Hi,
I would like to ask whether it is possible to allow permanent registration
with the server i.e. Once the SIP client registered successfully with the
server, the registration record will not expire until we delete it
manually.
Best Regards,
Thomas
Hi everyone,
I'm trying to make a call, based on, when a certain user auths my SER by mysql, if he dials "1", SER forwards the INVITE to Asterisk, The user dials "1" to open external calls and then the wanted number, doing this Asterisk forwards again to Go2call VoIP Server with a certain username and password, in order to auth my user and close the Tunnel, making the Call possible.
What I want to know is that if its possible, or am I crazy !!
I thinking of doing this by the following way:
1. Configure a Certain [context] block with the lines below at extensions.conf:
a)
Context "Go2Call_Calls":
Extensions Description
1 Mark
b)
exten => <external_phone_number>,1,Answer
exten => <external_phone_number>,2,Playback(tt-weasels)
exten => <external_phone_number>,3,VoiceMail(45)
exten => <external_phone_number>,4,Hangup
2. Configure a peer at sip.conf in order to my asterisk can talk to my SER Sip Proxy:
[<external_phone_number>]
type=peer
username=<external_phone_number>
insecure=yes
canreinvite=no
context=Go2Call_Calls
mailbox=<external_phone_number>
host=dynamic
Will it work ??? Does it work this way, or am I getting crazy ? Is there anyone who have already made that ? I'm gonna be doing this till I get our Telecom Sip Numbers to company from our Telecom Provider.
Anyone ??
--
Felipe Martins
Linux System Administrator
Tep Solution Provider
Mundivox Communications
Rua Lauro Muller, 116/Sala 505
RJ - Brasil - 22290-906
Tel.: 55 21 3820-8839
Fax.: 55 21 3820-8844
Hello,
I'm running ser 0.8.14 and rtpproxy CVS on debian sarge.
You can find my ser.cfg. below.
I'm forwarding my calls to a Cisco AS5350 whish serves as a PSTN
gateway. I can place a call to a PSTN number and the phne rings, but
then no audio is sent or received. I use kphone as a client behind NAT.
thnaks
Luca
--- ser.cfg routing ---
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# !! Nathelper
# Special handling for NATed clients; first,NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == "REGISTER" || ! search("^Record-Route:")) {
log("LOG: Someone trying to register from private IP,
rewriting\n");
# This will work only for user agents that support
symmetric
# communication. We tested quite many of them and
majority is
# smart enough to be symmetric. In some phones it takes
a configuration
# option. With Cisco 7960, it is called NAT_Enable=Yes,
with kphone it is
# called "symmetric media" and "symmetric signalling".
fix_nated_contact(); # Rewrite contact with source IP of
signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active to
SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
rewritehostport("$AS5350_IP:5060");
route(1);
break;
};
# 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("my.domain",
"subscriber")) {
www_challenge("my.domain", "0");
break;
};
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC
DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
# -- nathelper --
route[1]
{
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&& !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
# if client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for
example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did
not
# know at time of request processing ? (RFC1918
contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
Hello,
I'm trying to get asterisk to work with SER - I have ser forwarding
the sip connection to asterisk with the following routing logic.
if (uri=~"^sip:8[0-9]{7}.*") {
forward( 10.0.18.3, 5061 );
break;
Asterisk accepts the connection on port 5061 - but I get the following
once it connects to asterisk.
001 -- Executing Answer("SIP/ast.digicen.com-08289858", "") in new stack
002 -- Executing Dial("SIP/ast.digicen.com-08289858",
"IAX2/kknott@NuFone/12039063173") in new stack
003 -- Called kknott@NuFone/12039063173
004 -- Call accepted by 66.225.202.72 (format gsm)
005 -- Format for call is gsm
006 -- Hungup 'IAX2/NuFone/1'
007 == No one is available to answer at this time (1:0/0/0)
008 == Auto fallthrough, channel 'SIP/ast.digicen.com-08289858'
status is 'NOANSWER'
009 > cdr_odbc: Query Successful!
010 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
011 Jan 30 21:41:42 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
012 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
013 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
014 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2551 parse: Too many SIP
headers...
