Hi
I want to know the best way to place calls from SIP endpoints to PSTN phones
using my Siemens HiPath HG1500
PBX.
Is Asterisk the best solution?? redirecting SER calls to Asterisk, and then
to the Siemens PBX?
Cas I use Voice Modems instead?
Does someone have better ideas?
Thanks
Joao Pereira
I think kphone supports IPv6.
Unclassified.
>>> Kuniyoshi Murata <kuni(a)ej-interpreter.net> 01/04/05 01:01PM >>>
Hi,
I'm running SER on dual stack (IPv4 & IPv6) linux box and SER is
listening
on both IPv4 and IPv6 ports.
Now I want to try SER's connectivity on IPv6 and seeking IPv6 SIP
phone
(softphone).
Does anyone know IPv6 compatible softphone? (preferably free stuff)
--
Kuniyoshi Murata.........................iChat/AIM:macwebcaster
English-Japanese Interpreter mailto:kuni@ej-interpreter.net
Macintosh Webcast Specialist http://www.macwebcaster.com
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Is it G711 or G729 calls ?
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Adrian Georgescu
Sent: Monday, January 03, 2005 6:29 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] media proxy loading/performance issues
~ 70 calls on 1GHz CPU
~ 200 calls on 3 GHz CPU
There is a synthetic performance script in the MediaProxy distribution
which matched a real test bench so you can use that for an indication.
Adrian
>>>>>>>>>>>>>>
Hi eveyone,
I am gathering information about the performance of a media proxy.
Does anyone have statistics about how many calls and how long per phone
converstaion (in minutes) that a single media proxy handle?
thanks
proson
Hi
I am a newbie, I installed set with authentication digest. Now When I use to register VoIP device I get these messages.
First I was getting this message from SER
1d:0h:48m:43s ( lgr_stk_ses)(346 ) REGISTER sip:202.71.135.219 SIP/2.0
Via: SIP/2.0/UDP 202.71.135.197;branch=z9hG4bKacDvZhgcm
From: <sip:ritesh@sip.net4india.com>;tag=1c4430
To: <sip:ritesh@sip.net4india.com>
Call-ID: 19763147231472mCHM(a)202.71.135.197
CSeq: 8053 REGISTER
Expires: 180
Contact: <sip:ritesh@202.71.135.197;user=phone>;expires=180
Content-Length: 0
StackMngr: SIP Message arriving from IP: 202.71.135.219 ,
Message is : SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP 202.71.135.197;branch=z9hG4bKacDvZhgcm
From: <sip:ritesh@sip.net4india.com>;tag=1c4430
To: <sip:ritesh@sip.net4india.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.38c6
Call-ID: 19763147231472mCHM(a)202.71.135.197
CSeq: 8053 REGISTER
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 202.71.135.219:5060 "Noisy feedback
Then I Changed ser.cfg I commented these lines
#if (!mf_process_maxfwd_header("10")) {
# sl_send_reply("483","Too Many Hops");
# break;
#};
Then I started getting this error.
StackMngr: SIP Message arriving from IP: 202.71.135.219 , Message is :
1d:1h:12m:46s SIP/2.0 500 I'm terribly sorry, server error occured (2/TM)
Via: SIP/2.0/UDP 202.71.135.197;branch=z9hG4bKacGqEOdHT
From: <sip:ritesh@sip.net4india.com>;tag=1c64998
To: <sip:ritesh@sip.net4india.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8fa9
Call-ID: 217172319023190xROB(a)202.71.135.197
CSeq: 79267 REGISTER
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 39 StackMngr: SIP Message arriving from IP: 202.71.135.219 , Message is :
1d:1h:12m:50s SIP/2.0 500 I'm terribly sorry, server error occured (2/TM)
Via: SIP/2.0/UDP 202.71.135.197;branch=z9hG4bKacGqEOdHT
From: <sip:ritesh@sip.net4india.com>;tag=1c64998
To: <sip:ritesh@sip.net4india.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1711
Call-ID: 217172319023190xROB(a)202.71.135.197
CSeq: 79267 REGISTER
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 39my ser.cfg is attached. Pls. help
Thanks & Regards
Ritesh Jalan
Senior Engineer - Test & Audit
Net4india Ltd.
