:14 PM 1/8/2003, jaime.gill(a)orange.co.uk wrote:
>Hi,
>
>I am wondering if this is the right place to ask, but here it goes.
>
>Is the SER software going to incorporate the client side of FCP (Firewall
>Control Protcol) as a module at some point?
>
>Regards,
>
>Jaime
>
>
>
>*******************************************************************************
>Important.
>Confidentiality: This communication is intended for the above-named person and
>may be confidential and/or legally privileged. Any opinions expressed in this
>communication are not necessarily those of the company. If it has come to you
>in error you must take no action based on it, nor must you copy or show it to
>anyone; please delete/destroy and inform the sender immediately.
>
>Monitoring/Viruses
>Orange may monitor all incoming and outgoing emails in line with current
>legislation. Although we have taken steps to ensure that this email and
>attachments are free from any virus, we advise that in keeping with good
>computing practice the recipient should ensure they are actually virus free.
>
>Orange PCS Limited is a subsidiary of Orange SA and is registered in England No
>2178917, with its address at St James Court, Great Park Road, Almondsbury Park,
>Bradley Stoke, Bristol BS32 4QJ.
>*******************************************************************************
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
_______________________________________________
Serusers mailing list
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http://lists.iptel.org/mailman/listinfo/serusers
*******************************************************************************
Important.
Confidentiality: This communication is intended for the above-named person and
may be confidential and/or legally privileged. Any opinions expressed in this
communication are not necessarily those of the company. If it has come to you
in error you must take no action based on it, nor must you copy or show it to
anyone; please delete/destroy and inform the sender immediately.
Monitoring/Viruses
Orange may monitor all incoming and outgoing emails in line with current
legislation. Although we have taken steps to ensure that this email and
attachments are free from any virus, we advise that in keeping with good
computing practice the recipient should ensure they are actually virus free.
Orange PCS Limited is a subsidiary of Orange SA and is registered in England No
2178917, with its address at St James Court, Great Park Road, Almondsbury Park,
Bradley Stoke, Bristol BS32 4QJ.
*******************************************************************************
Hello,
I'm trying to compile the radius_acc modude but have run into the following error:
######
make[1]: Entering directory `/home/cthorell/ser-0.8.10/modules/radius_acc'
gcc -fPIC -DPIC -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.10"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 2.95"' -D__CPU_i386 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DDBG_QM_MALLOC -DFAST_LOCK -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -c acc.c -o acc.o
acc.c:38: radiusclient.h: File not found
make[1]: *** [acc.o] Error 1
make[1]: Leaving directory `/home/cthorell/ser-0.8.10/modules/radius_acc'
######
I can't find radiusclient.h either on ser-0.8.10 or freeradius-0.8.1 directories. Where can I find this file?
Thanks,
Claudio
I got SER up and running with a whopping total of three "domains"
being served on the same server. It was mostly painless, if missing some
features I'd like to have seen (more on that in later messages.)
What open source products are people using for voice mail, voice menu
prompts (press 1 for sales, press 2 for the executative restroom) and
for conference calls (which include PSTN calls as well as IP phones)?
--Michael
Hi,
Are there any plans to add SIP Symmetric Response support
(http://www.iptel.org/info/players/ietf/firewall/nat/draft-ietf-sip-symmetri…)
into SER, so that it would be possible to use compliant UAs behind NATs
with SER being on a public IP? The idea is simple: when the SIP packet
is received from UA, the server checks if an empty `rport' parameter
is present in the first Via field, and if yes then compares IP:PORT in
the first Via header with the actual IP:PORT this request came from,
and adds `received' and `rport' parameters into the header if either
doesn't match before forwarding request further.
Then, when replying or forwarding the request, the server checks if
any of received or rport is present and uses it instead of the
UA-provided values in corresponding Via header.
Please let me know if there are work in progress on this, so that
I will not have to reinvent the wheel.
Thanks!
-Maxim
Hi,
I am wondering if this is the right place to ask, but here it goes.
Is the SER software going to incorporate the client side of FCP (Firewall
Control Protcol) as a module at some point?
Regards,
Jaime
*******************************************************************************
Important.
Confidentiality: This communication is intended for the above-named person and
may be confidential and/or legally privileged. Any opinions expressed in this
communication are not necessarily those of the company. If it has come to you
in error you must take no action based on it, nor must you copy or show it to
anyone; please delete/destroy and inform the sender immediately.
Monitoring/Viruses
Orange may monitor all incoming and outgoing emails in line with current
legislation. Although we have taken steps to ensure that this email and
attachments are free from any virus, we advise that in keeping with good
computing practice the recipient should ensure they are actually virus free.
Orange PCS Limited is a subsidiary of Orange SA and is registered in England No
2178917, with its address at St James Court, Great Park Road, Almondsbury Park,
Bradley Stoke, Bristol BS32 4QJ.
*******************************************************************************
hi,
i have met the same problem with acc. i can get the invite message for accounting,but i can't get the bye message, i also capture the udp packages on ser server, i use MSN for UA, i find the sip messeges like: invite,100 trying,180 ringing ,200 ok(for invite),but when i hangup, there are not bye message, it's like sending bye message from UA to UA directly,while not going throug ser server,so there is not BYE message for accounting, is it true?
how to get the bye message for accounting? please help.
my ser.cfg is like:
..
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
#loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/print.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
..
modparam("acc", "db_url","sql://ser:heslo@localhost/ser")
modparam("acc","report_ack",1)
modparam("acc","log_level",1)
modparam("acc", "acc_flag", 1 )
modparam("acc", "missed_flag", 3 )
# -- tm params --
modparam("tm", "fr_timer", 10 )
modparam("tm", "fr_inv_timer", 20 )
modparam("tm", "wt_timer", 10 )
..
route{
# initial sanity checks -- messages with
# max_forwars==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};
# Do strict routing if pre-loaded route headers present
rewriteFromRoute();
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("192.168.139.125", "subscriber")) {
# www_challenge("192.168.139.125", "0");
# break;
# };
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# set for accounting (the same value as in acc_flag!)
if (method=="INVITE" || method=="BYE"){
setflag(1);
# ... and also report on missed calls ... note that reporting
# on missed calls is mutually exclusive with silent C timer
#
setflag(3);
};
# forward to current uri now
if (!t_relay()) {
sl_reply_error();
};
}
thanks!
Dear Sirs,
Our company is currently evaluating several freely available SIP
proxy/registrars to extend our VoIP billing engine, which currently
only supports H323, to support SIP as well. We found SER to be one of
the best for our needs: speed, documentation, flexibility - everything
is great. Hovewer, since our billing engine uses Radius for AAA, we need
to get Radius modules working. Unfortunately, we can't compile radius
modules with virgin radiusclient 0.3.2 as suggested in documentation - it
seems that some SIP-specific changes need to be applied to radiusclient
headers, see the following log:
gcc -fPIC -DPIC -pipe -O -mpreferred-stack-boundary=2 -march=pentium2 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -DNAME='"ser"' -DVERSION='"0.8.10"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DDBG_QM_MALLOC -DFAST_LOCK -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_SOCKADDR_SA_LEN -I/usr/local/include -c acc.c -o acc.o
acc.c: In function `radius_log_reply':
acc.c:125: `PW_SIP_METHOD' undeclared (first use in this function)
acc.c:125: (Each undeclared identifier is reported only once
acc.c:125: for each function it appears in.)
acc.c:136: `PW_SIP_RESPONSE_CODE' undeclared (first use in this function)
acc.c:156: `PW_SIP_FROM_TAG' undeclared (first use in this function)
acc.c:189: `PW_SIP_TO_TAG' undeclared (first use in this function)
acc.c:213: `PW_SIP_CSEQ' undeclared (first use in this function)
acc.c:288: `PW_SIP_TRANSLATED_REQ_URI' undeclared (first use in this function)
acc.c: In function `radius_log_ack':
acc.c:434: `PW_SIP_METHOD' undeclared (first use in this function)
acc.c:442: `PW_SIP_RESPONSE_CODE' undeclared (first use in this function)
acc.c:461: `PW_SIP_FROM_TAG' undeclared (first use in this function)
acc.c:491: `PW_SIP_TO_TAG' undeclared (first use in this function)
acc.c:514: `PW_SIP_CSEQ' undeclared (first use in this function)
acc.c:588: `PW_SIP_TRANSLATED_REQ_URI' undeclared (first use in this function)
acc.c: In function `rad_acc_request':
acc.c:733: `PW_SIP_METHOD' undeclared (first use in this function)
acc.c:744: `PW_SIP_RESPONSE_CODE' undeclared (first use in this function)
acc.c:764: `PW_SIP_FROM_TAG' undeclared (first use in this function)
acc.c:797: `PW_SIP_TO_TAG' undeclared (first use in this function)
acc.c:821: `PW_SIP_CSEQ' undeclared (first use in this function)
acc.c:904: `PW_SIP_TRANSLATED_REQ_URI' undeclared (first use in this function)
Would you be so kind to provide us with your changes for radiusclient or
point out where that modified version can be downloaded?
Thank you in advance!
On a separate note I would like to ask you to let me know if there is
any progress on pre-paid module discussed on this list recently. Since we
will also need this functionality, perhaps we can team-up to implement it.
Sincerely,
Maxim Sobolev
PortaOne Ltd
I really hate answering my own question :-)
> } else if(method=="BYE") {
> setflag(1);
> forward(uri:host, uri:port);
I changed this to t_relay() and then I get the BYE
ACC output in the syslog file. This explains something else
(I think). Sometimes the BYE would work, sometimes it would
not make it all the way to the other device (either the PSTN gateway or the
UA). That is probably because forward() simply bounces the
packet via UDP and the UDP can fail ??? t_relay() retries??
Does this make sense?
---greg
>
>Hi,
>
>I am trying to create a basic script for routing calls to and from
>the PSTN. I have experimented with many scripts over the last month,
>this included script is an example. I can use a UA on my
>laptop called 'estara' and REGISTER with my ser sip server.
>I can make a phone call (INVITE) to the PSTN. I can hang up the call
>from either end (BYE) and everything seems to work well.
>
>I am trying to get accounting working. I see the call acceptance
>record in my syslog file:
>
>Jan 7 00:21:09 build ser[3507]: ACC: transaction answered: method=INVITE, i-uri=sip:92143357976@build.august.net, o-uri=sip:2143357976@64.90.42.16:5060, call_id= 847b3d22-131d-46b0-abcb-dc8e6a9f32b5(a)192.168.100.100, from= greg <sip:1107@build.august.net>;tag=3a7b9c52, code=200
>
>However, I can't get the 'BYE' event to record the transaction to the
>syslog file. The 'BYE' event is bouncing through my SIP server.
>Can someone give me a hint?
>
>Thank you,
>---greg
>Greg Fausak
>
>Included file----example.cfg----------------------------------------
>debug=1 # debug level (cmd line: -dddddddddd)
>fork=yes
>log_stderror=no # (cmd line: -E)
>check_via=no # (cmd. line: -v)
>dns=no # (cmd. line: -r)
>rev_dns=no # (cmd. line: -R)
>port=5060
>children=4
>fifo="/tmp/ser_fifo"
>
>loadmodule "/usr/local/lib/ser/modules/sl.so"
>loadmodule "/usr/local/lib/ser/modules/tm.so"
>loadmodule "/usr/local/lib/ser/modules/acc.so"
>loadmodule "/usr/local/lib/ser/modules/rr.so"
>loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
>loadmodule "/usr/local/lib/ser/modules/usrloc.so"
>loadmodule "/usr/local/lib/ser/modules/registrar.so"
>modparam("acc", "log_level", 1)
>modparam("acc", "acc_flag", 1 )
>
>route{
> # filter too old messages
> if (!mf_process_maxfwd_header("10")) {
> log("LOG: Too many hops\n");
> sl_send_reply("483","Too Many Hops");
> break;
> };
> if (len_gt( max_len )) {
> sl_send_reply("513", "Wow -- Message too large");
> break;
> };
>
> /* ********* RR ********************************** */
>
> /* Do strict routing if route headers present */
> rewriteFromRoute();
> /* record-route INVITEs -- all subsequent requests must visit us */
> if (method=="INVITE") {
> addRecordRoute();
> } else if (method=="REGISTER") {
> log("here is a register");
> save("location");
> break;
> };
>
> # now check if it really is a PSTN destination which should be handled
> # by our gateway; if not, and the request is an invitation, drop it --
> # we cannot terminate it in PSTN; relay non-INVITE requests -- it may
> # be for example BYEs sent by gateway to call originator
> if (!uri=~"sip:9[2-9][0-9]{9}@.*") {
> if (method=="INVITE") {
> sl_send_reply("403", "Call cannot be served here");
> } else if(method=="BYE") {
> setflag(1);
> forward(uri:host, uri:port);
> } else {
> forward(uri:host, uri:port);
> };
> break;
> };
>
> # account completed transactions via syslog
> setflag(1);
>
> strip(1);
>
>
> rewritehostport("64.90.42.16:5060");
>
> # forward the request now
> if (!t_relay()) {
> sl_reply_error();
> break;
> };
>}
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
hi,
i have a problem about qm_malloc.
my ser has been running correctly before sevral days. but on the day before yesterday it doesn't work suddenly, in the debug mode, the output informations are:
..
