Hello,
I am working on a project in which calls are placed from an Asterisk server towards the PSTN or SIP clients and then connected to another phone of the PSTN or another SIP client. Those calls are placed by a user (caller) through an interface. First a call is placed towards the caller from an extension in Asterisk and once the caller picks up the phone, the second call is placed and both are connected.
In a first time all SIP clients were registered to Asterisk and there was no SER on the picture. Everything was working fine. But I have decided to use SER as SIP proxy now. So the SIP clients are registered with SER and Asterisk is still the originating point of all calls and provides the connection to the PSTN. Asterisk and SER are running on the same machine (Asterisk has port 5061 and SER port 5060) and Asterisk is registered to SER:
==sip.conf==
[general] port=5061 bindport=5061 bindaddr=0.0.0.0 disallow=all ;allow=gsm allow=ulaw allow=alaw ;context=bogon-calls context=from-sip autocreatepeer=yes
register=> asterisk:password@192.168.1.24:5060/maxAS
[SERADDRESS] type=friend username=asterisk secret=password host=192.168.1.24 fromdomain=192.168.1.24
[maxAS] type=friend secret=password2 qualify=yes nat=no host=dynamic canreinvite=no context=from-sip
Most things work fine: * a user can ask Asterisk to call a SIP client and connect it to another SIP client: Asterisk finds its way to SER and both calls are happily connected. * a user can also ask a call to be placed towards a SIP client and then this call is connected to a PSTN call, no problem either.
The problem is when the user wants a call towards the PSTN to be connected to a SIP client. Once the call going towards the PSTN has been answered, Asterisk attempts to contact a SIP client which is registered in SER but SER sees the call coming from "sip:(PSTN phone number of the first call)@hostIP". There is no such client in the database and it is therefore not allowed to connect. In the following example Zap/1-1 is the interface towards the PSTN, 0761111111 the phone number dialled, and 6644 an extension registered with SER. Here is the Asterisk output.
> Channel Zap/1-1 was answered. -- Executing Answer("Zap/1-1", "") in new stack -- Executing Goto("Zap/1-1", "from-sip|6644|1") in new stack -- Goto (from-sip,6644,1) -- Executing Answer("Zap/1-1", "") in new stack -- Executing Dial("Zap/1-1", "SIP/6644@SERADDRESS||r") in new stack -- Called 6644@SERADDRESS Oct 10 11:41:16 WARNING[4264]: chan_sip.c:7972 handle_response: Forbidden - wrong password on authentication for INVITE to '"0761111111" sip:0761111111@192.168.1.24;tag=as537d1ed7' -- SIP/SERADDRESS-50e7 is circuit-busy
Here is the portion of the Asterisk Dialplan that is relevant and the ser.cfg (which comes from http://siprouter.onsip.org/doc/ gettingstarted/):
==extensions.conf== [context1] exten => 666,1,Answer() exten => 666,2,Goto(from-sip,6644,1)
[from-sip] exten => _.,1,Answer exten => _.,2,Dial(SIP/${EXTEN}@SERADDRESS,,r) exten => _.,3,Hangup
==ser.cfg== debug=3 fork=no log_stderror=yes
listen=192.168.1.24 # INSERT YOUR IP ADDRESS HERE port=5060 children=4
dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:password@192.168.1.26/ser"
loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so"
modparam("auth_db|uri_db|usrloc", "db_url", "mysql:// ser:password@192.168.1.26/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1)
route {
# ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; };
if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; };
# ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); };
# ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; };
# ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; };
if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; };
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }
route[1] {
# ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; }
route[2] {
# ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying");
if (!www_authorize("192.168.1.24","subscriber")) { www_challenge("192.168.1.24","0"); break; };
if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; };
consume_credentials();
if (!save("location")) { sl_reply_error(); }; }
route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("192.168.1.24","subscriber")) { proxy_challenge("192.168.1.24","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; };
consume_credentials();
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }
I guess (but I am new to SER) that the idea is to make the call be placed as the SIP user "asterisk" and not as the SIP user "0761111111" but I don't know where to do that (inside asterisk before the SIP call is placed? inside ser.cfg?) or maybe just to skip the authentication for SIP URIs that look like a phone number. I also read in http://siprouter.onsip.org/doc/gettingstarted/ch09.html how to relay calls placed towards the PSTN from SER towards a gateway (Asterisk here) but I think it wouldn't do any good to send back the call to Asterisk. Asterisk would just end up placing another call towards the PSTN and that's not what I want.
Any suggestion will be greatly appreciated,
Thanks,
//Max
You can use allow_trusted() to allow calls from your Asterisk. see the chapter you referred to (Getting Started) for an example of how to use it. Just add the IP of your Asterisk to the table of trusted IPs. g-)
Max Weltz wrote:
Hello,I am working on a project in which calls are placed from an Asterisk server towards the PSTN or SIP clients and then connected to another phone of the PSTN or another SIP client. Those calls are placed by a user (caller) through an interface. First a call is placed towards the caller from an extension in Asterisk and once the caller picks up the phone, the second call is placed and both are connected.