504, Bhikaji Cama Bhawan
11, Bhikaji Cama Place
New Delhi 110066
Tel: 91 (011) (26160129 - 131) (Extn 131)
URL: http://www.net4india.com
I have researched this list and all the docs I can find, but I am having a problem I can't solve.
I am using a Sipura registering to my SER server which is replicating register requests to another SIP Gatekeeper. We are using the other SIP gatekeeper for routing, and it appears that the registration request is being replicated (but then it is removed after a short timeout period). Anyway, the username on the SIP device and in ser is a 11 digit phone number.
When I dial the phone number from another sip device that is registered, I get 404 Not Found. I am including my debug for this call as well as parts of my ser.cfg. Can anyone point me in the right direction? Theoretically, the SER server should send any 11 digit phone number registered to it directly to the registered endpoint, otherwise, relay it to the other SIP router...
Thanks for the help...
198.67.41.194 Is the SIP Router
198.67.41.196 Is my SER server
phone is the user placing the call
12523152553 is the number being called....
-----------------------------------ser.cfg Registration and Routing----------------------------
if (uri=~"sip:1[0-9]+@.*") {
rewritehostport("198.67.41.194:5060");
forward(uri:host, uri:port);
};
if (method=="REGISTER") {
if (!proxy_authorize("sip.simflex.net" /* realm */,
"subscriber" /* table name */ )) {
proxy_challenge("sip.simflex.net", "0");
setflag(1);
break;
};
if (!src_ip==198.67.41.194) {
if(t_newtran()){
t_replicate("198.67.41.194", "5060");
};
};
save("location");
break;
};
-----------------------------------DEBUG----------------------------
SIP/2.0 183 Session Progress
Call-ID: EBBF58CE-320D-4906-ADD5-34B16E162D8C(a)10.0.0.2
From: "sip.simflex.net" <sip:phone@sip.simflex.net:8286>;tag=3089475581
To: <sip:12523152553@sip.simflex.net>
Content-Length: 0
CSeq: 25148 INVITE
Via: SIP/2.0/UDP 10.0.0.2:8286;received=198.67.41.187;rport=8286;received=198.67.41.187;rport=8286;branch=z9hG4bKABE3A5E79A9544F6A76997C242A1E16C
Record-Route: <sip:198.67.41.194:5060;lr>
0(27492) DEBUG: reply relayed. buf=0x80c47a8: SIP/2.0 1..., shmem=0xb576a588: SIP/2.0 1
0(27492) DBG: callback type 7, id 1 entered
0(27492) DEBUG: add_to_tail_of_timer[1]: 0xb5766de4
0(27492) DEBUG:destroy_avp_list: destroing list (nil)
0(27492) receive_msg: cleaning up
0(27492) SIP Reply (status):
0(27492) version: <SIP/2.0>
0(27492) status: <183>
0(27492) reason: <Session Progress>
0(27492) parse_headers: flags=1
0(27492) end of header reached, state=9
0(27492) DEBUG: get_hdr_field: <To> [35]; uri=[sip:12523152553@sip.simflex.net]
0(27492) DEBUG: to body [<sip:12523152553@sip.simflex.net>
]
0(27492) DEBUG: get_hdr_body : content_length=219
0(27492) get_hdr_field: cseq <CSeq>: <25148> <INVITE>
0(27492) Found param type 232, <branch> = <0>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=1
0(27492) parse_headers: this is the first via
0(27492) After parse_msg...
