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FreeSWITCH manual
by Madovsky 24 Feb '10

24 Feb '10

Hi dev friends ! Is this manual yet available for the last trunk version ? http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life Thanks Franck

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How to set outbound caller id info for multiple users/extensions
by Joseph Puchalski 23 Feb '10

23 Feb '10

I'm having problems setting different outbound caller id info for different extensions/users. I've set up a small system with two active users. I set up my users by copying and modifying existing entries from the dialplan files that come with freeSWITCH Here are my two extensions: <include> <user id="5859"> <params> <param name="password" value="xxxxxx"/> <param name="vm-password" value="xxxxxx"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="5859"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="5859"/> <variable name="effective_caller_id_number" value="5859"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> <include> <user id="5515"> <params> <param name="password" value="xxxxxxx"/> <param name="vm-password" value="xxxxxxx"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="5515"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="5515"/> <variable name="effective_caller_id_number" value="5515"/> <variable name="outbound_caller_id_name" value="Jane Doe"/> <variable name="outbound_caller_id_number" value="5558395515"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> These extensions are in files named 5859.xml and 5515.xml respectively. I'm using a SIP trunk from Vitelity (in and out) with two DIDs corresponding to the extensions above. Inbound and outbound calling work as needed with one exception: Calls originating from user/extension 5515 go out with the caller ID of extension/user 5859. Extension 5859 was the first that I created. Where should I be setting the outbound caller id number for my second extension? I've been trying to track this down in the available documentation but have been unable to do so. I apologize ahead of time if this is answered somewhere obvious that I've missed. Thanks for any help. Joe (FreeSWITCH newbie)

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Question about sofia_contact
by Helmut Kuper 23 Feb '10

23 Feb '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to setup a FS sofia sip-profile which allows me to have multiple sip-profiles but one registration database. So I set the following parameters: <param name="challenge-realm" value="auto_from"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> where domain is set to "mydomain". "sofia status profile internal" delivers the following: Call-ID: 3c26705038e5-vwlg8u5q9cwe User: 2701@mydomain Contact: <sip:2701@85.16.245.208:1024;line=l2wksedo> Agent: snom370/8.2.22 Status: Registered(UDP)(unknown) EXP(2010-02-19 16:13:31) Host: ippbx-prod-node0 IP: 85.16.245.208 Port: 1024 Auth-User: 2701 Auth-Realm: mydomain MWI-Account: 2701@mydomain sofia_contact internal/2701@mydomain delivers this: error/user_not_registered The Phone is fully functional. I use SVN trunk 16601 regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFLfqlT4tZeNddg3dwRAgk1AJ4gtXoVLn8anWF6BX2nijuvApoJiwCbBYyT jo6HuOqH62g8Mmia5PEXII8= =fDxi -----END PGP SIGNATURE-----

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leg_timeout with ignore_early_media false
by TTNC - Technical 23 Feb '10

23 Feb '10

Hi Guys I'm trying to create a hunt group where ignore_early_media = false is set, so that the international ring tone is passed through to the caller. Setting ignore_early_media = false on the channel does what I want, but with this set leg_timeout is not honoured. I've switched to use bridge_answer_timeout which is honoured if ignore_early_media = false and the call progresses through the different legs, but bridge_answer_timeout times out the call after the set period, even if the call has been successfully answered and bridged. I've tried all different combinations of group_confirm_cancel_timeout [1|2|3], none of them seem to affect bridge_answer_timeout. Does anyone have a solution for timing out legs of a hunt group with ignore_early_media = false set? This is my dial string: '{caller-id-in-from=true,origination_caller_id_name=012345123123,origination_caller_id_number= 012345123123}[bridge_answer_timeout=20]sofia/internal/4412345123123(a)sipipgw.siphost.net |sofia/internal/4412345123123(a)sipipgw.siphost.net' And in my lua application, I'm setting the following: session:setVariable("group_confirm_cancel_timeout", "1"); -- substitute 1 for 1, 2 or 3, none work. session:setVariable("ignore_early_media", "false"); I'm using the latest trunk revision - it's still happening. Any suggestions welcome. Thanks Russ

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sending a sip notify with sendevent
by Christian Löschenkohl 23 Feb '10

