When I go to make a connection is appearing the following error in log:
Feb 23 14:19:07 tob04 /usr/local/sbin/ser[43997]: ERROR: udp_send: sendto(sock,0x284690f8,9272,0,0x28462230,16): Message too long(40)
Feb 23 14:19:07 tob04 /usr/local/sbin/ser[43997]: msg_send: ERROR: udp_send failed
Feb 23 14:19:07 tob04 /usr/local/sbin/ser[43997]: ERROR: t_forward_nonack: sending request failed
somebody would know what I am making of made a mistake?
Hi All,
I am facing problem to get SER to do a SRV lookup. I have the
following scenario…
Step 1:
SIP UA1 registers with SER (testdomain.com) with contact address
user(a)testdomain.com
SIP UA2 registers with SER (testdomain.com) with contact address
user(a)testdomain.com
Step 2:
SIP UA3 calls user(a)testdomain.com. This results in error "478
Unresolveable destination"
Problem
I would expect SER to do a DNS SRV lookup on testdomain.com and get
sip1.testdomain.com and sip2.testdomain.com. Unfortunately, SER only
does A and AAAA query. (verified via ethereal) and does not do SRV
lookup. How can I make it do SRV lookup ? I am using the default
ser.cfg file.
FYI: using host -t srv _sip._udp.testdomain.com gives me the correct
SRV response.
DNS entries
_sip._udp.testdomain.com. 3600 IN SRV
0 1 5060 sip1.testdomain.com.
_sip._udp.testdomain.com. 3600 IN SRV
0 1 5060 sip2.testdomain.com.
Any pointers would be really appreciated.
Thanks,
Chetan
Hi,
Does anyone know how to configure CDRTool global.inc.
I want to use it with SER ACC.
I have this error :
Database error: pconnect(db-log, cdradmin, $Password) failed.
MySQL Error: ()
Session halted.
Regards
Vos Solutions Voix-Data !
Nicolas Ruiz
Service Technique
Ligne directe : + 33 (0) 1 56 38 39 71
Fax :+ 33 (0) 1 47 24 74 77
nruiz(a)vivaction.com
Immeuble Plein Ouest
177 av. Georges Clemenceau
92024 Nanterre - France
Tel : 0 811 02 6000
www.vivaction.com
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Thanks,
That is correct, it hits this part:
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
And that is because the ACK keeps getting back at the SER. (It gets
routed back to the SER itself, as described below). I want to solve this
problem; the ACK should be routed to my UA, but apparently it does not
since it's local IP address it not stated in the SIP header. (Only the
public IP address of the SER-machine is stated in the SIP header.)
Any other ideas?
Roel
> -----Original Message-----
> From: Java Rockx [mailto:javarockx@gmail.com]
> Sent: Wednesday, February 23, 2005 13:17
> To: Roel van Dijk
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] error revisited: I won't send a reply for
ACK!!
>
> I had this problem just a while ago. It was report to me, from Richard
> I think, that somewhere in my ser.cfg an ACK was hitting an
> sl_send_reply() which is incorrect.
>
> ACK messages must hit a t_relay() to avoid this message.
>
> Regards,
> Paul
>
Here's my setup:
NATed UAs -- firewall SER/RTPProxy (bridging mode) -- internet
I'm using a slightly altered version of the 'alg.cfg' that comes with
the nathelper module.
I'm trying to call a UA inside my network from the internet using a
public iptel.org account. Here's the trace of the SIP messages:
UA1 firewall iptel
| | <--INVITE-- |
| | --trying--> |
| <--INVITE-- | |
| --trying--> | |
| --ringing--> | |
| --OK--> | |
| | --OK--> |
| | <--ACK-- |
| ACK |
| ACK |
| ACK |
| | |
If we look at the ACK that comes back from Iptel, we see that the IP
address of UA1 is missing, and therefore the routing fails, and never
leaves the SER on the firewall (obscured IP addresses, but given a
name):
U IPTEL.IP.ADDRESS:5060 -> FIREWALL.PUBLIC.IP.ADDRESS:5060
ACK sip:126@FIREWALL.PUBLIC.IP.ADDRESS:5060 SIP/2.0..Record-Route:
<sip:IPTEL.IP.ADDRESS;ftag=2664746953;lr=on>..Content-Length:
0..Contact: <sip:rdvdijk@HOME.PUBLIC.IP.ADDRESS:5060;nat=yes>..Call-ID:
C8E48A38-6C23-42AB-9EF1-6213B2DF4CD0@192.168.0.2..Max-Forwards:
16..CSeq: 2 ACK..From:<sip:rdvdijk@iptel.org>;tag=2664746953..Route:
<sip:FIREWALL.PUBLIC.IP.ADDRESS;ftag=2664746953;lr=on>..To:
<sip:126@my.realm>;tag=947fb3ad8fdd13a6..User-Agent:
SJLabs-SJphone/1.30.235d..Via: SIP/2.0/UDP
IPTEL.IP.ADDRESS;branch=0..Via: SIP/2.0/UDP
192.168.0.2:5060;rport=5060;received=HOME.PUBLIC.IP.ADDRESS;branch=z9hG4
bKc0a800020131c9b1421c4c560000318a000000bc..P-hint:
rr-enforced..P-NATed-URI: YES..P-RTP-Proxy: NO..P-Marked-Contact:
YES..P-Behind-NAT: Yes....
