fifo_server.c:98: `DEFAULT_FIFO_DIR' undeclared here (not in a function)
fifo_server.c: In function `init_fifo_server':
fifo_server.c:571: `sock_mode' undeclared (first use in this function)
fifo_server.c:571: (Each undeclared identifier is reported only once
fifo_server.c:571: for each function it appears in.)
fifo_server.c:584: `sock_uid' undeclared (first use in this function)
fifo_server.c:584: `sock_gid' undeclared (first use in this function)
Shiran Guez
Hi Java,
Presumably your users are registered with SER? And you have some
specific routing for when they check their mailbox? Either forwarding a
short code to VoiceMailMain or forwarding a variation of their extension
or something? How about simply checking for the "BYE" method in this
routing block and using the exec module in SER to force a recheck of the
script on the asterisk box?
D
-----Original Message-----
From: Java Rockx [mailto:javarockx@yahoo.com]
Sent: 19 September 2004 05:56
To: serusers(a)lists.iptel.org; Dave Bath
Subject: RE: [Serusers] Injecting SIP NOTIFY into ser FIFO
Dave,
I cheated big time. I use sipsak to send the message from my Asterisk
server to
my ser proxy. I realized to late that I mentioned FIFO in my original
post.
I use the Asterisk "externnotify" in the voicemail.conf file to specify
an
external bash script. This script then simply "touches" a file in
/var/spool/mwi/ for later processing by cron.
My cron job comes along after the fact and picks up all the files in
/var/spool/mwi/ and generates the appropriate SIP NOTIFY message and
then it
uses sipsak to "shoot" it to the ser proxy.
This works really well for me when I need to turn on the MWI for a UA.
What I
still can't do is turn the MWI off when there are no new messages. I'm
really
having a hard time trying to determine how to hook the "Hang-Up" event
in
Asterisk. I need to essentially have an "externnotify" for hang ups so I
can
check the message status for a the mailbox and cancel the MWI if there
are no
new messages.
I also need to handle SUBSCRIBE messages that the SIP proxy gets. I'm
planning
on have the ser.cfg file execute an external script that will get a
message to
the cron job on my Asterisk box. Once the Asterisk box gets notified of
the
SUBSCRIBE message, it will process the same as if the user just had a
voice
mail left and the externnotify event got fired.
Paul
--- Dave Bath <dave(a)fuuz.com> wrote:
> Hey Java,
>
> I know it's cheeky, but I was wondering if you might let me see the
> script you wrote to create the mwi message? Great piece of work!
>
> Dave
>
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
On
> Behalf Of Java Rockx
> Sent: 18 September 2004 03:49
> To: Java Rockx; serusers(a)lists.iptel.org; Fred Tips
> Subject: Re: [Serusers] Injecting SIP NOTIFY into ser FIFO
>
> I found my problem. I had to convert the NOTIFY message from Unix
format
> to
> Windoz format. I used the unix2dos command to do this.
>
> Everything is good now.
>
> P
>
> --- Java Rockx <javarockx(a)yahoo.com> wrote:
>
> > 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
> > >
> > >
> >
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > Yahoo! Mail - 50x more storage than other providers!
> > http://promotions.yahoo.com/new_mail
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
>
> __________________________________
> Do you Yahoo!?
> New and Improved Yahoo! Mail - Send 10MB messages!
> http://promotions.yahoo.com/new_mail
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
>
>
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - Send 10MB messages!
http://promotions.yahoo.com/new_mail
Can Someone explain the following:
if (len_gt( max_len )) {
sl_send_reply("513", "Wow -- Message too large");
break;
I always get error on this lines!
Shiran Guez
Can Someone help on the ACC module:
acc.c: In function `acc_db_bind':
acc.c:445: too many arguments to function `bind_dbmod'
acc.c:451: warning: implicit declaration of function `DB_CAPABILITY'
acc.c:451: `DB_CAP_INSERT' undeclared (first use in this function)
acc.c:451: (Each undeclared identifier is reported only once
acc.c:451: for each function it appears in.)
make: *** [acc.o] Error 1
Shiran Guez
Dave,
I cheated big time. I use sipsak to send the message from my Asterisk server to
my ser proxy. I realized to late that I mentioned FIFO in my original post.
I use the Asterisk "externnotify" in the voicemail.conf file to specify an
external bash script. This script then simply "touches" a file in
/var/spool/mwi/ for later processing by cron.
My cron job comes along after the fact and picks up all the files in
/var/spool/mwi/ and generates the appropriate SIP NOTIFY message and then it
uses sipsak to "shoot" it to the ser proxy.
