- FreeSWITCH-users - FreeSWITCH Mailing Lists

FXO Card or Gateway
by Blake Priddy 20 Dec '13

20 Dec '13

Recommendation? Cost effective and reliable please :)

3 3

help! did mapping to extension 1002 not working
by netandtech 20 Dec '13

20 Dec '13

I wand to achieve to incoming calls from did numbers to an extension So this is what I have with no success. In the default.xml <extension name="test_did"> <condition field="destination_number" expression="^(2281118003)$"> <action application="set" 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 $${domain} ${dialed_ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="user/1002@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1002"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> In the public.xml <extension name="test_did"> <condition field="destination_number" expression="^(2281118003)$"> <action application="transfer" data="1002 XML default"/> </condition> </extension> And last I created a test.did.xml In freeswitch/conf/directory/default With the following <include> <user id="test_did"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="test_did"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="test_did"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension test_did"/> <variable name="effective_caller_id_number" value="2281118003"/> <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> My log is the following 2013-12-20 12:12:56.376409 [NOTICE] switch_channel.c:968 New Channel sofia/internal/302281081619(a)109.68.164.21 [f80ee77e-6999-11e3-8083-11b99fb2bf0d] 2013-12-20 12:13:06.436421 [WARNING] switch_core_state_machine.c:514 f80ee77e-6999-11e3-8083-11b99fb2bf0d sofia/internal/302281081619(a)109.68.164.21 Abandoned 2013-12-20 12:13:06.436421 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/302281081619(a)109.68.164.21 [CS_NEW] [WRONG_CALL_STATE] 2013-12-20 12:13:06.436421 [NOTICE] switch_core_session.c:1506 Session 9 (sofia/internal/302281081619(a)109.68.164.21) Ended 2013-12-20 12:13:06.436421 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/302281081619(a)109.68.164.21 [CS_DESTROY] I was excepting a ring to my extension 1002 normally when someone calls the did number.. The did number should be mapped normally to the gateway account .. and not to my freeswitch server with extension 1002.. I am little confused here.. but I have left it normal setup for did number. Best regards

1 0

SIP Trace Question / NAT one way audio problem.
by Sean P Devoy 20 Dec '13

20 Dec '13

Hi all, I seem to have fallen off the list! No mail since 7/26/2013. Have I made someone mad?? :) I have also been blissfully running with ISSUES what so ever since that time. I have a NATed client with PERIODIC one way audio. The client end has a Cisco 220W router with SIP ALG disabled. The phones are CISCO 504Gs and Polycom 330s(?), basciallt the same as everywhere else we have phones. I did a sofia global sip trace on for a while and captured the output. After writing an app to sort those sip packets into logical stream files, I dug in to the SIP conversations. This has left me with som questions for the people who know these things: 1. I found all my phones are REGISTERing every 23 seconds! That seems absurd TO ME, so where have I screwed that up/where do I set it and what is a reasonable number? 2. The only difference I see in the SIP packets from my working phone and these non-working ones is of course NAT related. The working phone's REGISTER looks like this: recv 757 bytes from udp/[WW.XX.YY.93]:5063 at 17:53:12.164203: ------------------------------------------------------------------------ REGISTER sip: <MYDOMAIN.COM> SIP/2.0 Via: SIP/2.0/UDP WW.XX.YY.93:5063;branch=z9hG4bK-238fcc8b;rport But the failing phones look like this: recv 801 bytes from udp/[AA.BB.CC.38]:5104 at 17:53:40.693691: ------------------------------------------------------------------------ REGISTER sip: <MYDOMAIN.COM> SIP/2.0 Via: SIP/2.0/UDP 10.2.2.239:5104;branch=z9hG4bKb374d6507894B431 The response is still sent to [AA.BB.CC.38]:5104 I guess first, IS THAT A PROBLEM? So, can anyone share any insight into the CISCO 220W config that I might be missing? Why would it only be a periodic problem? Could this be a problem where one end is using a wider range of rports than the other supports? Thanks, Sean --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

4 4

Conference timer synchronization with T1 FreeTDM - how?
by Bob Hartwig 20 Dec '13

20 Dec '13

Looking at the source for mod_conference.c, I see that the timer_name is hard-coded to "soft", i.e. the 1000 Hz. system clock. How can this work with a T1-based FreeTDM system, where the FreeTDM channels are presumably synchronized to the T1 clock source? I'm seeing some weirdness with my FreeTDM / Dahdi system, where only FreeTDM channels are hearing distortion on conference, and I'm almost certain that this is the culprit, but I have no idea how to proceed to fix it. I'm pretty new to FreeTDM, so I'm hoping that I'm missing something simple. Any thoughts? Thanks, Bob

