August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] voicemail to email not working
by Kim Culhan 28 Aug '12

28 Aug '12

According to the docs sSMTP is usually used to send all messages to a single address specified in the ssmtp.conf file unless you set it up so 'address rewriting is disabled.' http://linux.die.net/man/5/ssmtp.conf Seems you might want to follow Michael's advice and use Tony's script or use the original sendmail instead. -kim On Tue, August 28, 2012 11:22 am, Abaci wrote: > using sSMTP > <http://wiki.freeswitch.org/wiki/Mod_voicemail#sSMTP_settings>which also > replaces sendmail > > On 8/28/2012 10:55 AM, Michael Collins wrote: >> What MTA are you using? I know that I've had much more success using >> Tony's sendmail replacement script. (src/scripts/perl/sendmail). >> >> -MC >> >> On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64(a)gmail.com >> <mailto:abaci64@gmail.com>> wrote: >> >> I'm trying to get FreeSWITCH to send my voicemail to my email and >> can't it to work. I have sSMTP installed and working (tested using >> voicemail.tpl) I see in the FreeSWITCH logs *"switch_utils.c:761 >> Emailed file [/tmp/mail.13461330724a96] to [me(a)mydomain.com >> <mailto:me@mydomain.com>]" *however in /var/log/maillog I see >> nothing and I don't get the email. does someone have a clue on >> what might be wrong here, or is there any debug I can enable for this? >> >> _*in switch.conf.xml I have:*_ >> <param name="mailer-app" value="sendmail"/> >> <param name="mailer-app-args" value="-t"/> >> >> _*in voicemail.conf.xml I have:*_ >> <email> >> <param name="template-file" value="voicemail.tpl"/> >> <param name="notify-template-file" >> value="notify-voicemail.tpl"/> >>  >> <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/> >>  >> </email> >> >> _*in the directory (xml_curl) I have:*_ >> <document type="freeswitch/xml"> >> <section name="directory"> >> <domain name="mydomain.com <http://mydomain.com>"> >> <user id="200"> >> <params> >> <param name="password" value="cj987r5y95"/> >> <param name="vm-password" value="1234"/> >> <param name="vm-enabled" value="true"/> >> <param name="vm-email-all-messages" value="true"/> >> <param name="vm-mailto" value="me(a)mydomain.com" >> <mailto:me@mydomain.com>/> >> <param name="vm-mailfrom" value="Voicemail Server >> <voicemail(a)mydomain.com> <mailto:voicemail@mydomain.com>"/> >> <param name="vm-attach-file" value="true" /> >> <param name="dial-string" >> value="${sofia_contact(200(a)mydomain.com)}" >> <mailto:$%7Bsofia_contact%28200@mydomain.com%29%7D>/> >> </params> >> <variables> >> <variable name="toll_allow" >> value="domestic,international,local"/> >> <variable name="user_context" value="routing"/> >> <variable name="effective_caller_id_name" value="test >> user"/> >> <variable name="effective_caller_id_number" >> value="2125551212 <tel:2125551212>"/> >> <variable name="outbound_caller_id_number" >> value="2125551212 <tel:2125551212>"/> >> </variables> >> </user> >> </domain> >> </section> >> </document>

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how to play a prompt after connection
by Dave Horton 28 Aug '12

28 Aug '12

I need to do the following: (1) connect an inbound caller to ivr, and collect a phone number (2) bridge the caller to the entered phone number (3) just after connection, play a prompt to the called party 'this call is being recorded' (4) make a recording of the call I basically have everything working except number 3 -- playing the prompt to the called party immediately after they pick up. Since I am successfully recording the call I assume we have a conference going on freeswitch, and I should be able to play the prompt to the conference somehow (it's ok if the calling party hears it as well). I'm doing this in lua, but I think if someone points me to the freeswitch commands to use to stream a file to both parties immediately after bridging that's all I need.

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Ringing after queue exit
by Tiago N. Sampaio 28 Aug '12

28 Aug '12

Hi all, Im very new to FS, im comming from asterisk world hehe! Today i put my 1st FS in operation!!! Lets go, i have some queues in my company (queue1 queue2, etc..) I have max-wait on all queues to 60 seconds, and after that time i wanna bridge caller with some user, ex: <action application="callcenter" data="queue1@default" /> <action application="bridge" data="[leg_timeout=15]user/1002" /> <action application="voicemail" data="default $${domain} 1002"/> But when "max-wait-time" arrive, and fs try bridge to user/1002, no ringing or moh playing, and callers think that call was droped.. Only when voicemail is launched caller hear the sound of voicemail announce... Simplifying my question, how to put ringing on line when calling user/1002 after unsucessfull join on queue1? Hugs Tiago N. Sampaio

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Subscribe to sofia:: events?
by Avi Marcus 28 Aug '12

