Hi there!
We have two cisco 7912 ip phone using SIP firmware behind Cisco PIX
firewall. These two phone is configured to register to a SER proxy which is on
the public Internet. These two Cisco 7912 ip phones are PATor NAPT
to a single IP.
The problem we have is when these two phones call each other, the
two party cannot hear each other's voice.
If we replace the PIX firewall with a Cisco router, there is no such problem.
Does anyone encounter this problem before? I am clueless on how to
resolve this problem. Any pointers will be much appreciated.
Thank you very much!
Best regards
sekchye
Hi. Is there a tool like wget for SIP? Seems like it could be useful at
least for testing and admin scripts.
I know that SER has fifo on server. Is there some trivial way to use it
from a remote machine?
Thanks,
John.
Hello all,
I have not been able to get my SER to work properly on this SuSE Linux
Pro 9.2 machine. SER appears to hang endlessly. The binary compiled
from src just fine. I can't tell exactly where the problem is, can
anyone lend help in deciphering where to find the culprit, please?
One clue: SER is trying to open the config file because when I rename
it, it errors out. But when it opens the file, it just hangs. I have
even tried inserting random text, and it still hangs. So it appears to
not be parsing the text?
machine:/usr/local/etc/ser #
machine:/usr/local/etc/ser # mv ser.cfg ser.bak
machine:/usr/local/etc/ser # ser
ERROR: loading config file(/usr/local/etc/ser/ser.cfg): No such file or
directory
machine:/usr/local/etc/ser # mv ser.bak ser.cfg
machine:/usr/local/etc/ser # ser
Maybe an error in your script. you obviously tried to send a reply
using sl_send_reply in response to the ACK. What you really wanted
to do was to forward the ACK.
-jiri
At 11:16 PM 12/23/2004, Amozurrutia Jesus wrote:
>I double checked and the ACK are ignored by SER.
>
>The log file thows this:
>Dec 23 13:26:13 sip2 /usr/sbin/ser[21127]: Warning: sl_send_reply: I won't
>send a reply for ACK!!
>
>
>Jesus
>
>-----Original Message-----
>From: Greger V. Teigre [mailto:greger@teigre.com]
>Sent: Lunes, 20 de Diciembre de 2004 01:43 a.m.
>To: Amozurrutia Jesus; serusers(a)lists.iptel.org
>Subject: Re: [Serusers] Problem with ACK
>
>
>Are you sure that the ACK stops at ser? The ACKs should just flow through
>ser. I have seen a similar problem with XLite where Cisco drops the ACK
>because XLite there is a bug in XLIte's Via parsing. This is a Cisco
>gateway appending an x-route-tag to via. Turn on debugging (ALL) on Cisco
>and check if it drops the ACK due to wrong/no branch info in Via.
>g-)
>
>Amozurrutia Jesus wrote:
>> I'm implementing the scenario shown below.
>>
>> ___ ______
>> / 0 \ ---/ \ |
>> /_\ | ATA1 |---| ____ _ _ _ _ _____
>> \______/ | / \ / \/ \/ \/ \ / \
>> |---| FW |---| IP Net |---| CCM |
>> | \____/ \_/\_/\_/\_/ \_____/
>> | | |
>> | | |
>> | |
>> --- ---
>> / \ / \
>> |NAT| |SER|
>> |-T | | |
>> \___/ \___/
>>
>> When calling between the CCM (Cisco CallManager) and the ATA, SER
>> simply ignores the call ACK.
>> The ACK looks like:
>>
>> ACK sip:5559853979*sip1.mcm.net.mx=X.X.71.2+17081@X.X.81.92:5063
>> SIP/2.0 Via: SIP/2.0/UDP X.X.67.218:5060;branch=z9hG4bK29b2c750
>> From: "5559852600" <sip:5559852600@X.X.67.218>;tag=16781758
>> To: <sip:5559853979@mcm.net.mx>;tag=2602576443
>> Date: Tue, 30 Nov 2004 23:53:15 GMT
>> Call-ID: fef8ed00-1da1612d-24d-da4334c8(a)X.X.67.218
>> Route: <sip:5559853979@X.X.81.94;ftag=16781758;lr>
>> Max-Forwards: 70
>> CSeq: 101 ACK
>> Content-Length: 0
>>
>> The call flow:
>>
>> CallManager SIP Server ATA
>> | | |
>> |-- INVITE -------->| |
>> |<-- trying --------| |
>> | |-- INVITE -------->|
>> | |<-- Trying --------|
>> | |<-- Ringing -------|
>> |<-- Ringing -------| |
>> | |<-- OK ------------|
>> |<-- OK ------------| |
>> |-- ACK ----------->| |
>> | |<-- OK ------------|
>> |<-- OK ------------| |
>> |-- ACK ----------->| |
>> | |<-- OK ------------|
>> |<-- OK ------------| |
>> |-- ACK ----------->| |
>> | |<-- OK ------------|
>> |<-- OK ------------| |
>> |-- ACK ----------->| |
>> | |<-- OK ------------|
>> ......