In a first time all SIP clients were registered to Asterisk and there was no SER on the picture. Everything was working fine. But I have decided to use SER as SIP proxy now. So the SIP clients are registered with SER and Asterisk is still the originating point of all calls and provides the connection to the PSTN. Asterisk and SER are running on the same machine (Asterisk has port 5061 and SER port 5060) and Asterisk is registered to SER:
==sip.conf==
[general] port=5061 bindport=5061 bindaddr=0.0.0.0 disallow=all ;allow=gsm allow=ulaw allow=alaw ;context=bogon-calls context=from-sip autocreatepeer=yes
register=> asterisk:password@192.168.1.24:5060/maxAS
[SERADDRESS] type=friend username=asterisk secret=password host=192.168.1.24 fromdomain=192.168.1.24
[maxAS] type=friend secret=password2 qualify=yes nat=no host=dynamic canreinvite=no context=from-sip
Most things work fine:
- a user can ask Asterisk to call a SIP client and connect it to
another SIP client: Asterisk finds its way to SER and both calls are happily connected.
- a user can also ask a call to be placed towards a SIP client and
then this call is connected to a PSTN call, no problem either.
The problem is when the user wants a call towards the PSTN to be connected to a SIP client. Once the call going towards the PSTN has been answered, Asterisk attempts to contact a SIP client which is registered in SER but SER sees the call coming from "sip:(PSTN phone number of the first call)@hostIP". There is no such client in the database and it is therefore not allowed to connect. In the following example Zap/1-1 is the interface towards the PSTN, 0761111111 the phone number dialled, and 6644 an extension registered with SER. Here is the Asterisk output.
> Channel Zap/1-1 was answered. -- Executing Answer("Zap/1-1", "") in new stack -- Executing Goto("Zap/1-1", "from-sip|6644|1") in new stack -- Goto (from-sip,6644,1) -- Executing Answer("Zap/1-1", "") in new stack -- Executing Dial("Zap/1-1", "SIP/6644@SERADDRESS||r") in new stack -- Called 6644@SERADDRESSOct 10 11:41:16 WARNING[4264]: chan_sip.c:7972 handle_response: Forbidden - wrong password on authentication for INVITE to '"0761111111" sip:0761111111@192.168.1.24;tag=as537d1ed7' -- SIP/SERADDRESS-50e7 is circuit-busy
Here is the portion of the Asterisk Dialplan that is relevant and the ser.cfg (which comes from http://siprouter.onsip.org/doc/ gettingstarted/):
==extensions.conf== [context1] exten => 666,1,Answer() exten => 666,2,Goto(from-sip,6644,1)
[from-sip] exten => _.,1,Answer exten => _.,2,Dial(SIP/${EXTEN}@SERADDRESS,,r) exten => _.,3,Hangup
==ser.cfg== debug=3 fork=no log_stderror=yes
listen=192.168.1.24 # INSERT YOUR IP ADDRESS HERE port=5060 children=4
dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:password@192.168.1.26/ser"
loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so"
modparam("auth_db|uri_db|usrloc", "db_url", "mysql:// ser:password@192.168.1.26/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1)
route {
# ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; };
if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; };
# ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); };
# ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; };
# ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; };
if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; };
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }
route[1] {
# ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; }
route[2] {
# ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying");
if (!www_authorize("192.168.1.24","subscriber")) { www_challenge("192.168.1.24","0"); break; };
if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; };
consume_credentials();
if (!save("location")) { sl_reply_error(); }; }
route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("192.168.1.24","subscriber")) { proxy_challenge("192.168.1.24","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; };
consume_credentials();
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }
I guess (but I am new to SER) that the idea is to make the call be placed as the SIP user "asterisk" and not as the SIP user "0761111111" but I don't know where to do that (inside asterisk before the SIP call is placed? inside ser.cfg?) or maybe just to skip the authentication for SIP URIs that look like a phone number. I also read in http://siprouter.onsip.org/doc/gettingstarted/ch09.html how to relay calls placed towards the PSTN from SER towards a gateway (Asterisk here) but I think it wouldn't do any good to send back the call to Asterisk. Asterisk would just end up placing another call towards the PSTN and that's not what I want.
Any suggestion will be greatly appreciated,
Thanks,
//Max _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
OK,
thank you for the hint. It works now even though for some reason I need to have the IP of Asterisk both in the 'trusted' table and the 'permissions.allow' file.
Thanks
Max
On Oct 11, 2007, at 6:39 , Greger V. Teigre wrote:
You can use allow_trusted() to allow calls from your Asterisk. see the chapter you referred to (Getting Started) for an example of how to use it. Just add the IP of your Asterisk to the table of trusted IPs. g-)
Max Weltz wrote:
Hello,I am working on a project in which calls are placed from an Asterisk server towards the PSTN or SIP clients and then connected to another phone of the PSTN or another SIP client. Those calls are placed by a user (caller) through an interface. First a call is placed towards the caller from an extension in Asterisk and once the caller picks up the phone, the second call is placed and both are connected.
