Fred,
I actually messed up my original post. I stated that I am placing my NOTIFY
messages directly in the ser FIFO. In reality I should have said I'm using
sipsak to send the message to ser just like any normal UA would.
In my asterisk file sip.conf I have the following:
[general]
register => vmserver:number4@sip01.mycompany.com/vmserver
So asterisk is a registered UA with ser. Doing this I assumed I could simply
use sipsak from the asterisk server to send NOTIFY messages to the ser box.
I can infact do this, becuase when I do so the "TO:" SIP UA crashes. That tells
me that everything is working, but my message is not 100% correct.
I also have the user extension configured in ser and asterisk and all is
working because I can leave voice mail and listen to voice mail normally. It's
just the MWI that is giving me grief.
Paul
--- Fred Tips <fred(a)callcarrera.com> wrote:
> Do you have mailbox= in * under the sip peer ?
>
> ----- Original Message -----
> From: "Java Rockx" <javarockx(a)yahoo.com>
> To: <serusers(a)lists.iptel.org>
> Sent: Friday, September 17, 2004 3:14 PM
> Subject: [Serusers] Injecting SIP NOTIFY into ser FIFO
>
>
> > Hello everyone.
> >
> > Can anyone tell me what I'm doing wrong? I am using Asterisk as a voice
> mail
> > server for my ser SIP proxy. Everything works fine expect the Message
> Waiting
> > Indicator (MWI). I've read a few articles and got some good feed back from
> this
> > mailing list but here is my problem.
> >
> > My NOTIFY message is crashing my SIP phone. Here is my setup. I'm using
> > "externnotify" in Asterisk to call a /usr/bin/mwi script that I wrote. All
> this
> > script does is create file in /var/spool/mwi that contains the
> externnotify
> > information.
> >
> > Then cron periodically processes the messages in /var/spool/mwi. By
> process I
> > mean it will create the SIP NOTIFY messages and place them directly in to
> the
> > ser FIFO. The problem is when this happens the SIP phone crashes.
> >
> > Here is a sample of my NOTIFY message that my Asterisk server cron job is
> > sending to the ser FIFO. I wish there was a way to just have Asterisk
> register
> > with the ser proxy and handle this automatically.
> >
> > Can anyone see a problem?
> > Regards,
> > Paul
> >
> > --------------------------------------------------
> >
> > NOTIFY sip:1002@sip01.mycompany.com SIP/2.0
> > Via: SIP/2.0/UDP 4.4.242.201:5060
> > From: "vmserver" <sip:vmserver@sip01.mycompany.com>
> > To: <sip:1002@sip01.mycomany.com>
> > Contact: <sip:vmserver@sip01.mycompany.com>
> > Call-ID: 5b8be1521efe68b5365a36466ad3b87(a)4.4.242.201
> > CSeq: 1101 NOTIFY
> > User-Agent: VoiceMail
> > Event: message-summary
> > Content-Type: application/simple-message-summary
> > Content-Length: 38
> >
> >
> > Messages-Waiting: yes
> > Voicemail: 13/0
> >
>
> >
> >
> >
> >
> >
> > _______________________________
> > Do you Yahoo!?
> > Declare Yourself - Register online to vote today!
> > http://vote.yahoo.com
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
__________________________________
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Hi
I have set:
echo "uname: ".$uname."<br>\n";
echo "passw: ".$passw."<br>\n";
echo "okey_x: ".$okey_x."<br>\n";
and the result was :
uname: admin
passw: heslo
okey_x: 0
and in your document you have wriiten that okey_x: need to be anything but 0
can you tell me what do I need to do to change this?
Because I am always getting bad username or password!
Shiran Guez
Hello everyone.
Can anyone tell me what I'm doing wrong? I am using Asterisk as a voice mail
server for my ser SIP proxy. Everything works fine expect the Message Waiting
Indicator (MWI). I've read a few articles and got some good feed back from this
mailing list but here is my problem.
My NOTIFY message is crashing my SIP phone. Here is my setup. I'm using
"externnotify" in Asterisk to call a /usr/bin/mwi script that I wrote. All this
script does is create file in /var/spool/mwi that contains the externnotify
information.
Then cron periodically processes the messages in /var/spool/mwi. By process I
mean it will create the SIP NOTIFY messages and place them directly in to the
ser FIFO. The problem is when this happens the SIP phone crashes.
Here is a sample of my NOTIFY message that my Asterisk server cron job is
sending to the ser FIFO. I wish there was a way to just have Asterisk register
with the ser proxy and handle this automatically.
Can anyone see a problem?