5 7

ESL Outbound for driving IVRs
by Alex Balashov 19 Dec '13

19 Dec '13

Hi, Is there any practical way to take drive IVRs dynamically - via an API - while taking advantage of the more high-level features of the 'ivr' module? As I understand it, there's no way to pass in the IVR configuration parameters, i.e. the <menu> tags and their parameters, via ESL. At least, I can't find 'ivr' API commands anywhere. It seems to me that the only way to do this dynamically would be to mod_xml_curl to dynamically generate the XML config data in those sections. Can anyone confirm or deny that? It would certainly be nice to be able to do this purely through ESL outbound instead. Of course, at the end of the day, all these IVR actions can be manually scripted out through dialplan applications and orchestrated through ESL that way. But that implies a fair amount of wheel invention, on the basis that the 'ivr' module provides higher-level wrapping for a lot of it. I'd like to use it if possible. Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 2

in case of failure multiple condition and action bridge not working how to dial from different gateways according prefix numbers?
by netandtech 19 Dec '13

19 Dec '13

Thank you for that ! That helps a lot . I am having another question In case of failure can I add another condition below with another bridge to another gateway? What I want to achieve is in case of a gateway failure the call to continue with another gateway .. Also I am wondering how to make dial tone when someone open the handset and when a numbers is dialed how to hear the normal dialing tones/ Thank you in advanced! From: Andrew Cassidy [mailto:andrew@cassidywebservices.co.uk] Sent: Thursday, December 19, 2013 12:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] multiple condition and action bridge not working how to dial from different gateways according prefix numbers? Try adding break="on-true" to this tag: <condition field="destination_number" expression="^69(\d+)$"> -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director <http://www.cassidywebservices.co.uk/static/emailsig.png> <mailto:info@cassidywebservices.co.uk> T 03300 100 960 <mailto:info@cassidywebservices.co.uk> F 03300 100 961 <mailto:info@cassidywebservices.co.uk> E andrew(a)cassidywebservices.co.uk <mailto:andrew@cassidywebservices.co.uk> <mailto:info@cassidywebservices.co.uk> W www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk>

1 0

Can API commands return something other than CSV?
by Fraser Redmond 19 Dec '13

19 Dec '13

I'm connecting to FS from PHP over a socket, and parsing results using the php function str_getcsv() In my dialplan I have: <action application="playback" data="tone_stream://%(250,250,600,120)"/> When I run "api show channels" if a channel is currently playing that tone, then within the returned results I get these 3 headers and values: <snip>,application,application_data,dialplan,<snip> <snip>,playback,tone_stream://%(250,250,600,120);loops=3,XML,<snip> So the data-separator is the comma, but the data also contains commas without being escaped in any way. Is there any way to change the results to be returned in json? Or change the delimiter? I've made a regex replace to get around this particular case by matching parentheses... But I can see this happening again with some other value that has a comma outside of parentheses, and would rather get the results in a better format if possible. (In the wiki, I can see that CDRs and events can be sent in json, and the serialize command for events, but none of those would help this particular case.) Cheers, Fraser

3 3

Changing callee_id_name in a call
by Markus Carlstedt 19 Dec '13

19 Dec '13

Hi, I am trying to change the effective_callee_id_name when in a call. I tried using: uuid_setvar <uuid> effective_callee_id_name Testing I can see that the variable is set via: uuid_getvar <uuid> effective_callee_id_name But the info is not sent to the telephone, atleast not what I could see. / Markus

2 3

multiple condition and action bridge not working how to dial from different gateways according prefix numbers?
by netandtech 19 Dec '13

19 Dec '13

From: netandtech [mailto:bill_vus@hotmail.com] Sent: Wednesday, December 18, 2013 5:28 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] FW: how to dial from different gateways according prefix numbers? From: netandtech [mailto:bill_vus@hotmail.com] Sent: Wednesday, December 18, 2013 5:27 PM To: freeswitch Subject: how to dial from different gateways according prefix numbers? The following is the extension number that I use in the default.xml What I want to achieve is to Route the calls for landlines to different gateways and calls to mobiles from another gateway. <extension name="1002">   <condition field="destination_number" expression="^69(\d+)$"> <action application="transfer" data="sofia/gateway/myvoip/$0"/> </condition> <condition field="destination_number" expression="^22810(\d+)$"/> <action application="transfer" data="sofia/gateway/voipdito2/$0"/> </extension>

2 1

issue of freeswitch working with doubango webrtc clients
by Vincent Xia 19 Dec '13

19 Dec '13

im testing the interoperability of freeswitch and doubango sip clients including boghe and imsdroid, that both have the setting of media profile: default or webrtc, setting to default the call is fine as from or to the boghe/imdroid client, but setting to webrtc, these clients could only make outgoing calls, when receiving calls they respond the fs invite message with 488 "bad content", the fs console says the call failed due to incompitalbe destination: 2013-12-17 18:03:50.158267 [NOTICE] switch_ivr_originate.c:2699 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION] 2013-12-17 18:03:50.158267 [INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION the instruction document from doubango says for the webrtc clients to work with the "legacy" sip network a webrtc2sip module is required to as a sip proxy and webrtc breaker, but now freeswitch supports webrtc and sip over websocket, is a webrtc breaker still mandatory?

2 5