VoIP Mailing List Archives

[bipin at xbipin.com]

the dialplan is the standard config that comes with FS, no changes to it



Yossi Neiman wrote:

[freeswitch at cartisso...]

I believe the problem might lie in the dialplan that you are using to
get the user into the mailbox. If you can resubmit this section from
the user directory plus the snippet from the dialplan that pertains to
this, that would make it easier to help you. Alternatively, you can
stop by the #freeswitch channel on freenode and we can help in real time
then.

--
Yossi Neiman
Cartis Solutions, Inc.
http://www.cartissolutions.com



xbipin wrote:

[freeswitch at cartisso...]

Hit the F8 key when you're on the FreeSWITCH console and try calling
voicemail when you have the userid set as alphanumeric instead of
numeric, and send me the log of the debug information. Again, this will
be greatly easier if you join the IRC channel (#freeswitch on
irc.freenode.net), as we can work with you in real time.

--
Yossi Neiman
Cartis Solutions, Inc.
http://www.cartissolutions.com



xbipin wrote:

[brian at freeswitch.org]

You can also override this via the ${sip_mailbox} variable instead of
the dialed_ext that is a variable that is magically set for you based
on the mailbox = attr.

/b

On Sep 29, 2008, at 9:55 AM, xbipin wrote:

[bipin at xbipin.com]

what needs to be changed here, im a bit confused

<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="set" data="dialed_ext=$1"/>
<action application="export" data="dialed_ext=$1"/>
</condition>
<condition field="destination_number"
expression="^${caller_id_number}$">
<action application="set" data="voicemail_authorized=${sip_authorized}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default $${domain}
${dialed_ext}"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<anti-action application="bind_meta_app" data="1 b s execute_extension::dx
XML features"/>
<anti-action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<anti-action application="bind_meta_app" data="3 b s execute_extension::cf
XML features"/>
<anti-action application="set" data="transfer_ringback=${us-ring}"/>
<anti-action application="set" data="call_timeout=30"/>
<!-- <anti-action application="set"
data="sip_exclude_contact=${network_addr}"/> -->
<anti-action application="set" data="hangup_after_bridge=true"/>
<!--<anti-action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="db"
data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
<anti-action application="db"
data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
<anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/>
<anti-action application="answer"/>
<!--<anti-action application="send_display" data="Voicemail for
${dialed_ext}"/>-->
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default $${domain}
${dialed_ext}"/>
</condition>
</extension>




Brian West-3 wrote:

[brian at freeswitch.org]

You have some options here. I highly recommend you check out the
changes I have done to the default configs btw.

<anti-action application="voicemail" data="default $${domain} $
{user_data(${dialed_ext}@$${domain} param real_mailbox)}"/>

And just set the param of the real_mailbox for that user and it'll go
snag it out of the directory for that user.

Same for the check version:

<action application="voicemail" data="check default $${domain} $
{user_data(${dialed_ext}@$${domain} param real_mailbox)}"/>

Now if you really want to get crazy you can do something like this:

<action application="set" data="real_mailbox=${user_data(${dialed_ext}@
$${domain} param real_mailbox)}"/>
<action application="voicemail" data="default $${domain} ${cond($
{real_mailbox} == "" ? ${real_mailbox} : ${dialed_ext})}"/>

That would use the real_mailbox param on the user or the dialed_ext...
I didn't test that but it looks correct.

Chances are I have done nothing but confuse you.

/b



On Sep 29, 2008, at 11:19 AM, xbipin wrote:

[bipin at xbipin.com]

i tried it but still doesnt work, the below is the log

the suerid is bipin and the mailbox is 1001


2008-09-29 21:07:16 [NOTICE] switch_channel.c:538 switch_channel_set_name()
New
Channel sofia/internal/bipin@sip.xbipin.com
[52b5650d-3d2f-1f44-b36b-05bcd1f9c47
d]
2008-09-29 21:07:16 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing
bip
in->1001 in context default
2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481
switch_ivr_bind_dtmf_meta_ses
sion() Bound B-Leg: 1 execute_extension::dx XML features
2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481
switch_ivr_bind_dtmf_meta_ses
sion() Bound B-Leg: 2 record_session::C:Program Files
(x86)FreeSWITCH/recordings
/bipin.2008-09-29-21-07-17.wav
2008-09-29 21:07:17 [INFO] switch_ivr_async.c:1481
switch_ivr_bind_dtmf_meta_ses
sion() Bound B-Leg: 3 execute_extension::cf XML features
2008-09-29 21:07:18 [WARNING] mod_dptools.c:1972 user_outgoing_channel()
can't f
ind user [1001@sip.xbipin.com]
2008-09-29 21:07:18 [ERR] switch_ivr_originate.c:958 switch_ivr_originate()
Cann
ot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2008-09-29 21:07:18 [INFO] mod_dptools.c:1814 audio_bridge_function()
Originate
Failed. Cause: SUBSCRIBER_ABSENT
2008-09-29 21:07:18 [NOTICE] switch_channel.c:1426
switch_channel_perform_mark_p
re_answered() Ring-Ready sofia/internal/bipin@sip.xbipin.com!
2008-09-29 21:07:18 [NOTICE] sofia_glue.c:2032 sofia_glue_tech_media()
Pre-Answe
r sofia/internal/bipin@sip.xbipin.com!
2008-09-29 21:07:18 [NOTICE] mod_dptools.c:606 answer_function() Channel
[sofia/
internal/bipin@sip.xbipin.com] has been answered
2008-09-29 21:07:21 [NOTICE] sofia.c:2588 sofia_handle_sip_i_state() Hangup
sofi
a/internal/bipin@sip.xbipin.com [CS_EXECUTE] [NORMAL_CLEARING]
2008-09-29 21:07:21 [ERR] mod_sndfile.c:175 sndfile_file_open() Error
Opening Fi
le [C:\Program Files
(x86)\FreeSWITCH\storage\voicemail\default\sip.xbipin.com\(
null)\msg_52b5650d-3d2f-1f44-b36b-05bcd1f9c47d.wav] [System error : The
system c
annot find the file specified.
]
2008-09-29 21:07:21 [NOTICE] switch_core_session.c:812
switch_core_session_threa
d() Session 3 (sofia/internal/bipin@sip.xbipin.com) Ended
2008-09-29 21:07:21 [NOTICE] switch_core_session.c:814
switch_core_session_threa
d() Close Channel sofia/internal/bipin@sip.xbipin.com [CS_HANGUP]







Brian West-3 wrote:

[brian at freeswitch.org]

Why are you calling with the @sign? REMOVE THAT.



/b

On Sep 29, 2008, at 12:09 PM, xbipin wrote:

[bipin at xbipin.com]

im not using it by myself but actually my softphone calls my server whicha
ctually has openSBC as my session border controller which is in upper
registration mode so its proxys all the calls and then passes it over to FS
so that might be doing it. from my dialer i simply call 1001.
if i dont use opensbc then i cant use voip at all coz my isp blocks voip
calls and opensbc does the needed encryption and decryption.

is there anyway round it?



Brian West-3 wrote:

[bipin at xbipin.com]

according to what discussion we had in irc, atleast with the default config
and the changes u suggested, im supposed to atleast be able to hear my own
messages from my mailbox but i cant, it gives me record message everytime so
atleast this portion can be solved by checking caller id based on
authorization and if its the same as dialed extension then go to own mailbox



Brian West-3 wrote: