[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
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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
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#
# $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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