[Serusers] SER + ASTERISK + CISCO2600

Ahmed Boreau ahmed.boreau at esmt.sn
Wed Dec 1 16:34:56 CET 2004


Hi,

I'm setting up a ser + * + cisco 2600 architecture where SER's clients got 
a 72.. (4 digits) plan number and Cisco's users got a 1... (4digits).
I got 2 x-lite clients, MSN Messenger and a SJPhone registered to SER; A 
analog phone, an IP Phone and a Wireless IP Phone registered to my cisco 
gateway.

Below are my troubles:

1 - Calls between 2 SER's clients are OK; caller got a tone ringing but 
nothing for the called (no indication and no callerID). Do I need to add a 
setcallerdId in my extensions.conf ? Previously, I was not obliged and I 
had callerid and an indication;
2 - When a x-lite client calls the analog phone linked to cisco's gateway, 
it's OK (ringing, answering, hanging up). life is great !!!!;
3 - When a x-lite client calls an IP Phone (wired or wireless), called got 
the ringing, caller heared the ringing tone but when caller tried to 
answer, call is destroyed;
4 - When an IP phone (wired or wireless) send a call to an x-lite client, 
it is not ringing in the called side (no ringing tone, no callerid 
indication) and asterisk got a such message:
   == Spawn extension (default, 7201, 1) exited non-zero on 
'Local/7201 at default-9565,2'

and then destroyed the call

Destroying call '502DA4E9-42E211D9-8AACFF3B-E217174C at 10.0.0.8'.

I thaught, I got some problems in my extensions.conf may be I'm wrong !!!

my configurations files are linked.

Thanks in advance




Ahmed Boreau
D. Informatique
ESMT  
-------------- next part --------------
dial-peer voice 123 voip
        description Route to Asterisk server
        destination-pattern 72..
        session protocol sipv2
        session target ipv4:10.0.0.13:5070
        session transport udp
        dtmf-relay rtp-nte
        codec gsm
        no vad
sip-ua
        retry invite 3
        retry response 3
        retry bye 3
        retry cancel 3
        timers trying 1000
        sip-server ipv4:10.0.0.13
-------------- next part --------------
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#

# ----------- global configuration parameters ------------------------

#debug=3         # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no        # (cmd line: -E)

/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/

check_via=no    # (cmd. line: -v)
dns=no           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
port=5060
children=4
# ajout du 11.11.04
listen=10.0.0.242
listen=127.0.0.1
alias=ser.esmt.sn
alias=127.0.0.1
#-----------------------------
fifo="/tmp/ser_fifo"

# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database
# 14.10.04
loadmodule "/usr/local/lib/ser/modules/mysql.so"

loadmodule "/usr/local/lib/ser/modules/sl.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("usrloc", "db_mode", 2)

# -- auth params --
# Uncomment if you are using auth module
#
#ajout du 11.11.04

#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)

#11.11.04
#modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("tm","fr_inv_timer",15)
modparam("tm","fr_timer",10)
# -------------------------  request routing logic -------------------

# main routing logic

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;
        }

        # 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("10.0.0.242", "subscriber")) {
#                               www_challenge("10.0.0.242", "0");
#                               break;
#                       };
                        save("location");
                        break;
                }
                # native SIP destinations are handled using our USRLOC DB
                if (!lookup("location")) {
#                       sl_send_reply("404", "Not Found");
#                       Redirection to CISCO GATEWAY via *
                        rewritehostport("10.0.0.13:5070");
                        t_relay();
                        break;
                }
        }
        # forward to current uri now; use stateful forwarding; that
        # works reliably even if we forward from TCP to UDP

        if (method=="INVITE"){
                if (uri=~"^sip:72[0-9]{2}@*") {
                        log(1,"Forwarding to Asterisk\n");
                        rewritehostport("10.0.0.13:5070");
                        t_relay();
                        break;
                }
#
        }

        if(!t_relay()){
                sl_reply_error();
        }

}
                              
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