Regards,
Paul
--------------------------------------------------
NOTIFY sip:1002@sip01.mycompany.com SIP/2.0
Via: SIP/2.0/UDP 4.4.242.201:5060
From: "vmserver" <sip:vmserver@sip01.mycompany.com>
To: <sip:1002@sip01.mycomany.com>
Contact: <sip:vmserver@sip01.mycompany.com>
Call-ID: 5b8be1521efe68b5365a36466ad3b87(a)4.4.242.201
CSeq: 1101 NOTIFY
User-Agent: VoiceMail
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 38
Messages-Waiting: yes
Voicemail: 13/0
_______________________________
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I am using SER 0.8.14, FreeRadius 1.0.0 and the latest version of
radiusclient. When I try to start SER, I get
"ERROR: auth_radius: cant get code for the Sip-Session attribute value"
I found some posts about this issue, but the replies did not help me.
My dictionary file includes the standard values, sip, and cisco
values. Is this just a line I need to add to my dictionary file as an
avp? If so, what is the correct avp for sip-session, I cant find it
anywhere. Thank you.
Hi..I hust installed sereweb..everything is fine..i can connect to the
admin page but it seems that it needs a username and a password ...the
problem is that I don't know the username and password to login..how can I
find them ?
Hi..serweb works fine..I can see the login page but I can\t login..I tried
with admin and password: heslo but it doesn't work..i mention that I
changed the ser database password but it doesn't work with that too...Can
anyone help me ?..or can someone tell me how to change the adminuser or
password ?,..is there a cfg file ?
|Thanks
I have radiusclient-0.4.3 installed .What version is the right one for the
GSM module to compile? Should I use the previous versions ?
Thanks again
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:daniel@iptel.org]
Sent: Friday, August 27, 2004 1:54 PM
To: aimable
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Problems compiling GSM authentication module
You are using a wrong version of radiusclient library. What is yours?
When I will find some time to spare I will update it to use
radiusclient-ng library, to be the same as latest ser.
Daniel
On 8/27/2004 12:56 PM, aimable wrote:
> Hi all,
>
> I have been using SER for 6 months (together with Asterisk ) and every
> thing seems to be fine ( voicemail, accounting, authentication, NAT
> and so on)
>
> But I recently wanted to test the GSM authentication module but when I
> tried to compile it and install it I failed .I am using version: ser
> 0.8.12-tcp_nonb (i386/linux) of SER on RH 9
>
> Here is the error I get when I try to compile gsm module
>
> auth_gsm.c:117: warning: passing arg 1 of `rc_conf_str' from
> incompatible pointer type
>
> auth_gsm.c:117: too few arguments to function `rc_conf_str'
>
> auth_gsm.c:117: warning: passing arg 1 of `rc_read_dictionary' from
> incompatible pointer type
>
> auth_gsm.c:117: too few arguments to function `rc_read_dictionary'
>
> make: *** [auth_gsm.o] Error 1
>
>
>
> can anyone help me about this?
>
> Thanks
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
>
Can anyone advise me on how to get the Message Waiting Indicator in Asterisk to
notify phones that voice mail is waiting?
I'm posting to this mail list because I'm using ser as my SIP proxy and
Asterisk as a voice mail server. I'm not using any other Asterisk functionality
other than voice mail.
I assume that asterisk would have to send the SIP NOTIFY message to the
registered SIP clients via ser. So will I have to create a custom Asterisk and
ser modules to get this to work?
Regards,
Paul
__________________________________
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Hello,
my configuration follows:
SIPclientA <--> SER(0.8.14) <--> SIPproxy <--> SIPclientB
SER performs authorization and forwards all REGISTER requets to SIPproxy.
Scenario:
SIPclientA calls to SIPclientB, then changes his mind and releases call.
SIPclientA generates CANCEL, but this CANCEL isn't forwarded to SIPproxy
by SER and SIPproxy continues its conversation with SIPclientB by
forwarding "180 Ringing" to SIPclientA.
This isn't the case if SIPclientA receives "180 Ringing" by the time it
sends CANCEL.
What client gets in first case is:
t_reply( t_cancel, cancel_msg, 200, "ok, no more pending branches" );
and to my understanding is should be:
t_reply( t_cancel, cancel_msg, 100, "trying to cancel" );
ngrep looks like:
U 82.135.142.16:5060 -> 213.226.186.195:5060
CANCEL sip:70000037@213.226.186.195 SIP/2.0..
From: <sip:70000031@213.226.186.195>;tag=0-13c4-4148435c-1f825298-792d..
To: <sip:70000037@213.226.186.195>..