>>
>> My guess is that SER does not accept the URI
>> "5559853979*sip1.mcm.net.mx=X.X.71.2+17081@X.X.81.92:5063" because it
>> contains "*+=" signs ore something similar.
>> When calling from ATA - ATA there is no preoblem because ATAs
>> construct the ACK message different (URI and Rote flipped).
>>
>> Any ideas?
>>
>> Thanks in advance,
>>
>> Jesus
>>
>>
>> _______________________________________________
>> 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
--
Jiri Kuthan http://iptel.org/~jiri/
Hello,
welcome to the world of SIP, where the end devices are supposed to be the
intelligent network, not the network itself.
To really make this and other features ISDN-like features work in the
network, one needs a b2bua somewhere.
A---SER---B2BUA---SER---B
C---^
^accounting SER
A calls from PSTN, B2BUA receives the call, sends INVITE to B
B sends 302 to B2BUA.
B2BUA sends INVITE in the name of B to C.
Regards,
Martin
> -----Original Message-----
> From: serusers-bounces(a)lists.iptel.org
> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Atle Samuelsen
> Sent: Wednesday, December 22, 2004 9:40 AM
> To: Richard
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] call log and accounting for forwarded
> and referredcalls
>
>
> At the moment SER does'nt do recursion on 302. If ser had
> done this, The
> world would be a bether place for all us.
> Anyhow. in a A-B-C-D scenario.. A should pay to B, B to C and C to D.
>
>
>
> -Atle
>
> * Richard <richard(a)o-matrix.org> [041222 09:04]:
> >
> >
> > > -----Original Message-----
> > > From: Juha Heinanen [mailto:jh@tutpro.com]
> > > Sent: Tuesday, December 21, 2004 9:25 PM
> > > To: Richard
> > > Cc: serusers(a)lists.iptel.org
> > > Subject: [Serusers] call log and accounting for forwarded
> and referred
> > > calls
> > >
> > > Richard writes:
> > >
> > > > When a SIP call is blind-transferred with REFER and
> forwarded with "302
> > > > moved temporarily", UA would start a brand new call.
> The problem is how
> > > to
> > > > log and account for their calls. For example, A calls
> B, B sends 302
> > > and
> > > > uses C's number as contact. The new call is made from
> A to C. The call
> > > log
> > > > would show it is from A to C. The call log should at
> least have an
> > > > indication of B forwarding the call. Also B is
> supposed to pay the
> > bill.
> > > It
> > > > is not A although the call log shows it is A to C. A
> has no knowledge
> > > that a
> > > > toll call is made when calling B.
> > >
> > > richard,
> > >
> > > i disagree that in case of 302, b should pay the bill.
> 302 means "b has
> > > moved to c and it is up to you if you want to try this new uri".
> > >
> > > if you want b to pay the bill, then b should configure
> ser to FORWARD
> > > the call to c, not to REDIRECT a to c.
> > >
> >
> > The issue is that A has no choice to be forwarded or not.
> When a 302 is
> > received by A, there is no option for A to continue or
> reject the call. In
> > this example, B (an IP phone) sets his phone forwarding to
> C which is a long
> > distance number. A is from PSTN. When A makes a call to B,
> B sends 302 to
> > the PSTN gateway. The gateway forwards the call to ser
> which routes it back
> > to C via the PSTN gatway. So in ser's call log, I see a
> call from A to C.
> > Apparently I can't charge A or C. Only B is in my domain.
> But B is not even
> > in the second call log. In my understanding, if B sets the
> forward setting
> > on his phone to a toll number, he should be the one paying the bill.
> >
> > This also applies even if A is in my domain.
> >
> > Cheers,
> > Richard
> >
> >
> > _______________________________________________
> > 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
>
Hi:
Merry Christmas everyone.
When I use SER as proxy server, I found a strange problem.
My SIP UA have audio and video two media stream. After transfered
by SER, the SIP message recieved by my callee UA lost its port information.
And I tested it on the sip.iptel.org's SER, this time the audio port
info is not lost,
but has changed. The video port is still lost( I wonder whether this is
caused by
NAT). My SER is located in the same private network, why the SER lost the
audio and video port info while other info is untouched ????
