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mod_commands linker error on Visual studio 2008
by B Karthik 25 Apr '08

25 Apr '08

Hi All, mod_commands does not seem to link on VS 2008. It complains about two unresolved functions - switch_console_set_alias, switch_console_set_complete. I understand that this is because libedit library used by freeswitch is not compiled on windows. It was solved by adding the two functions in switch_console.c. Both the functions returns SWITCH_STATUS_FALSE. Is there a better way to do this? Thanks in advance. Warm Regards, B Karthik.

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Massively Scalable Voicemail-to-Email Farm
by Guy Fawkes 24 Apr '08

24 Apr '08

FreeSWITCH Users, We are looking to set-up a simple, inbound-only, massively scalable voicemail platform running on Debian or FreeBSD. The goals are as follows: Set-up one or more SIP peers for inbound traffic. All calls are inbound; no outbound dialing is allowed. Only options are to dial-in leave a message or enter your PIN and record a new custom greeting. No messages are stored on the server and all are emailed out as mp3 attachments. The database should contain the mappings for the telephone number, user pin, and email address target. Creation and management of the user data such as the email address and PIN will be handled by an external process. Obviously, the custom greeting identifer will have to get linked to the telephone number data somehow so the system plays the correct custom greeting for any given dialed number. What would be the recommended minimum (stripped down) Freeswitch install to achieve this? We don't want to install elements that serve no functional purpose for this model nor do we want unneccessary processes consuming resources. Can someone give us some idea of sizing? Since the users will live in a database, the only sizing concern would be how many concurrent calls (assuming all are actively recording messages) for a given hardware configuration, say dual Xeon 2.8GHz with 4GB RAM? We did see the FAQ (How many concurrent calls can it support? Any benchmarks?), but were hoping for some real-world feedback, or even suggestions, if available, as we will be starting with several used servers of the aforementioned configuration. We had already built a relatively robust call recording application using Asterisk some time ago, and found that writing to ramdisk was the only way to achieve any worthwhile per-server performance, and as we are FreeSWITCH newbies, we decided it wouldn't hurt to ask. Cheers!

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FS and Asterisk connectivity
by Brian Snipes 24 Apr '08

24 Apr '08

I wish to connect FS to * for interoperability testing and can't seem to get my configs correct. Has anyone done this already and if so can you post your configs? I can call from FS to the asterisk side via a dialplan entry where ^9(.*)$ passes to a gateway I've setup: <extension name="out9_asterisk"> <condition field="destination_number" expression="^9(.*)$"> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> sip_profiles/default/asterisk.xml : <include> <gateway name="asterisk"> <param name="username" value="freeswitch"/> <param name="realm" value="x.x.x.x"/> <param name="password" value="password"/> <param name="register" value="false"/> </gateway> </include> The problem I get with this is no audio either way. I am on rev 8099. TIA, Brian

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Dialplan, default configuration. Dialing 1 goes to busy signal.
by Bryan Reynolds 24 Apr '08

24 Apr '08

I am new to Freewitch and VOIP in general so I apologize for the dumb questions in advance. Which of course makes it alright! I successfully downloaded and installed CentOS and compiled and installed Freeswitch. I have 3 polycom 501 phones connected and working. They can dial most of the features from the default dialplan, like conferencing, MOH, etc. When I dial 1 immediately goes to busy signal and has this output on the CLI. 2008-04-19 05:24:38 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1002(a)10.0.2.117 [45bf1fa0-66cc-4f1f-9e8d-6d45bd8ef391] 2008-04-19 05:24:38 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002->1@default 2008-04-19 05:24:38 [NOTICE] switch_ivr.c:1032 switch_ivr_session_transfer() Transfer sofia/default/1002(a)10.0.2.117 to enum[1@default] 2008-04-19 05:24:38 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting 2008-04-19 05:24:38 [NOTICE] switch_core_state_machine.c:114 switch_core_standard_on_ring() Hangup sofia/default/1002(a)10.0.2.117 [CS_RING] [NO_ROUTE_DESTINATION] 2008-04-19 05:24:38 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 61 (sofia/default/1002(a)10.0.2.117) Ended 2008-04-19 05:24:38 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1002(a)10.0.2.117 [CS_HANGUP] This should not be the case because the regular expression below in the default plan should pick this up. I change the expression to "^9(10[01][0-9])$" and then I dialed 9 and the extension and it worked! Does anyone have an idea why that is happening? Is there some setting on the phone that makes it recognize 1 as something else? Here is the dialplan for local extensions. <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <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 $${domain} ${dialed_ext}"/> <anti-action application="bind_meta_app" data="1 a a execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 a a record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 a a execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/>  <anti-action application="set" data="hangup_after_bridge=true"/>  <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${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}@$${domain}"/> <anti-action application="answer"/>  <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> Bryan Reynolds