23 Feb '10

hi i try to send a sip notify message to a registered sip device "sofia profile nat flush_inbound_reg 10(a)vts.vie.xpirio.net reboot" works, but i need to send "check-sync;reboot=false" - so the device does a resync and don't do a reboot my message looks like this sendevent NOTIFY profile: nat event-string: check-sync;reboot=false user: 10 host: vts.vie.xpirio.net content-type: application/simple-message-summary if i listen on the loopback interface i do see ## T 2010/02/22 18:11:59.083204 127.0.0.1:51840 -> 127.0.0.1:8021 [AP] sendevent NOTIFY profile: nat event-string: check-sync;reboot=false user: 10 host: vts.vie.xpirio.net content-type: application/simple-message-summary ## T 2010/02/22 18:11:59.084032 127.0.0.1:8021 -> 127.0.0.1:51840 [AP] Content-Type: command/reply Reply-Text: -ERR invalid -------- i don't get what it is wrong. i also rechecked the registered user in the sqlite database and this looks good to me. no message is send to the user. we do use multiple domains, so user could also be 10(a)somedomain.com - or am i wrong on this? could somebody please bring some light in this. we do use trunk rev. 16631 br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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rtp timeout and call hangs up
by Madovsky 23 Feb '10

23 Feb '10

Hello, A leg local extension (codec GSM) -----> B leg local extension (codec PCMU) rtp timeout after 300 sec the call hangs up, I can ear audio and no problem of configuration. did I forget to set anything ? below the sip trace : ACK sip:1001@67.xx.xx.138:5060;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport;branch=z9hG4bK4t7xGwrintJDYr4HPfz0UQ.. From: "1000" <sip:1000@67.xx.xx.138:5080>;tag=52834814554 To: "Extension 1001" <sip:1001@67.xx.xx.138>;tag=3SKarSKZB63DF Contact: <sip:1000@67.xx.xx.138:62690> CSeq: 1 ACK Max-Forwards: 70 Call-ID: 2053375473(a)67.xx.xx.138 ------------------------------------------------------------------------ 2010-02-23 02:02:19.649316 [NOTICE] mod_sofia.c:853 Hangup sofia/external/1000@67.xx.xx.138:5080 [CS_EXECUTE] [MEDIA_TIMEOUT] 2010-02-23 02:02:19.649316 [NOTICE] switch_ivr_bridge.c:634 Hangup sofia/internal/sip:1001@70.81.84.218:2249 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] send 622 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.653239: ------------------------------------------------------------------------ BYE sip:1000@67.xx.xx.138:62690 SIP/2.0 Via: SIP/2.0/UDP 67.xx.xx.138;rport;branch=z9hG4bK790g187j50eNF Max-Forwards: 70 From: "Extension 1001" <sip:1001@67.xx.xx.138>;tag=3SKarSKZB63DF To: "1000" <sip:1000@67.xx.xx.138:5080>;tag=52834814554 Call-ID: 2053375473(a)67.xx.xx.138 CSeq: 127335933 BYE Contact: <sip:1001@67.xx.xx.138:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: FreeSWITCH;cause=604;text="MEDIA_TIMEOUT" Content-Length: 0 ------------------------------------------------------------------------ 2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1179 Session 5 (sofia/external/1000@67.xx.xx.138:5080) Ended 2010-02-23 02:02:19.649316 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/1000@67.xx.xx.138:5080 [CS_DESTROY] recv 289 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.656927: ------------------------------------------------------------------------ SIP/2.0 200 OK Content-Length: 0 Via: SIP/2.0/UDP 67.xx.xx.138;rport=5060;branch=z9hG4bK790g187j50eNF From: "Extension 1001" <sip:1001@67.xx.xx.138>;tag=3SKarSKZB63DF To: "1000" <sip:1000@67.xx.xx.138:5080>;tag=52834814554 CSeq: 127335933 BYE Call-ID: 2053375473(a)67.xx.xx.138 ------------------------------------------------------------------------ recv 391 bytes from udp/[67.xx.xx.138]:62690 at 07:02:19.660173: ------------------------------------------------------------------------ BYE sip:1001@67.xx.xx.138:5060;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA.. From: "1000" <sip:1000@67.xx.xx.138:5080>;tag=52834814554 To: "Extension 1001" <sip:1001@67.xx.xx.138>;tag=3SKarSKZB63DF Contact: <sip:1000@67.xx.xx.138:62690> CSeq: 2 BYE Max-Forwards: 70 Call-ID: 2053375473(a)67.xx.xx.138 ------------------------------------------------------------------------ send 505 bytes to udp/[67.xx.xx.138]:62690 at 07:02:19.660607: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 67.xx.xx.138:62690;rport=62690;branch=z9hG4bKGTIQmsc5dUvSdP0ZtJFKcA.. From: "1000" <sip:1000@67.xx.xx.138:5080>;tag=52834814554 To: "Extension 1001" <sip:1001@67.xx.xx.138>;tag=3SKarSKZB63DF Call-ID: 2053375473(a)67.xx.xx.138 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16676M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 669 bytes to udp/[70.81.84.218]:2249 at 07:02:19.869656: ------------------------------------------------------------------------ BYE sip:1001@70.81.84.218:2249;rinstance=f17f8a2dd028c23d SIP/2.0 Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport;branch=z9hG4bKQ1y6QNQ56aDDS Max-Forwards: 70 From: "1000" <sip:1000@67.xx.xx.138>;tag=SK3p6Frt5pa7D To: <sip:1001@70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549 Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2 CSeq: 127335918 BYE Contact: <sip:mod_sofia@67.xx.xx.138:5080> User-Agent: FreeSWITCH and Rock! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1179 Session 6 (sofia/internal/sip:1001@70.81.84.218:2249) Ended 2010-02-23 02:02:19.869666 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/sip:1001@70.81.84.218:2249 [CS_DESTROY] recv 411 bytes from udp/[70.81.84.218]:2249 at 07:02:19.986415: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 67.xx.xx.138:5080;rport=5080;branch=z9hG4bKQ1y6QNQ56aDDS Contact: <sip:1001@70.81.84.218:2249;rinstance=f17f8a2dd028c23d> To: <sip:1001@70.81.84.218:2249;rinstance=f17f8a2dd028c23d>;tag=df099549 From: "1000"<sip:1000@67.xx.xx.138>;tag=SK3p6Frt5pa7D Call-ID: 263d4030-9aec-122d-6d87-00e0ed0b00c2 CSeq: 127335918 BYE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 ------------------------------------------------------------------------ Thanks Franck