This works really well for me when I need to turn on the MWI for a UA. What I
still can't do is turn the MWI off when there are no new messages. I'm really
having a hard time trying to determine how to hook the "Hang-Up" event in
Asterisk. I need to essentially have an "externnotify" for hang ups so I can
check the message status for a the mailbox and cancel the MWI if there are no
new messages.
I also need to handle SUBSCRIBE messages that the SIP proxy gets. I'm planning
on have the ser.cfg file execute an external script that will get a message to
the cron job on my Asterisk box. Once the Asterisk box gets notified of the
SUBSCRIBE message, it will process the same as if the user just had a voice
mail left and the externnotify event got fired.
Paul
--- Dave Bath <dave(a)fuuz.com> wrote:
> Hey Java,
>
> I know it's cheeky, but I was wondering if you might let me see the
> script you wrote to create the mwi message? Great piece of work!
>
> Dave
>
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
> Behalf Of Java Rockx
> Sent: 18 September 2004 03:49
> To: Java Rockx; serusers(a)lists.iptel.org; Fred Tips
> Subject: Re: [Serusers] Injecting SIP NOTIFY into ser FIFO
>
> I found my problem. I had to convert the NOTIFY message from Unix format
> to
> Windoz format. I used the unix2dos command to do this.
>
> Everything is good now.
>
> P
>
> --- Java Rockx <javarockx(a)yahoo.com> wrote:
>
> > 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
> > >
> > >
> >
> >
> >
> >
> > __________________________________
> > Do you Yahoo!?
> > Yahoo! Mail - 50x more storage than other providers!
> > http://promotions.yahoo.com/new_mail
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
>
> __________________________________
> Do you Yahoo!?
> New and Improved Yahoo! Mail - Send 10MB messages!
> http://promotions.yahoo.com/new_mail
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
>
>
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - Send 10MB messages!
http://promotions.yahoo.com/new_mail
Could someone please help me with this? I am running SER 0.8.14 and
Freeradius 1.0.0. They are on sperate servers, but I can use
radiusclient on the SER box and succesfully authenticate a test
account on the remote radius box. When my sipura spa-200 trys to
register to SER, SER does not appear to be making a request to radius.
I have followed the ser-radius how-to, and still no good. Below are my
configs and debug. Thank you all for the help that you have given me
in the past and hopefully someone can help with this question.
Config
# ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd)
#fork=yes
log_stderror=yes # (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"
alias=****************
alias=*****************
# ------------------ module loading ----------------------------------
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"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
loadmodule "/usr/local/lib/ser/modules/uri_radius.so"
loadmodule "/usr/local/lib/ser/modules/group_radius.so"
loadmodule "/usr/local/lib/ser/modules/pa.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "sql://ser:*********@***********/ser")
# -- auth radius params --
modparam("auth_radius", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_radius", "service_type", 15)
# -- URI radius params --
modparam("uri_radius", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("uri_radius", "service_type", 10)
# -- Group radius params --
modparam("group_radius", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("group_radius", "use_domain", 0)
# -- Presence params --
modparam("pa", "default_expires", 3600)
# -- Nathelper params --
modparam("nathelper", "natping_interval", 10)
# -- Msilo params --
modparam("msilo", "db_url", "sql://ser:********@*********/ser")
modparam("msilo", "db_table", "silo")
modparam("msilo", "expire_time", 36000)
modparam("msilo", "check_time", 20)
modparam("msilo", "clean_period", 3)
modparam("msilo", "use_contact", 1)
# -- 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=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!radius_www_authorize("")) {
www_challenge("", "0"); # I have also
tried 1 in place of 0 #
};
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();
};
}
Debug from SER console
5(16293) uri: <sip:192.168.1.119>
5(16293) version: <SIP/2.0>
5(16293) parse_headers: flags=1
5(16293) Found param type 232, <branch> = <z9hG4bK-4d418a32>; state=16
5(16293) end of header reached, state=5
5(16293) parse_headers: Via found, flags=1
5(16293) parse_headers: this is the first via
5(16293) After parse_msg...
5(16293) preparing to run routing scripts...