In a first time all SIP clients were registered to Asterisk and there was no SER on the picture. Everything was working fine. But I have decided to use SER as SIP proxy now. So the SIP clients are registered with SER and Asterisk is still the originating point of all calls and provides the connection to the PSTN. Asterisk and SER are running on the same machine (Asterisk has port 5061 and SER port 5060) and Asterisk is registered to SER:
==sip.conf==
[general] port=5061 bindport=5061 bindaddr=0.0.0.0 disallow=all ;allow=gsm allow=ulaw allow=alaw ;context=bogon-calls context=from-sip autocreatepeer=yes
register=> asterisk:password@192.168.1.24:5060/maxAS
[SERADDRESS] type=friend username=asterisk secret=password host=192.168.1.24 fromdomain=192.168.1.24
[maxAS] type=friend secret=password2 qualify=yes nat=no host=dynamic canreinvite=no context=from-sip
Most things work fine:
- a user can ask Asterisk to call a SIP client and connect it to
another SIP client: Asterisk finds its way to SER and both calls are happily connected.
- a user can also ask a call to be placed towards a SIP client
and then this call is connected to a PSTN call, no problem either.
The problem is when the user wants a call towards the PSTN to be connected to a SIP client. Once the call going towards the PSTN has been answered, Asterisk attempts to contact a SIP client which is registered in SER but SER sees the call coming from "sip: (PSTN phone number of the first call)@hostIP". There is no such client in the database and it is therefore not allowed to connect. In the following example Zap/1-1 is the interface towards the PSTN, 0761111111 the phone number dialled, and 6644 an extension registered with SER. Here is the Asterisk output.
> Channel Zap/1-1 was answered. -- Executing Answer("Zap/1-1", "") in new stack -- Executing Goto("Zap/1-1", "from-sip|6644|1") in new stack -- Goto (from-sip,6644,1) -- Executing Answer("Zap/1-1", "") in new stack -- Executing Dial("Zap/1-1", "SIP/6644@SERADDRESS||r") in newstack -- Called 6644@SERADDRESS Oct 10 11:41:16 WARNING[4264]: chan_sip.c:7972 handle_response: Forbidden - wrong password on authentication for INVITE to '"0761111111" sip:0761111111@192.168.1.24;tag=as537d1ed7' -- SIP/SERADDRESS-50e7 is circuit-busy
Here is the portion of the Asterisk Dialplan that is relevant and the ser.cfg (which comes from http://siprouter.onsip.org/doc/ gettingstarted/):
==extensions.conf== [context1] exten => 666,1,Answer() exten => 666,2,Goto(from-sip,6644,1)
[from-sip] exten => _.,1,Answer exten => _.,2,Dial(SIP/${EXTEN}@SERADDRESS,,r) exten => _.,3,Hangup
==ser.cfg== debug=3 fork=no log_stderror=yes
listen=192.168.1.24 # INSERT YOUR IP ADDRESS HERE port=5060 children=4
dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:password@192.168.1.26/ser"
loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so"
modparam("auth_db|uri_db|usrloc", "db_url", "mysql:// ser:password@192.168.1.26/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1)
route {
#
# Sanity Check Section
#
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; };
if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; };
#
# Record Route Section
#
if (method!="REGISTER") { record_route(); };
#
# Loose Route Section
#
if (loose_route()) { route(1); break; };
#
# Call Type Processing Section
#
if (uri!=myself) { route(1); break; };
if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; };
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }
route[1] {
#
# Default Message Handler
#
if (!t_relay()) { sl_reply_error(); }; }
route[2] {
#
# REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying");
if (!www_authorize("192.168.1.24","subscriber")) { www_challenge("192.168.1.24","0"); break; };
if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; };
consume_credentials();
if (!save("location")) { sl_reply_error(); }; }
route[3] {
#
# INVITE Message Handler
#
if (!proxy_authorize("192.168.1.24","subscriber")) { proxy_challenge("192.168.1.24","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; };
consume_credentials();
lookup("aliases"); if (uri!=myself) { route(1); break; };
if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; };
route(1); }
I guess (but I am new to SER) that the idea is to make the call be placed as the SIP user "asterisk" and not as the SIP user "0761111111" but I don't know where to do that (inside asterisk before the SIP call is placed? inside ser.cfg?) or maybe just to skip the authentication for SIP URIs that look like a phone number. I also read in http://siprouter.onsip.org/doc/ gettingstarted/ch09.html how to relay calls placed towards the PSTN from SER towards a gateway (Asterisk here) but I think it wouldn't do any good to send back the call to Asterisk. Asterisk would just end up placing another call towards the PSTN and that's not what I want.
Any suggestion will be greatly appreciated,
Thanks,
//Max _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Greger V. Teigre -
Max Weltz