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Is there any way to loop a dialplan?
by Mark Sobkow 23 Feb '10

23 Feb '10

Let me explain what it is I'm trying to do. Maybe there's another way to achieve it. When an operator dials in to the log-in line (e.g. Extension 6000), I use play_and_get_digits to collect the operator's PIN. I then need to be able to fire up some Erlang (or Javascript) to verify the PIN, and after verification, put the call into a park state, collecting the UUID. I then need to fire an event to Erlang passing along the parked UUID and the operator's PIN so that Erlang can direct received customer calls to the operators based on relatively complex criteria that won't fit in a dialplan. The catch is that when I get a customer call, I collect their info via IVR menus, park the call, and fire an event to Erlang with the UUID of the parked call and info collected from the IVR. Erlang analyses the info, selects an operator who is free, and bridges the calls. The problem I'm having is figuring out a way to get the operator leg to go back into a park state after handling the customer's call. Ideally I want the operator to be presented with a short IVR to collect info about how the call was handled, but I can do that through a custom application GUI if I need to. Regardless, once an operator logs in, they need to _stay_ logged in until they explicityly log out, but I can't figure out how I'm supposed to do that without some sort of looping capability. One thing I was thinking that might work is to set up a set of "dummy" extensions that I can have Erlang dial and bridge which contain a dialplan fragment to collect the IVR call result, park the call, and issue the operator PIN and parked UUID again to Erlang. That way between Erlang events and the dialplan fragment I end up with an effective "loop". (Though I've yet to figure out how I can break out of that loop. Maybe it'll have to be an IVR option for logging out.) Sample code/dialplans would be good, but for now I'll settle for knowing whether I'm at least on the right track for how to implement this beast. Note that I only want to drop into Javascript if I can't figure out how to do it with dialplans and Erlang. Thanks for any ideas and suggestions. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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ESF_PAGE_GROUP in a box with 2 interfaces
by Luis F Urrea 23 Feb '10

23 Feb '10

Hi all, I have a FS process running on a fw with 2 ethernet interfaces, the FS process is bound to the internal iface but when I use esf_page_group it tries to forward multicast packets through the external iface, is there a config parameter maybe to be able to control this? TIA