5(16293) DEBUG : is_maxfwd_present: searching for max_forwards header
5(16293) parse_headers: flags=128
5(16293) end of header reached, state=9
5(16293) DEBUG: get_hdr_field: <To> [31]; uri=[sip:test@192.168.1.119]
5(16293) DEBUG: to body [test <sip:test@192.168.1.119>
]
5(16293) get_hdr_field: cseq <CSeq>: <169> <REGISTER>
5(16293) DEBUG: is_maxfwd_present: value = 70
5(16293) DEBUG: add_param: tag=79b50153b98e6976
5(16293) end of header reached, state=29
5(16293) parse_headers: flags=256
5(16293) DEBUG: get_hdr_body : content_length=0
5(16293) found end of header
5(16293) find_first_route(): No Route headers found
5(16293) loose_route(): There is no Route HF
5(16293) check_nonce(): comparing
[414c8e3fd21e4da1a441587f79ce539efb32846a] and
[414c8e3f34ca12a5043f59618fcadba81be75c2e]
5(16293) pre_auth(): Invalid nonce value received
5(16293) build_auth_hf(): 'WWW-Authenticate: Digest
realm="192.168.1.119",
nonce="414c90ab7f933a6b3c06a4bbbca22ce39fbf8012"
'
5(16293) parse_headers: flags=-1
5(16293) check_via_address(192.168.1.122, 192.168.1.122, 0)
5(16293) DEBUG:destroy_avp_list: destroing list (nil)
5(16293) receive_msg: cleaning up
7(16295) SIP Request:
7(16295) method: <REGISTER>
7(16295) uri: <sip:192.168.1.119>
7(16295) version: <SIP/2.0>
7(16295) parse_headers: flags=1
7(16295) Found param type 232, <branch> = <z9hG4bK-5579ff0b>; state=16
7(16295) end of header reached, state=5
7(16295) parse_headers: Via found, flags=1
7(16295) parse_headers: this is the first via
7(16295) After parse_msg...
7(16295) preparing to run routing scripts...
7(16295) DEBUG : is_maxfwd_present: searching for max_forwards header
7(16295) parse_headers: flags=128
7(16295) end of header reached, state=9
7(16295) DEBUG: get_hdr_field: <To> [31]; uri=[sip:test@192.168.1.119]
7(16295) DEBUG: to body [test <sip:test@192.168.1.119>
]
7(16295) get_hdr_field: cseq <CSeq>: <170> <REGISTER>
7(16295) DEBUG: is_maxfwd_present: value = 70
7(16295) DEBUG: add_param: tag=79b50153b98e6976
7(16295) end of header reached, state=29
7(16295) parse_headers: flags=256
7(16295) DEBUG: get_hdr_body : content_length=0
7(16295) found end of header
7(16295) find_first_route(): No Route headers found
7(16295) loose_route(): There is no Route HF
7(16295) check_nonce(): comparing
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012] and
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012]
8(16296) SIP Request:
8(16296) method: <REGISTER>
8(16296) uri: <sip:192.168.1.119>
8(16296) version: <SIP/2.0>
8(16296) parse_headers: flags=1
8(16296) Found param type 232, <branch> = <z9hG4bK-5579ff0b>; state=16
8(16296) end of header reached, state=5
8(16296) parse_headers: Via found, flags=1
8(16296) parse_headers: this is the first via
8(16296) After parse_msg...
8(16296) preparing to run routing scripts...
8(16296) DEBUG : is_maxfwd_present: searching for max_forwards header
8(16296) parse_headers: flags=128
8(16296) end of header reached, state=9
8(16296) DEBUG: get_hdr_field: <To> [31]; uri=[sip:test@192.168.1.119]
8(16296) DEBUG: to body [test <sip:test@192.168.1.119>
]
8(16296) get_hdr_field: cseq <CSeq>: <170> <REGISTER>
8(16296) DEBUG: is_maxfwd_present: value = 70
8(16296) DEBUG: add_param: tag=79b50153b98e6976
8(16296) end of header reached, state=29
8(16296) parse_headers: flags=256
8(16296) DEBUG: get_hdr_body : content_length=0
8(16296) found end of header
8(16296) find_first_route(): No Route headers found
8(16296) loose_route(): There is no Route HF
8(16296) check_nonce(): comparing
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012] and
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012]
5(16293) SIP Request:
5(16293) method: <REGISTER>
5(16293) uri: <sip:192.168.1.119>
5(16293) version: <SIP/2.0>
5(16293) parse_headers: flags=1
5(16293) Found param type 232, <branch> = <z9hG4bK-5579ff0b>; state=16
5(16293) end of header reached, state=5
5(16293) parse_headers: Via found, flags=1
5(16293) parse_headers: this is the first via
5(16293) After parse_msg...
5(16293) preparing to run routing scripts...