0(27492) forward_reply: found module tm, passing reply to it
0(27492) DEBUG: t_check: msg id=19 global id=18 T start=0xffffffff
0(27492) parse_headers: flags=17
0(27492) parse_headers: flags=4
0(27492) DEBUG: t_reply_matching: failure to match a transaction
0(27492) DEBUG: t_check: msg id=19 global id=19 T end=(nil)
0(27492) parse_headers: flags=2
0(27492) Found param type 232, <branch> = <z9hG4bK174a.53dcb714.0>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=2
0(27492) parse_headers: this is the second via
0(27492) old size: 784, new size: 743
0(27492) build_res_from_sip_res: copied size: orig:311, new: 270, rest: 473 msg=
SIP/2.0 183 Session Progress
Call-ID: EBBF58CE-320D-4906-ADD5-34B16E162D8C(a)10.0.0.2
From: "sip.simflex.net" <sip:phone@sip.simflex.net:8286>;tag=3089475581
To: <sip:12523152553@sip.simflex.net>
Content-Length: 219
Content-Type: application/sdp
CSeq: 25148 INVITE
Via: SIP/2.0/UDP 198.67.41.196;branch=z9hG4bK174a.53dcb714.0
Via: SIP/2.0/UDP 10.0.0.2:8286;received=198.67.41.187;rport=8286;received=198.67.41.187;rport=8286;branch=z9hG4bKABE3A5E79A9544F6A76997C242A1E16C
Record-Route: <sip:198.67.41.194:5060;lr>
v=0
o=MG4000|1.0 111 12345 IN IP4 198.67.41.194
s=-
c=IN IP4 198.67.41.194
t=0 0
m=audio 1695 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=ptime:10
0(27492) update_sock_struct_from_via: using via host
0(27492) update_sock_struct_from_via: trying SRV lookup
0(27492) reply forwarded to 198.67.41.196:0
0(27492) DEBUG:destroy_avp_list: destroing list (nil)
0(27492) receive_msg: cleaning up
0(27492) SIP Reply (status):
0(27492) version: <SIP/2.0>
0(27492) status: <183>
0(27492) reason: <Session Progress>
0(27492) parse_headers: flags=1
0(27492) end of header reached, state=9
0(27492) DEBUG: get_hdr_field: <To> [35]; uri=[sip:12523152553@sip.simflex.net]
0(27492) DEBUG: to body [<sip:12523152553@sip.simflex.net>
]
0(27492) DEBUG: get_hdr_body : content_length=219
0(27492) get_hdr_field: cseq <CSeq>: <25148> <INVITE>
0(27492) Found param type 232, <branch> = <z9hG4bK174a.53dcb714.0>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=1
0(27492) parse_headers: this is the first via
0(27492) After parse_msg...
0(27492) forward_reply: found module tm, passing reply to it
0(27492) DEBUG: t_check: msg id=20 global id=19 T start=(nil)
0(27492) parse_headers: flags=17
0(27492) parse_headers: flags=4
0(27492) DEBUG: t_reply_matching: hash 42097 label 1098632501 branch 0
0(27492) DEBUG: t_reply_matching: reply matched (T=0xb5766ca8)!