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Re: [Freeswitch-users] Is there any way to loop a dialplan?
by Mark Sobkow 22 Feb '10

22 Feb '10

Apparently you can't have that first call waiting for the FIFO to be picked up. It'll bridge a FIFO, but the remainder of the dial plan never executes. *sigh* What is INTENDED to happen is that the operator dials in to extension 6000 and enters their PIN, then gets put in the FIFO. The idea is that after they've handled a call from the FIFO, the dialplan for extension 6000 resumes, collects the call result, and then enters the looping dialplan for extension 6001, which just keeps putting them back in the queue, collecting a result code, and repeating ad-nauseum. I've also tried calling a loopback extension so I could properly bridge the call, thinking that would respect the set hangup_after_bridge=false, but that doesn't work either. I may be frustrated, but I'm having fun... <extension name=\"TestBridging\"> \t<condition field=\"destination_number\" expression=\"^6000$\"> \t\t<action application=\"answer\" /> \t\t<action application=\"play_and_get_digits\" data=\"4 4 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav operator_pin \\d+\" /> \t\t<action application=\"log\" data=\"INFO operator_pin is ${operator_pin}\" /> \t\t<action application=\"log\" data=\"INFO Queing operator to fifo\" /> \t\t<action application=\"set\" data=\"hangup_after_bridge=false\" /> \t\t<action application=\"bridge\" data=\"loopback/6003\" /> \t\t<action application=\"log\" data=\"INFO Handled first call\" /> \t\t<action application=\"play_and_get_digits\" data=\"1 2 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav call_result \\d+\" /> \t\t<action application=\"log\" data=\"INFO call_result is ${call_result}\" /> \t\t<action application=\"set\" data=\"hangup_after_bridge=true\" /> \t\t<action application=\"bridge\" data=\"loopback/6001\" /> \t</condition> </extension> <extension name=\"TestLoop\"> \t<condition field=\"destination_number\" expression=\"^6001$\"> \t\t<action application=\"answer\" /> \t\t<action application=\"log\" data=\"INFO Requeing operator to fifo\" /> \t\t<action application=\"set\" data=\"hangup_after_bridge=false\" /> \t\t<action application=\"bridge\" data=\"loopback/6003\" /> \t\t<action application=\"log\" data=\"INFO Handled another call\" /> \t\t<action application=\"play_and_get_digits\" data=\"1 2 1 5000 # /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav /opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav call_result \\d+\" /> \t\t<action application=\"log\" data=\"INFO call_result is ${call_result}\" /> \t\t<action application=\"set\" data=\"hangup_after_bridge=true\" /> \t\t<action application=\"bridge\" data=\"loopback/6001\" /> \t</condition> </extension> <extension name=\"TestCallIn\"> \t<condition field=\"destination_number\" expression=\"^6002$\"> \t\t<action application=\"answer\" /> \t\t<action application=\"log\" data=\"INFO Pulling operator\" /> \t\t<action application=\"set\" data=\"hangup_after_bridge=true\" /> \t\t<action application=\"fifo\" data=\"6003@${domain_name} out nowait\" /> \t\t<action application=\"log\" data=\"INFO Done call\" /> \t</condition> </extension> <extension name=\"park\"> \t<condition field=\"destination_number\" expression=\"^6003$\"> \t\t<action application=\"fifo\" data=\"6003@${domain_name} in\" /> \t</condition> </extension> <extension name=\"unpark\"> \t<condition field=\"destination_number\" expression=\"^6004$\"> \t\t<action application=\"answer\" /> \t\t<action application=\"fifo\" data=\"6003@${domain_name} out nowait\" /> \t</condition> </extension> -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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8000 rate .wav files
by Mark Sobkow 22 Feb '10

22 Feb '10

I've got the 8000 sample rate .wav files installed for Freeswitch. According to the logs, my SIP phone is connecting with an 8000 rate. However, when I try to play_and_get_digits using those sound files, I get errors: 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file format [wav] for [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav]! 2010-02-22 11:15:22.029347 [ERR] switch_core_file.c:122 Invalid file format [wav] for [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-bad-pin.wav]! Aren't .wav files supposed to be compatible with all codecs for playback? If not, what do I have to do to convert them to the proper formats? How do I find out what the proper formats are? -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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