015 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
016 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
017 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
018 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
019 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
020 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
021 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
022 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
023 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
024 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
025 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
026 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
027 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
028 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
029 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
030 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
031 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
032 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
033 Jan 30 21:41:43 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
034 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
035 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
036 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
037 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
038 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
039 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
040 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
041 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
042 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
043 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
044 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
045 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
046 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
047 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
048 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
049 Jan 30 21:41:44 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
050 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
051 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
052 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
053 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
054 Jan 30 21:41:45 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
055 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum
retries exceeded on call
5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 52428
(Non-critical Response)
056 Jan 30 21:41:46 WARNING[9829]: chan_sip.c:729 retrans_pkt: Maximum
retries exceeded on call
5A2650C2-94B0-44AB-B241-66DB67BE2CF7(a)192.168.1.104 for seqno 102
(Non-critical Request)
057 Jan 30 21:41:50 WARNING[9829]: chan_sip.c:2385 find_call: Call
missing call ID from '24.181.176.62'
058
059 *CLI>
060 *CLI>
Asterisk will usually core dump after that - I have no idea where to
start to fix it - No machines are behind NAT.
Thanks,
Patrick
Please can anyone tell me where I can get a complete and full definition of
the criteria to fulfil function is_from_local()
Thanks
Chris
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.8.5 - Release Date: 03/02/2005
dear users
can i get the RTPPROXY runing in another PC, maybe in a CO-located PC in
another country.
all for the expensive cost of internet in mine.
or maybe i need to move all , the server and the rtpproxy module to another
PC ???
if yes, how can be done ??
some examples will be apreciates.
Regards
Gustavo Villegas
Hello all!! We are implementing a voip network in Brasil with 2 or more servers using asterisk and SER. We are still studying the configuration for it but it would be nice to know about similar implementations around the world.
Is there anybody using a SER environment where a registered SIP client from sever A can conect directly (peer to peer) to a user registered in a server B?
Thanks in advance for this case report.
Rgds, Ricardo Poppi
______________________________________________________________________
Giro, a Internet rápida que acompanha você. http://www.giro.com.br
Hi all,
i have a little problem with aliases, that is:
I have created some numeric aliases for some users with serctl in this way:
# serctl alias add 1101 sip:user@mydomain.foo
the command goes right, i can reach the correct users using aliases and i
can see
that records are correctly inserted in the "aliases" table in the mysql database.
The problem is that when i reboot ser aliases don't work even if they are
correctly
in the aliases tables; it seems they are not online or not registered.
The routing script correctly looks for users in the "aliases", but not after
restart!
How can i fix this problem?
Thanks for your help!
Stefano
__________________________________________________________________
Tiscali Adsl 3 Mega Flat, 3 MESI GRATIS!
Con Tiscali Adsl 3 Mega Flat navighi in Rete alla supervelocita'
a soli 29.95 euro al mese senza limiti di tempo. Attivati entro
il 15 Febbraio 2005, 3 MESI sono GRATIS
Scopri come http://abbonati.tiscali.it/adsl/sa/2flat_tc/
Didn't get any response about this bug !
:-((
MOhammad
info(a)beeplove.com wrote on 02/04/2005, 12:32:02 AM:
>
> My mysql database server is different computer than my ser server.
> ser was not able to connect to database.
>
> I put some debug code in authdb_mod.c
>
> static int child_init(int rank)
> {
> db_handle = db_init(db_url);
> LOG(L_ERR, "authdb_mod.c#child_init(): db_url: %s\n", db_url);
> if (!db_handle) {
> LOG(L_ERR, "auth_db:init_child(): Unable to connect
> database\n");
> return -1;
> }
> return 0;
>
>
> }
>
> I have found db_url is not getting the value from my ser.cft.
> Its getting the default value from config.h that was defined before
> compiled.
>
>
> I compiled my ser after editing config.h with mysql param that match my
> situation and my ser is up !!
>
> Thanks,
>
> --
> Mohammad Khan
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
--
Mohammad Khan
sip:beeplove@iptel.org
Hi,
I put the proxy and registar at the two machine.
when proxy deal with calling, I found that proxy
don't query the callee information from the database.
I sniffed the data traffic with ethereal. the Ser log
said in the registrar/lookup.c lines 81 that:
"lookup(): '8644' Not found in usrloc".
What I want to know when the proxy and registrar are
located on the different devices. how can the registrar
tell the proxy the new registered sip ua if the proxy
didn't look the callee in the location table.
If I am not wrong, the usrloc function get_urecord()
did not query the table for every coming call. It attempts
to find callee in a hash table. Can that guarantee to
find the newest information in the database???
Thanks for your instructions.
Best Regards.
Zongjun