0(27492) DBG: callback type 1, id 5 entered
0(27492) DEBUG: t_check: msg id=20 global id=20 T end=0xb5766ca8
0(27492) DEBUG: t_on_reply: org. status uas=183, uac[0]=183 local=0 is_invite=1)
0(27492) ->>>>>>>>> T_code=183, new_code=183
0(27492) DEBUG: relay_reply: branch=0, save=0, relay=0
0(27492) old size: 743, new size: 681
0(27492) build_res_from_sip_res: copied size: orig:332, new: 270, rest: 411 msg=
SIP/2.0 183 Session Progress
Call-ID: EBBF58CE-320D-4906-ADD5-34B16E162D8C(a)10.0.0.2
From: "sip.simflex.net" <sip:phone@sip.simflex.net:8286>;tag=3089475581
To: <sip:12523152553@sip.simflex.net>
Content-Length: 219
Content-Type: application/sdp
CSeq: 25148 INVITE
Via: SIP/2.0/UDP 10.0.0.2:8286;received=198.67.41.187;rport=8286;received=198.67.41.187;rport=8286;branch=z9hG4bKABE3A5E79A9544F6A76997C242A1E16C
Record-Route: <sip:198.67.41.194:5060;lr>
v=0
o=MG4000|1.0 111 12345 IN IP4 198.67.41.194
s=-
c=IN IP4 198.67.41.194
t=0 0
m=audio 1695 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=ptime:10
0(27492) DEBUG: reply relayed. buf=0x80c5a68: SIP/2.0 1..., shmem=0xb576a7e0: SIP/2.0 1
0(27492) DBG: callback type 7, id 1 entered
0(27492) DEBUG: add_to_tail_of_timer[1]: 0xb5766de4
0(27492) DEBUG:destroy_avp_list: destroing list (nil)
0(27492) receive_msg: cleaning up
0(27492) SIP Reply (status):
0(27492) version: <SIP/2.0>
0(27492) status: <404>
0(27492) reason: <Not Found>
0(27492) parse_headers: flags=1
0(27492) end of header reached, state=9
0(27492) DEBUG: get_hdr_field: <To> [35]; uri=[sip:12523152553@sip.simflex.net]
0(27492) DEBUG: to body [<sip:12523152553@sip.simflex.net>
]
0(27492) DEBUG: get_hdr_body : content_length=0
0(27492) get_hdr_field: cseq <CSeq>: <25148> <INVITE>
0(27492) Found param type 232, <branch> = <0>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=1
0(27492) parse_headers: this is the first via
0(27492) After parse_msg...
0(27492) forward_reply: found module tm, passing reply to it
0(27492) DEBUG: t_check: msg id=21 global id=20 T start=0xffffffff
0(27492) parse_headers: flags=17
0(27492) parse_headers: flags=4
0(27492) DEBUG: t_reply_matching: failure to match a transaction
0(27492) DEBUG: t_check: msg id=21 global id=21 T end=(nil)
0(27492) parse_headers: flags=2
0(27492) Found param type 232, <branch> = <z9hG4bK174a.53dcb714.0>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=2
0(27492) parse_headers: this is the second via
0(27492) old size: 482, new size: 441
0(27492) build_res_from_sip_res: copied size: orig:271, new: 230, rest: 211 msg=
SIP/2.0 404 Not Found
Call-ID: EBBF58CE-320D-4906-ADD5-34B16E162D8C(a)10.0.0.2
From: "sip.simflex.net" <sip:phone@sip.simflex.net:8286>;tag=3089475581
To: <sip:12523152553@sip.simflex.net>
Content-Length: 0
CSeq: 25148 INVITE
Via: SIP/2.0/UDP 198.67.41.196;branch=z9hG4bK174a.53dcb714.0
Via: SIP/2.0/UDP 10.0.0.2:8286;received=198.67.41.187;rport=8286;received=198.67.41.187;rport=8286;branch=z9hG4bKABE3A5E79A9544F6A76997C242A1E16C
0(27492) update_sock_struct_from_via: using via host
0(27492) update_sock_struct_from_via: trying SRV lookup
0(27492) reply forwarded to 198.67.41.196:0
0(27492) DEBUG:destroy_avp_list: destroing list (nil)
0(27492) receive_msg: cleaning up
0(27492) SIP Reply (status):
0(27492) version: <SIP/2.0>
0(27492) status: <404>
0(27492) reason: <Not Found>
0(27492) parse_headers: flags=1
0(27492) end of header reached, state=9
0(27492) DEBUG: get_hdr_field: <To> [35]; uri=[sip:12523152553@sip.simflex.net]
0(27492) DEBUG: to body [<sip:12523152553@sip.simflex.net>
]
0(27492) DEBUG: get_hdr_body : content_length=0
0(27492) get_hdr_field: cseq <CSeq>: <25148> <INVITE>
0(27492) Found param type 232, <branch> = <z9hG4bK174a.53dcb714.0>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=1
0(27492) parse_headers: this is the first via
0(27492) After parse_msg...
0(27492) forward_reply: found module tm, passing reply to it
0(27492) DEBUG: t_check: msg id=22 global id=21 T start=(nil)
0(27492) parse_headers: flags=17
0(27492) parse_headers: flags=4
0(27492) DEBUG: t_reply_matching: hash 42097 label 1098632501 branch 0
0(27492) DEBUG: t_reply_matching: reply matched (T=0xb5766ca8)!
0(27492) DBG: callback type 1, id 5 entered
0(27492) DEBUG: t_check: msg id=22 global id=22 T end=0xb5766ca8
0(27492) DEBUG: t_on_reply: org. status uas=183, uac[0]=183 local=0 is_invite=1)
0(27492) parse_headers: flags=4
0(27492) ->>>>>>>>> T_code=183, new_code=404
0(27492) DBG: callback type 6, id 4 entered
0(27492) DEBUG: relay_reply: branch=0, save=0, relay=0
0(27492) old size: 441, new size: 379
0(27492) build_res_from_sip_res: copied size: orig:292, new: 230, rest: 149 msg=
SIP/2.0 404 Not Found
Call-ID: EBBF58CE-320D-4906-ADD5-34B16E162D8C(a)10.0.0.2
From: "sip.simflex.net" <sip:phone@sip.simflex.net:8286>;tag=3089475581
To: <sip:12523152553@sip.simflex.net>
Content-Length: 0
CSeq: 25148 INVITE
Via: SIP/2.0/UDP 10.0.0.2:8286;received=198.67.41.187;rport=8286;received=198.67.41.187;rport=8286;branch=z9hG4bKABE3A5E79A9544F6A76997C242A1E16C
0(27492) DEBUG: reply relayed. buf=0x80c47a8: SIP/2.0 4..., shmem=0xb57681b8: SIP/2.0 4
0(27492) DBG: callback type 7, id 1 entered
0(27492) DEBUG: cleanup_uacs: RETR/FR timers reset
0(27492) DEBUG: add_to_tail_of_timer[4]: 0xb5766d5c
0(27492) DEBUG: add_to_tail_of_timer[0]: 0xb5766d70
0(27492) DEBUG:destroy_avp_list: destroing list (nil)
0(27492) receive_msg: cleaning up
0(27492) SIP Request:
0(27492) method: <ACK>
0(27492) uri: <sip:12523152553@sip.simflex.net>
0(27492) version: <SIP/2.0>
0(27492) parse_headers: flags=1
0(27492) Found param type 235, <rport> = <n/a>; state=6
0(27492) Found param type 232, <branch> = <z9hG4bKABE3A5E79A9544F6A76997C242A1E16C>; state=16
0(27492) end of header reached, state=5
0(27492) parse_headers: Via found, flags=1
0(27492) parse_headers: this is the first via
0(27492) After parse_msg...
0(27492) preparing to run routing scripts...
0(27492) DEBUG : sl_filter_ACK: to late to be a local ACK!
0(27492) lookup(): '12523152553' Not found in usrloc
0(27492) DEBUG : is_maxfwd_present: searching for max_forwards header
0(27492) parse_headers: flags=128
0(27492) end of header reached, state=9
0(27492) DEBUG: get_hdr_field: <To> [35]; uri=[sip:12523152553@sip.simflex.net]
0(27492) DEBUG: to body [<sip:12523152553@sip.simflex.net>
]
0(27492) get_hdr_field: cseq <CSeq>: <25148> <ACK>
0(27492) DEBUG: is_maxfwd_present: value = 70
0(27492) DEBUG: add_param: tag=3089475581
0(27492) end of header reached, state=29
0(27492) parse_headers: flags=256
0(27492) DEBUG: get_hdr_body : content_length=0
0(27492) found end of header
0(27492) find_first_route(): No Route headers found
0(27492) loose_route(): There is no Route HF
0(27492) DEBUG: mk_proxy: doing DNS lookup...