0(1114) find_export: found <~db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so]
0(1114) find_export: found <~db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so]
0(1114) find_export: found <~db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so]
0(1114) qm_malloc(0x80adf80, 24) called from dbase.c: db_init(266)
0(1114) qm_malloc(0x80adf80, 24) returns address 0x80b43d0 on 0 -th hit
0(1114) qm_malloc(0x80adf80, 30) called from dbase.c: connect_db(69)
0(1114) qm_malloc(0x80adf80, 32) returns address 0x80b4418 on 0 -th hit
0(1114) qm_malloc(0x80adf80, 544) called from dbase.c: connect_db(89)
0(1114) qm_malloc(0x80adf80, 544) returns address 0x80b4468 on 0 -th hit
0(1114) qm_free(0x80adf80, 0x80b4418), called from dbase.c: connect_db(106)
0(1114) qm_free: freeing block alloc'ed from dbase.c: connect_db(69)
0(1114) mod_init(): Database connection opened successfuly
Maxfwd module- initializing
rr - initializing
0(1114) TM - initializing...
0(1114) DEBUG: register_fifo_cmd: new command (t_uac) registered
0(1114) DEBUG: register_fifo_cmd: new command (t_uac_from) registered
0(1114) DEBUG: register_fifo_cmd: new command (t_hash) registered
0(1114) qm_malloc(0x4212f000, 1572864) called from h_table.c: init_hash_table(276)
0(1114) qm_malloc(0x4212f000, 1572864) returns address 0x421354b8 on 0 -th hit
0(1114) DEBUG: lock_initialize: lock initialization started
0(1114) qm_malloc(0x4212f000, 16) called from lock.c: lock_initialize(169)
0(1114) qm_malloc(0x4212f000, 16) returns address 0x422b54e8 on 0 -th hit
0(1114) qm_malloc(0x4212f000, 448) called from timer.c: tm_init_timers(484)
0(1114) qm_malloc(0x4212f000, 448) returns address 0x422b5528 on 0 -th hit
0(1114) qm_malloc(0x4212f000, 40) called from t_stats.c: init_tm_stats(110)
0(1114) qm_malloc(0x4212f000, 40) returns address 0x422b5718 on 0 -th hit
0(1114) qm_malloc(0x4212f000, 12) called from t_stats.c: init_tm_stats(118)
0(1114) qm_malloc(0x4212f000, 12) returns address 0x422b5770 on 0 -th hit
..
stateless - initializing
0(1114) qm_malloc(0x4212f000, 320) called from sl_stats.c: init_sl_stats(128)
0(1114) qm_malloc(0x4212f000, 320) returns address 0x422b5824 on 0 -th hit
0(1114) DEBUG: register_fifo_cmd: new command (sl_stats) registered
0(1114) DEBUG: MD5 calculated: d907c037823644515dfe0ede38ca9976
0(1114) qm_malloc(0x4212f000, 4) called from sl_funcs.c: sl_startup(65)
0(1114) qm_malloc(0x4212f000, 4) returns address 0x422b5994 on 0 -th hit
mysql - initializing
0(1114) qm_malloc(0x4212f000, 396) called from main.c: main(1363)
0(1114) qm_malloc(0x4212f000, 396) returns address 0x422b59c8 on 0 -th hit
0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply
0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply
0(0) fixing /usr/local/lib/ser/modules/rr.so rewriteFromRoute
0(0) fixing /usr/local/lib/ser/modules/registrar.so save
0(0) qm_malloc(0x4212f000, 16) called from dlist.c: new_dlist(87)
0(0) qm_malloc(0x4212f000, 16) returns address 0x422b5b84 on 0 -th hit
...
0(0) qm_malloc(0x80adf80, 96) called from db_row.c: convert_row(49)
0(0) qm_malloc(0x80adf80, 96) returns address 0x80b47cc on 0 -th hit
0(0) qm_malloc(0x80adf80, 96) called from db_row.c: convert_row(49)
0(0) qm_malloc(0x80adf80, 96) returns address 0x80b485c on 0 -th hit
Segmentation fault
it seems that ser can't allocate memory by qm_malloc function. what's wrong with it?
i have tried to reinstall my linux8.0 OS, but the problem is still existed.
please help me!
thanks!
regards,Joe_chen
Hi,
I am trying to create a basic script for routing calls to and from
the PSTN. I have experimented with many scripts over the last month,
this included script is an example. I can use a UA on my
laptop called 'estara' and REGISTER with my ser sip server.
I can make a phone call (INVITE) to the PSTN. I can hang up the call
from either end (BYE) and everything seems to work well.
I am trying to get accounting working. I see the call acceptance
record in my syslog file:
Jan 7 00:21:09 build ser[3507]: ACC: transaction answered: method=INVITE, i-uri=sip:92143357976@build.august.net, o-uri=sip:2143357976@64.90.42.16:5060, call_id= 847b3d22-131d-46b0-abcb-dc8e6a9f32b5(a)192.168.100.100, from= greg <sip:1107@build.august.net>;tag=3a7b9c52, code=200
However, I can't get the 'BYE' event to record the transaction to the
syslog file. The 'BYE' event is bouncing through my SIP server.