28 Aug '12

I seem to be unable to subscribe to sofia events. Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register responds: "-ERR no keywords supplied" Only subscribing to ALL shows them (with type CUSTOM). Am I doing something wrong? Is this a bug? -Avi

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Re: [Freeswitch-users] Subscribe to sofia:: events?
by Peter Olsson 28 Aug '12

28 Aug '12

"CUSTOM sofia::register" should work /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Avi Marcus Skickat: den 28 augusti 2012 16:03 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Subscribe to sofia:: events? Ideas...? Or should I open a jira on this? -Avi On Sat, Aug 25, 2012 at 11:36 PM, Avi Marcus <avi(a)avimarcus.net<mailto:avi@avimarcus.net>> wrote: I seem to be unable to subscribe to sofia events. Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register responds: "-ERR no keywords supplied" Only subscribing to ALL shows them (with type CUSTOM). Am I doing something wrong? Is this a bug? -Avi !DSPAM:503ccd2732761726646270!

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Open Source unified autodialer software released
by tahir almas 28 Aug '12

28 Aug '12

Pleased to announce the release of open source Fax , SMS and Voice broadcasting software solution ICTDialer http://www.ictdialer.org developed over reknown Drupal Conent Mnagment System and powerfull Plivo Communication framework , Your contribution and suggestions are welcome Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation.

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mod_CallCenter: Agents not being logged out
by Nitin Tomer 28 Aug '12

28 Aug '12

Hi, I've configured FreeSWITCH with 4 agents, in the following manner: <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1001@default" type="callback" contact="[origination_caller_id_name='Queue Caller',call_timeout=10]user/1001" status="Available" max-no-answer="30" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1002@default" type="callback" contact="[call_timeout=10]user/1002" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1003" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> </agents> <tiers> <tier agent="1000@default" queue="sales@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="1"/> <tier agent="1003@default" queue="sales@default" level="1" position="1"/> </tiers> Now I am trying to set an agent's status with this command at the console: callcenter_config [agent set status 'Logged out'] [1000@default] This command is executed alright, but when I execute "callcenter_config agent list", the status of 1000@default is still shown as "available"; why so? Also please tell me what should be the syntax if I want to set status when agent dials a number. And is there a way to set this status from outside, like from a web page click? Regards Nitin Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL) accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.

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mod_radius_cdr problem
by Evgeniy Movlyan 28 Aug '12

28 Aug '12

Hi all, I am currently running two billing modules: mod_radius_cdr and mod_nibblebill. I'm using mod_nibblebill only for hangup the call when user's balance is 0, for all other things i'm using radius. My problem is that when mod_nibblebill transfer the call to hangup extension - radius accounting is failing. My hangup extension: <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup" data="NORMAL_CLEARING"/> </condition> </extension> RADIUS log: http://pastebin.com/yeGPJZH9 I think my problem is that Freeswitch-Dst changing: in start packet Freeswitch-Dst = "0675420288" and in the stop packet Freeswitch-Dst = "hangup". But i don't know how to fix it. -- Evgeniy Movlyan, BestNet Ltd.

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Calls drop after ~1:48
by R W 28 Aug '12

28 Aug '12

We're seeing this same exact issue with a number of our phones (specifically CounterPath Bria for iOS and others). Calls placed disconnect after ~1:48 on some NATs, (specifically our wireless data carrier, AT&T 3G). http://lists.freeswitch.org/pipermail/freeswitch-users/2012-March/081317.ht… In freeswitch, the CLI shows: 2012-08-27 00:41:59.471201 [DEBUG] switch_core_session.c:1429 Session 174 (sofia/internal/2000(a)mydomain.com) Locked, Waiting on external entities 2012-08-27 00:41:59.471201 [NOTICE] switch_core_session.c:1447 Session 174 (sofia/internal/2000(a)mydomain.com) Ended 2012-08-27 00:41:59.471201 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/2000(a)mydomain.com [CS_DESTROY] 2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/2000(a)mydomain.com) Callstate Change HANGUP -> DOWN 2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/2000(a)mydomain.com) Running State Change CS_DESTROY 2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/2000(a)mydomain.com) State DESTROY 2012-08-27 00:41:59.471201 [DEBUG] mod_sofia.c:374 sofia/internal/2000(a)mydomain.com SOFIA DESTROY 2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:86 sofia/internal/2000(a)mydomain.com Standard DESTROY 2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/2000(a)mydomain.com) State DESTROY going to sleep We fixed this by shutting of the "Passive Session Timer" in the "SIP Miscellaneous." However, I'm curious to know if anyone can explain why this happens on some NATs and not others? Is there s problem with disabling the "Passive Session Timer" on our clients? -Rob

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forgot password on fusionpbx
by BookBag 27 Aug '12

27 Aug '12

Hello all, I accidentally typed the wrong password when setting up fusionpbx web login. Does anyone know how I can reset it from the console or terminal. I installed it with postgresql

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