0(27492) check_via_address(198.67.41.187, 10.0.0.2, 0)
0(27492) Sending:
Your write, I just was confused because the PCphoneline.com guys told me
that this was a Wireless VoIP solution...
Now I understood that they mean a Wireless solution by connecting to
wireless laptops..... duh
Being the VPT1000 out of option, can someone tell me about a good WiFi SIP
endpoint? I would like one that supports authentication with 802.1x... if
these SIP phones exist...
Thanks
Joao Pereira
----- Original Message -----
From: "Karl H. Putz" <kputz(a)columbus.rr.com>
To: "Joao Pereira" <joao.pereira(a)fccn.pt>
Sent: Monday, January 03, 2005 5:36 PM
Subject: RE: [Serusers] SIP endpoint VPT1000
Look more closely at the details on this phone. It is a wired handset
connected via USB to another computer.
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On
Behalf Of Joao Pereira
Sent: Monday, January 03, 2005 12:29 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] SIP endpoint VPT1000
Hello to all
I would like to know if someone tried the VPT1000 WiFi phone from:
http://pcphoneline.com/
I think the phone is cheap... but I wanted to know the opinion of someone
that had tried it.
Thanks
João Pereira
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hello to all
I would like to know if someone tried the VPT1000 WiFi phone from:
http://pcphoneline.com/
I think the phone is cheap... but I wanted to know the opinion of someone
that had tried it.
Thanks
João Pereira
Hi,
Thanks for the reply! I did however just get 407 instead of 401.
But...I found the fault! :-)
In for example the client program X-Lite you specify 'Domain/Realm' in the menu (i.e. one field), were I've specified the SIP servers IP address. But when the user was created (with serctl) the SIP_DOMAIN variable was set to something else. So when I deleted the user, sat the variable to the servers IP address and recreated the user, the auth register msg went thru.
Regards,
//Magnus
-----Original Message-----
From: innovation.interops(a)wipro.com [mailto:innovation.interops@wipro.com]
Sent: den 3 januari 2005 08:16
To: Magnus Sörman (AL/EAB); serusers(a)lists.iptel.org
Subject: RE: [Serusers] Digest Authentication
Hello,
Try these inclusions pls for a simple straight forward Digest Auth...
modparam("auth_db", "db_url","sql://ser:heslo@localhost/ser")
# main routing logic
route{
if(!proxy_authorize("yourdomain.com" /* realm */,
"subscriber" /* table name */ ))
{
proxy_challenge("yourdomain.com", "0");
break;
}
sl_send_reply("200", "ok");
karthikeyan.k
_____
From: serusers-bounces(a)lists.iptel.org on behalf of Magnus Sörman (AL/EAB)
Sent: Thu 12/30/2004 3:45 PM
To: 'serusers(a)lists.iptel.org'
Subject: [Serusers] Digest Authentication
Hi,
I need some help with digest authentication.
When I uncomment those lines in ser.cfg, the register msg stops to work. In the trace, see below, you can see the nonce being sent in the re-register msg, but the server still responds with 401 Unauthorized. I've tried with both 0 and 1 in the www_challenge.
Without the digest authentication the register works fine.