5(16293) DEBUG : is_maxfwd_present: searching for max_forwards header
5(16293) parse_headers: flags=128
5(16293) end of header reached, state=9
5(16293) DEBUG: get_hdr_field: <To> [31]; uri=[sip:test@192.168.1.119]
5(16293) DEBUG: to body [test <sip:test@192.168.1.119>
]
5(16293) get_hdr_field: cseq <CSeq>: <170> <REGISTER>
5(16293) DEBUG: is_maxfwd_present: value = 70
5(16293) DEBUG: add_param: tag=79b50153b98e6976
5(16293) end of header reached, state=29
5(16293) parse_headers: flags=256
5(16293) DEBUG: get_hdr_body : content_length=0
5(16293) found end of header
5(16293) find_first_route(): No Route headers found
5(16293) loose_route(): There is no Route HF
5(16293) check_nonce(): comparing
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012] and
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012]
6(16294) SIP Request:
6(16294) method: <REGISTER>
6(16294) uri: <sip:192.168.1.119>
6(16294) version: <SIP/2.0>
6(16294) parse_headers: flags=1
6(16294) Found param type 232, <branch> = <z9hG4bK-5579ff0b>; state=16
6(16294) end of header reached, state=5
6(16294) parse_headers: Via found, flags=1
6(16294) parse_headers: this is the first via
6(16294) After parse_msg...
6(16294) preparing to run routing scripts...
6(16294) DEBUG : is_maxfwd_present: searching for max_forwards header
6(16294) parse_headers: flags=128
6(16294) end of header reached, state=9
6(16294) DEBUG: get_hdr_field: <To> [31]; uri=[sip:test@192.168.1.119]
6(16294) DEBUG: to body [test <sip:test@192.168.1.119>
]
6(16294) get_hdr_field: cseq <CSeq>: <170> <REGISTER>
6(16294) DEBUG: is_maxfwd_present: value = 70
6(16294) DEBUG: add_param: tag=79b50153b98e6976
6(16294) end of header reached, state=29
6(16294) parse_headers: flags=256
6(16294) DEBUG: get_hdr_body : content_length=0
6(16294) found end of header
6(16294) find_first_route(): No Route headers found
6(16294) loose_route(): There is no Route HF
6(16294) check_nonce(): comparing
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012] and
[414c90ab7f933a6b3c06a4bbbca22ce39fbf8012]
10(16299) MSILO:clean_silo: cleaning stored messages - 20
In using SER as proxy and asterisk as the PBX, do we
need to define anything in either of these two to make
asterisk send a response back to the SIP clients
behind the proxy? We have a defined dialplan in
extensions.conf in asterisk for recognizing dialed
digits of 911 and play a recorded message, but I do
not see any response from asterisk to the INVITE.
Our softphone is X-lite and we are making it dial 911
which should be forwarded blindly to a preset port on
asterisk via SER. This portion of the call works fine.
But when the INVITE reaches asterisk, it does nothing.
So, I wanted to know if asterisk needs to know about
the SER or the X-lite user for accepting and
processing the call besides the 911 dialplan
definition which is defined as follows:
exten => 911,1,Ringing
exten => 911,2,Wait,30
exten => 911,3,VoicemailMain
exten => 911,4,Goto(s,6)
Thanks in advance.
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hi My name is Jack I too am looking for the same answer and alos any other
comment made on this matter can you pls point me to where a dialogue is
going on about this
Thanks
jack
[Serusers] ser call records[Serusers] ser call records
Java Rockx javarockx at yahoo.com
Tue Sep 14 22:40:47 CEST 2004
Previous message: [Serusers] Click to dial - error 408
Next message: [Serusers] MediaProxy Working at half capacity.
Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
Hello All.
I'm a complete ser newbie and have just a few questions about ser.
Q: What options are available for collecting call detail records? The only
thing I've see so far would be RADIUS accounting.
Q: Can anyone point me to documentation or guidelines for creating a fault
tolerant ser proxy? I need to have a softswitch with 5-nines (99.999%)
uptime
or as close to it as possible. Aside from have a fault tolerant MySQL
backend,
I'm not sure what is needed or how to configure a redundant ser machine.
Q: Can anyone tell me how well ser really scales? Can it handle 100000
users?
Q: Can anyone point me to documentation for things like call waiting, call
forwarding, etc. I believe these calling features are implemented using CPL
but
I'm having problems finding the documentation for ser's CPL.
Best Regards,
Paul
_______________________________
Do you Yahoo!?
Declare Yourself - Register online to vote today!
http://vote.yahoo.com/
Previous message: [Serusers] Click to dial - error 408
Next message: [Serusers] MediaProxy Working at half capacity.
Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
More information about the Serusers mailing list
Hi,
did you set
$this->realm=$this->domainname=$this->default_domain="yourdomain";
try an echo to the domain name to see if the domain is set too. (the select statement of the script searches an user entry with a domain, but if no domain is set, it will searches an entry with an emtpy domain, which does not exist).
by
thomas jungbauer
________________________________
Von: serusers-bounces(a)lists.iptel.org im Auftrag von Shiran Guez
Gesendet: Fr 17.09.2004 23:25
An: serusers(a)lists.iptel.org
Betreff: [Serusers] SerWeb authentication fails,
Hi
I have set:
echo "uname: ".$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