Call-ID: cc4698-0-13c4-4148435c-1f825298-157@213.226.186.195..CSeq: 2
CANCEL..Via: SIP/2.0/UDP 82.135.142.16:5060;branch=z9hG4bK-4148435c-1f82552c-60b0..
Max-Forwards: 70..
Supported: 100rel..
User-Agent: SIP-RG..
Authorization: Digest username="70000031", realm="213.226.186.195",
nonce="414844890f0b1edbefcbd
badce4c02ed59f058ef", uri="sip:70000037@213.226.186.195",
response="5d22fe8fe3517aa72a51b8abc83bd9f2",
algorithm=MD5..Content-Length: 0....
#
U 213.226.186.195:5060 -> 82.135.142.16:5060
SIP/2.0 200 ok -- no more pending branches..
From: <sip:70000031@213.226.186.195>;tag=0-13c4-4148435c-1f825298-792d..
To: <sip:70000037@213.226.186.195>;tag=855fc458e92794ef4dd58c58d38d34d1-0196..Call-ID:
cc4698-0-13c4-4148435c-1f825298-157@213.226.186.195..CSeq: 2 CANCEL..
Via: SIP/2.0/UDP 82.135.142.16:5060;branch=z9hG4bK-4148435c-1f82552c-60b0..Server: Sip EXpress router
(0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
213.226.186.250:5060 "Noisy feedback tells: pid=17371 req_src_ip=82.135.142.16 req_src_port=5060
in_uri=sip:70000037@213.226.186.195 out_uri=sip:70000037@213.226.186.195
via_cnt==1".
Somehow SER doesn't detect in this case that it already forwarded INVITE
and that is needs to be CANCEL'ed.
Here i provide logs for both cases. If more logs are needed, tell me.
What could be the reason of such behaviour?
SER handling of CANCEL in case of proper termination:
ser[17364]: method: <CANCEL>
ser[17364]: uri: <sip:70000037@213.226.186.195>
ser[17364]: version: <SIP/2.0>
ser[17364]: parse_headers: flags=1
ser[17364]: end of header reached, state=9
ser[17364]: DEBUG: get_hdr_field: <To> [32];
uri=[sip:70000037@213.226.186.195]
ser[17364]: DEBUG: to body [<sip:70000037@213.226.186.195>^M ]
ser[17364]: get_hdr_field: cseq <CSeq>: <2> <CANCEL>
ser[17364]: Found param type 232, <branch> =
<z9hG4bK-41483835-1f56c420-429>; state=16
ser[17364]: end of header reached, state=5
ser[17364]: parse_headers: Via found, flags=1
ser[17364]: parse_headers: this is the first via
ser[17364]: After parse_msg...
ser[17364]: preparing to run routing scripts...
ser[17364]: DEBUG: add_param: tag=0-13c4-41483835-1f56c394-362
ser[17364]: end of header reached, state=29
ser[17364]: parse_headers: flags=256
ser[17364]: DEBUG: get_hdr_body : content_length=0
ser[17364]: found end of header
ser[17364]: find_first_route(): No Route headers found
ser[17364]: loose_route(): There is no Route HF
ser[17364]: DEBUG: t_addifnew: msg id=14502 , global msg id=14501 , T on
entrance=0xffffffff
ser[17364]: parse_headers: flags=-1
ser[17364]: parse_headers: flags=60
ser[17364]: t_lookup_request: start searching: hash=21608, isACK=0
ser[17364]: DEBUG: RFC3261 transaction matching failed
ser[17364]: DEBUG: t_lookup_request: no transaction found
ser[17364]: DEBUG: t_lookupOriginalT: searching on hash entry 21608
ser[17364]: DEBUG: RFC3261 transaction matched, tid=-41483835-1f56c420-429
ser[17364]: DEBUG: t_lookupOriginalT: canceled transaction found
(0x402ea680)!