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question about the conference module: DTMF collection via XMPP?
by Brian Davis 24 Apr '08

24 Apr '08

I am actually not yet a user of FreeSWITCH, but I have been reading the online documentation to determine whether or not I could use it as part of a project I am currently working on that requires conferencing. I see that there are caller controls that can be used to play audio files to specific participants in the conference, using XMPP. I need to play an audio file and then be notified if/when the participant acknowledges the prompt in the audio file by pressing a DTMF key. I do not see a way to do this according to the documentation, but maybe there is something I am missing (or something not documented). Do any of you out there know of a way to collect DTMF over XMPP? Thanks, Brian

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Call for IVRs
by Brian Snipes 24 Apr '08

24 Apr '08

I would like to ask anyone that has working IVRs to please post them to the list or on the wiki ( http://wiki.freeswitch.org ). If you post to the list I will add them to the wiki for you. The few items on the wiki now were created no less than 11 months ago. There have been a lot of questions regarding IVRs and if you have benefited either from the list or from IRC please give back by sharing and helping others learn from your examples. Thanks, Brian

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Question about socket client
by Pete Kay 24 Apr '08

24 Apr '08

Hi, I have tried out the following command from Telnet: api originate sofia/default/1005 10000 &park Content-Type: api/response Content-Length: 41 +OK 8dc747d4-1261-11dd-92a8-2750a3aa164c SendMsg 8dc747d4-1261-11dd-92a8-2750a3aa164c Call-command: execute execute-app-name: fifo execute-app-arg:myqueue out nowait /tmp/announce.wav /usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav Content-Type: command/reply Reply-Text: -ERR invalid session id [8dc747d4-1261-11dd-92a8-2750a3aa164c] I tried to send a command to a uuid that is returned from the "originate" command, but it giving ERR. So, which uuid should I be using? Thank you in advance for your kind inputs. Regards, Pete

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Variable Manipulation
by Luis Jimenez 24 Apr '08

24 Apr '08

Hi, is there any way to manipulate variables in the dialplan? functions like cut or strip, replace, etc...

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recording callers separately
by jonathan augenstine 23 Apr '08

23 Apr '08

I have an application were I need to record the callers separately. I know this can be done on Aculab hardware but I would rather use Freeswitch. The requirements are that for 2 or more callers record each caller unmixed with the audio from the other callers. Can I do this on Freeswitch? Jonathan

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Calling Queue Implementation
by Pete Kay 23 Apr '08

23 Apr '08

Hi, I am in the processing of implementing a typical scenario for service call. So, in my scenario, I would have several agents who may log in and log out of the queue at different times of the day. People can call in and will be routed to one of the available agents. If no agent responses within 1 mins, then they would be asked to call back again. I checked the fifo mod, and I see how I can pack and unpark a user in the queue. However, how would I be able to route call to the agent automatically? Does it mean the agent have to initiate by pressing a # key and then have the dialplan to fetch the first user from the queue (using fifo out ) ? How is it possible for the agent's phone to ring automatically when a user is added to the queue? If someone can show me with an example, it will be greatly appreciated. Regards, Pete

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