Thanks in advance,
//Magnus
ser.cfg (ser 0.8.12 running on a Fedora box. Used for test purpose only):
====================================================
# ----------- 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
fifo="/tmp/ser_fifo"
sip_warning=no
alias="sip_server_ip"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
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/pa.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
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)
# ------------------------- 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=="SUBSCRIBE") {
if(t_newtran()){
handle_subscription("registrar");
break;
};
};
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("sip_server_ip", "subscriber")) {
www_challenge("sip_server_ip", "1");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
Register trace:
==========
REGISTER sip:sip_server_ip SIP/2.0
Via: SIP/2.0/UDP local_pc_ip:5060;rport;branch=z9hG4bK4268DFDFE5EE410C8DB113A6223C800C
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>
Contact: "Magnus" <sip:magnus@local_pc_ip:5060>
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6590 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP local_pc_ip:5060;rport=5060;branch=z9hG4bK4268DFDFE5EE410C8DB113A6223C800C
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0d0e
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6590 REGISTER
WWW-Authenticate: Digest realm="sip_server_ip", nonce="41d1321431d402c1af9617eb73deccbce7e532d5", qop="auth"
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
REGISTER sip:sip_server_ip SIP/2.0
Via: SIP/2.0/UDP local_pc_ip:5060;rport;branch=z9hG4bK1813C486770C442BB51E58686A61921F
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>
Contact: "Magnus" <sip:magnus@local_pc_ip:5060>
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6591 REGISTER
Expires: 1800
Authorization: Digest username="magnus",realm="sip_server_ip",nonce="41d1321431d402c1af9617eb73deccbce7e532d5",response="27ea80aed1b9f5086b396c8f86bcec60",uri="sip:sip_server_ip",qop=auth,cnonce="9F5BBA98D6724D909C6560E8A045A300",nc=00000006
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP local_pc_ip:5060;rport=5060;branch=z9hG4bK1813C486770C442BB51E58686A61921F
From: Magnus <sip:magnus@sip_server_ip>;tag=470300110
To: Magnus <sip:magnus@sip_server_ip>;tag=b27e1a1d33761e85846fc98f5f3a7e58.9cf2
Call-ID: EB7272E371C24F6C8F24DB47A53EE7CB@sip_server_ip
CSeq: 6591 REGISTER
WWW-Authenticate: Digest realm="sip_server_ip", nonce="41d1321431d402c1af9617eb73deccbce7e532d5", qop="auth"
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
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Hi all,
i've solved my problem
(http://developer.berlios.de/bugs/?func=detailbug&bug_id=2979&group_id=1208)
upgrading radiusclient to 0.4.8
(http://download.berlios.de/radiusclient-ng/radiusclient-0.4.8.tar.gz).
Have a nice 2005 !!!
Oz
Detail of BUG:
Program received signal SIGSEGV, Segmentation fault.
0xb5e244e0 in set_option_srv (filename=0xb5e34320
"/usr/local/etc/radiusclient/radiusclient.conf", line=37,
option=0x81d0400, p=0xbffff35c "localhost") at config.c:126
126 serv->port[serv->max] = ntohs ((unsigned int) svp->s_port);
(gdb) backtrace
#0 0xb5e244e0 in set_option_srv (filename=0xb5e34320
"/usr/local/etc/radiusclient/radiusclient.conf", line=37,
option=0x81d0400, p=0xbffff35c "localhost") at config.c:126
#1 0xb5e2499b in rc_read_config (filename=0xb5e34320
"/usr/local/etc/radiusclient/radiusclient.conf") at config.c:288
#2 0xb5e3283c in mod_init () at authrad_mod.c:135
#3 0x0806e8a7 in init_mod (m=0x80c90d8) at sr_module.c:420
#4 0x0806ea24 in init_modules () at sr_module.c:443
#5 0x0805af87 in main (argc=1, argv=0xbffff854) at main.c:1759
(gdb)
--
------
O-Zone ! www.zerozone.it
I'm not sure if this is a valid statement but it worked for me, I had to
make my log_* flags different numbers in order to catch them both.
Remember, log_level is to logger and log_flag is to your db.
-----Original Message-----
From: Alexander [mailto:alexander_gav@yahoo.com]
Sent: Sunday, January 02, 2005 4:22 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Some acc module problems.
I have ser version version: 0.8.14 (i386/linux)
I have compiled acc module with support for SQL.
I am trying to setup the ser.cfg to write information to the
mysql database to acc table (with no success).
modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1 )
setflag(1);
I have those settings in my cfg file, I can register users, and
i see them in the location table.
I can make local calls, but i don't see anything in the acc
table.
Any help will be more than welcome.
_____
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