ser[17364]: DEBUG: t_lookupOriginalT completed
ser[17364]: check_via_address(82.135.142.16, 82.135.142.16, 0)
ser[17364]: DEBUG: add_to_tail_of_timer[4]: 0x402ed988
ser[17364]: DEBUG: add_to_tail_of_timer[0]: 0x402ed99c
ser[17364]: DEBUG: e2e_cancel: e2e cancel proceeding
ser[17364]: parse_headers: flags=-1
ser[17364]: check_via_address(82.135.142.16, 82.135.142.16, 0)
ser[17364]: WARNING:vqm_resize: resize(0) called
ser[17364]: DEBUG: cleanup_uacs: RETR/FR timers reset
ser[17364]: DEBUG: add_to_tail_of_timer[2]: 0x402ed8a8
ser[17364]: DEBUG: reply sent out. buf=0x80e07f0: SIP/2.0 2...,
shmem=0x402cc9b8: SIP/2.0 2
ser[17364]: DEBUG: t_reply: finished
ser[17364]: DEBUG: e2e_cancel: sending 487
ser[17364]: parse_headers: flags=-1
ser[17364]: check_via_address(82.135.142.16, 82.135.142.16, 0)
ser[17364]: DEBUG: cleanup_uacs: RETR/FR timers reset
ser[17364]: DEBUG: add_to_tail_of_timer[4]: 0x402ea734
ser[17364]: DEBUG: add_to_tail_of_timer[0]: 0x402ea748
ser[17364]: DEBUG: reply sent out. buf=0x80e07f0: SIP/2.0 4...,
shmem=0x402d14b8: SIP/2.0 4
ser[17364]: DEBUG: t_reply: finished
ser[17364]: SER: new transaction fwd'ed
ser[17364]: DEBUG:destroy_avp_list: destroing list (nil)
ser[17364]: receive_msg: cleaning up
SER handling of CANCEL in case of inproper termination:
ser[17359]: SIP Request:
ser[17359]: method: <CANCEL>
ser[17359]: uri: <sip:70000037@213.226.186.195>
ser[17359]: version: <SIP/2.0>
ser[17359]: parse_headers: flags=1
ser[17359]: end of header reached, state=9
ser[17359]: DEBUG: get_hdr_field: <To> [32];
uri=[sip:70000037@213.226.186.195]
ser[17359]: DEBUG: to body [<sip:70000037@213.226.186.195>^M ]
ser[17359]: get_hdr_field: cseq <CSeq>: <2> <CANCEL>
ser[17359]: Found param type 232, <branch> =
<z9hG4bK-414837ae-1f54b338-47f1>; state=16
ser[17359]: end of header reached, state=5
ser[17359]: parse_headers: Via found, flags=1
ser[17359]: parse_headers: this is the first via
ser[17359]: After parse_msg...
ser[17359]: preparing to run routing scripts...
ser[17359]: DEBUG: add_param: tag=0-13c4-414837ad-1f54b1bc-5d0a
ser[17359]: end of header reached, state=29
ser[17359]: parse_headers: flags=256
ser[17359]: DEBUG: get_hdr_body : content_length=0
ser[17359]: found end of header
ser[17359]: find_first_route(): No Route headers found
ser[17359]: loose_route(): There is no Route HF
ser[17359]: DEBUG: t_addifnew: msg id=14611 , global msg id=14608 , T on
entrance=(nil)
ser[17359]: parse_headers: flags=-1
ser[17359]: parse_headers: flags=60
ser[17359]: t_lookup_request: start searching: hash=36669, isACK=0
ser[17359]: DEBUG: RFC3261 transaction matching failed
ser[17359]: DEBUG: t_lookup_request: no transaction found
ser[17359]: DEBUG: t_lookupOriginalT: searching on hash entry 36669
ser[17359]: DEBUG: RFC3261 transaction matched,
tid=-414837ae-1f54b338-47f1
ser[17359]: DEBUG: t_lookupOriginalT: canceled transaction found
(0x402ea680)!
ser[17359]: DEBUG: t_lookupOriginalT completed
ser[17359]: DEBUG: e2e_cancel: e2e cancel -- no more pending branches
ser[17359]: parse_headers: flags=-1
ser[17359]: check_via_address(82.135.142.16, 82.135.142.16, 0)
ser[17359]: WARNING:vqm_resize: resize(0) called
ser[17359]: DEBUG: cleanup_uacs: RETR/FR timers reset
ser[17359]: DEBUG: add_to_tail_of_timer[2]: 0x402e4f98
ser[17359]: DEBUG: reply sent out. buf=0x80e1540: SIP/2.0 2...,
shmem=0x402d0828: SIP/2.0 2
ser[17359]: DEBUG: t_reply: finished
ser[17359]: DEBUG: e2e_cancel: sending 487
ser[17359]: parse_headers: flags=-1
ser[17359]: check_via_address(82.135.142.16, 82.135.142.16, 0)
ser[17359]: DEBUG: cleanup_uacs: RETR/FR timers reset
ser[17359]: DEBUG: add_to_tail_of_timer[4]: 0x402ea734
ser[17359]: DEBUG: add_to_tail_of_timer[0]: 0x402ea748
ser[17359]: DEBUG: reply sent out. buf=0x80e0c90: SIP/2.0 4...,
shmem=0x402dada0: SIP/2.0 4
ser[17359]: DEBUG: t_reply: finished
ser[17359]: SER: new transaction fwd'ed
ser[17359]: DEBUG:destroy_avp_list: destroing list (nil)
Antanas