Appreciation for your help and instructions
Best Regards.
--
IPv6 on Everything, Power to Everything. Everything runs on Linux
--------------------------------------------
Sun Zongjun
Sinux Sun
Programmer
BII Group Holdings Ltd (BII Group)
E-mail: zjsun(a)biigroup.com
Web: http://www.biigroup.com; http://www.ipv6.net.cn
Phone Office: +86-10-65836569-175
Fax Office: +86-10-65836565
Hello Guys, I am running ser-0.9.0 on a freeBSD machine. Igot the following error when compiling:
sip# gmake modules=modules/uri_radius modules
gmake[1]: Entering directory `/root/src/ser/ser-0.9.0/modules/uri_radius'
../../Makefile.rules:77: checks.d: No such file or directory
../../Makefile.rules:77: urirad_mod.d: No such file or directory
gmake[1]: Leaving directory `/root/src/ser/ser-0.9.0/modules/uri_radius'
gmake[1]: Entering directory `/root/src/ser/ser-0.9.0/modules/uri_radius'
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -I/usr/local/include -c checks.c -o checks.o
checks.c: In function `radius_does_uri_exist':
checks.c:77: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast
checks.c:77: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast
checks.c:77: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast
checks.c:77: error: too many arguments to function `rc_avpair_add'
checks.c:85: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast
checks.c:85: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast
checks.c:85: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast
checks.c:85: error: too many arguments to function `rc_avpair_add'
checks.c:92: warning: passing arg 1 of `rc_auth' makes integer from pointer without a cast
checks.c:92: warning: passing arg 3 of `rc_auth' from incompatible pointer type
checks.c:92: warning: passing arg 4 of `rc_auth' from incompatible pointer type
checks.c:92: error: too many arguments to function `rc_auth'
gmake[1]: *** [checks.o] Error 1
gmake[1]: Leaving directory `/root/src/ser/ser-0.9.0/modules/uri_radius'
sip#
sip# gmake modules=modules/auth_radius modules
gmake[1]: Entering directory `/root/src/ser/ser-0.9.0/modules/auth_radius'
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_freebsd -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -I/usr/local/include -c authrad_mod.c -o authrad_mod.o
authrad_mod.c: In function `mod_init':
authrad_mod.c:110: error: `DICT_VENDOR' undeclared (first use in this function)
authrad_mod.c:110: error: (Each undeclared identifier is reported only once
authrad_mod.c:110: error: for each function it appears in.)
authrad_mod.c:110: error: `vend' undeclared (first use in this function)
authrad_mod.c:134: warning: assignment makes pointer from integer without a cast
authrad_mod.c:139: error: too many arguments to function `rc_conf_str'
authrad_mod.c:139: error: too many arguments to function `rc_read_dictionary'
authrad_mod.c:144: warning: implicit declaration of function `rc_dict_findvend'
authrad_mod.c:159: error: too many arguments to function `rc_dict_findattr'
authrad_mod.c:159: error: too many arguments to function `rc_dict_findval'
gmake[1]: *** [authrad_mod.o] Error 1
gmake[1]: Leaving directory `/root/src/ser/ser-0.9.0/modules/auth_radius'
sip#
What could be wrong? I had the same errors when using 0.8.14
thanks,
Pablo.
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Hi
First happy holidays to all, I have setup SER Proxy
and working fine as long as I used the URI of type
User(a)mydomain.com. Now I want to assign each user an
ID of 7471234 which they have to prefix with * (e.g
*7471234) to call other user. Lets say UserA with with
number 7471234 wants to dial UserB with Tel # 7472222
dials *7472222 and gets connected as long as the UserB
is registered with my SER Proxy. If not 404 msg is
sent UserA. if UserA dial a PSTN numbe either using +
& CounryCode & Number, or 00 & CounryCode & Number, &
or 011 & CounryCode & Number, they get forwarded to
my Asterisk Server.
Please help me identify what should I use the Aliases
Table or Subscriber Table or DNS NAPTR to map uri to
Number (eg 7471234 maps to UserA(a)mydomain.com).
Thanks in Advance, Merry Xmas and Happy New Year!
Abid Mirza
=====
Abid A. Mirza
Bits & Byte Tech., Inc.
146 West 29 Street , 10 Fl.
New York, NY 10001
Tel : +1 (212) 967-1616 Fax : +1 (212) 967-0672
Mob : +1 (917) 582-2290
Email : amirza_nyc(a)yahoo.com