As we see, the UA1's local IP address is nowhere to be found in the ACK
coming back from Iptel. The other local IP address from my the
iptel-account can be found (192.168.0.2). What we do see is: Route:
<sip:FIREWALL.PUBLIC.IP.ADDRESS;ftag=2664746953;lr=on>, and the ACK is
routed back to the SER a few times until I get this message in my logs:
Warning: sl_send_reply: I won't send a reply for ACK!!
When I hang up both sides, I see the same behaviour for the BYE message,
which is routed back to the SER itself as well.
So, what is my problem?
Roel van Dijk
PS config files can be supplied if needed, but it looks very much like
the original alg.cfg, no serious changes as far as I can see.
Hi guys,
I'm experiencing troubles w/ dial plans. Here is part of my ser.cfg :
if (is_user_in("From", "pstn-allowed")){
if (uri=~"^sip:0") {
strip(0);
t_relay();
log(0, "**************** LOCAL ***************\n");
break;
break;
break;
} else {
if (uri=~"^sip:[1-9]") {
log(0, "\n\n************ PSTN ***********\n");
rewritehostport("xx.xx.xx.xx:5060");
append_branch("xx.xx.xx.xx");
t_relay();
t_relay_to_tcp("xx.xx.xx.xx", "5060");
log(0, "********************************************\n\n\n");
break;
};
};
When I call local user 123456 this way 0123456 ( 0 is the prefix ) it's forwarded to the pstn gw???!!!
If I call any pstn number it works fine... only internal calls doesn't work the way they have to.
Somebody can tell me where's the problem.
Thanks in advance!
Hi All,
I'm presently working on a SER v. 0.8.14 with Postgres db module for
accounting purposes.
There is one inconvenience we discovered while working with
failure_route feature.
In case when the call is handled by Cisco_A we have INVITE,ACK,BYE
messages stored in db which is good.
The problem is when the call is redirected from Cisco_A to Cisco_B we
get only BYE.
Awcourse following the ngrep it appears that the redirection is created
by using INVITE, ...., BYE.
The question is why i cannot obtain full information about the call
stages in db in case of redirection.
Is there anything which has to be added into config file.
Regards.
Pawel.
Dear ALL:
I find a program named "asterisk_b2bua" on
http://developer.berlios.de/projects/b2bua/
And I also download them(two components) and try to test it.
But I have not enough knowledge about asterisk. It seems a Software PBX.
Does asterisk_b2bua work? Does anybody ever try it?
I have questions about my scenario.
|======================> UA2 (Internet)
|
UA1 ===> SER ===> Asterisk B2BUA ===> Trunking A (PSTN)
|
|
CDR + Prepaid + Handle Calls(Tear-down when call
during limited)
|
|
Authentication ( Radius / DB )
Q1. Will Asterisk's B2BUA pass through the "To" (such as
0939749xxx(a)ser.xxx.net.tw PSTN number) to Trunking A?
In another word, is the B2BUA not necessary to rewrite the phone number?
Q2. How can B2BUA know when to tear-down this call if the call has some limits.
For example, UA1 has only 120 seconds to use this International call.
Q3. Is it necessary for Radius Authentication? Is it possible if no
radius exists
and SER and Asterisk use the same MySQL database.
Q4. What do I need to download and setup when I use Asterisk
to start this prepaid compoment? The first one shall be
asterisk-1.0.5.tar.gz.
Maybe some Mysql supported scripts or installaion. And any more??
Best Regard
Charles
Hi,
I have ser 0.8.12 and NATHELPER module launch.
But the function "nat_uac_test" don't start.
What can I do, tu use this function? recompile NATHELPER.so???
Thanks in advance
Nicolas RUIZ