Can someone give me a hint?
Thank you,
---greg
Greg Fausak
Included file----example.cfg----------------------------------------
debug=1 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
modparam("acc", "log_level", 1)
modparam("acc", "acc_flag", 1 )
route{
# filter too old messages
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Wow -- Message too large");
break;
};
/* ********* RR ********************************** */
/* Do strict routing if route headers present */
rewriteFromRoute();
/* record-route INVITEs -- all subsequent requests must visit us */
if (method=="INVITE") {
addRecordRoute();
} else if (method=="REGISTER") {
log("here is a register");
save("location");
break;
};
# now check if it really is a PSTN destination which should be handled
# by our gateway; if not, and the request is an invitation, drop it --
# we cannot terminate it in PSTN; relay non-INVITE requests -- it may
# be for example BYEs sent by gateway to call originator
if (!uri=~"sip:9[2-9][0-9]{9}@.*") {
if (method=="INVITE") {
sl_send_reply("403", "Call cannot be served here");
} else if(method=="BYE") {
setflag(1);
forward(uri:host, uri:port);
} else {
forward(uri:host, uri:port);
};
break;
};
# account completed transactions via syslog
setflag(1);
strip(1);
rewritehostport("64.90.42.16:5060");
# forward the request now
if (!t_relay()) {
sl_reply_error();
break;
};
}
Hi All,
I am newbie of this field, thanks everyone help me.
I am interesting in B2BUA, however, except some brief defination in 3261, I could not find any further defination or how to implement about B2BUA, I noticed that SER could be implemented as a B2BUA, where can I find some implementation? or where can I get any description?
Koalas
I just setup an account on iptel.org (joejoe(a)iptel.org)
and I connected it to it fine with Messenger which leads me to believe
that I have done something wrong with my ser installation.
thanks,
-joe
Hi,
I've got ser installed and running (0.8.10 on Linux 2.4.18-18.8.0) I
attempt to connect to it with MS "Windows Messenger" version
4.7.0104 on WindowsXP, but it never connected and I don't see
anything in the logfile showing even an attempt to connect.
Just to make sure that I didn't have any firewall rules blocking
access, I wrote a little script to connect to port 5060 on the sip
server and send it "Asdf\n", that appeared to work fine, as this
appeared in my log:
Jan 6 10:57:39 dell1 /usr/local/sbin/ser[14077]: ERROR: parse_first_line: message too short: 5
Jan 6 10:57:39 dell1 /usr/local/sbin/ser[14077]: ERROR:parse_first_line: bad message
Jan 6 10:57:39 dell1 /usr/local/sbin/ser[14077]: ERROR: parse_msg: message=<Asdf >
Jan 6 10:57:39 dell1 /usr/local/sbin/ser[14077]: ERROR: receive_msg: parse_msg failed
So it appears that packets can get to the sip server fine from the
machine that runs the Messenger client.
I also ran serctl moni and didn't see any different output when I
tried to connect with Messenger, i.e. there is no indication that a
connection attempt is even made.
Is there another IM client I can try connecting with? Any other
tricks to know about Messenger other than what I found at:
http://www.iptel.org/ietf55/use_msn.html
BTW, this is using the mysql module and I created a user with serctl
Jan 6 10:53:50 dell1 ser: ser startup succeeded
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14068]: mod_init(): Database connection opened successfuly
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14068]: INFO: udp_init: SO_RCVBUF is initially 65535
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14068]: INFO: udp_init: SO_RCVBUF is finally 131070
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14068]: INFO: udp_init: SO_RCVBUF is initially 65535
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14068]: INFO: udp_init: SO_RCVBUF is finally 131070
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14080]: INFO: fifo process starting: 14080
Jan 6 10:53:50 dell1 /usr/local/sbin/ser[14080]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
thanks,
-joe
I'm seeing the following errors in syslog when I place a call to one
particular URI in my system. There are 5 devices registered to receive
calls for this URI. Is this a problem? Is there a limit to how many
telephone sets can have the same number appearance?
Jan 3 12:15:44 sip /usr/sbin/ser[9845]: lookup(): Error while appending a branch
Jan 3 12:15:44 sip /usr/sbin/ser[9845]: USRLOC DB lookup not found...
Jan 3 12:15:52 sip /usr/sbin/ser[9846]: ERROR: append_branch: max nr of branches
exceeded