December 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 102, Issue 69
by Ahmed Habiba 17 Dec '14

17 Dec '14

Thanks you really Michael. My point is as below: 1-when FS get Invite from A number the TO is as below: To: <sip:2020@71.233.135.12> 2-and when I run answer application, FS send invite to A number with TO as below: To: <sip:2020@71.233.135.12> 3-I want when I run answer application to edit the TO header and to add some prefix like 505 as below: To: <sip:5052020@71.233.135.12> is this possible, by editing sip header, and if so how can I do so? Your kind usual support will be appreciated. > > From: Michael Jerris <mike(a)jerris.com <mailto:mike@jerris.com>> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Date: December 13, 2014 at 12:37:42 AM GMT+3 > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Subject: Re: [Freeswitch-users] SIP Header Modification / Manipulation > > > if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params. > >> On Dec 12, 2014, at 2:39 PM, Ahmed Habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: >> >> Dears, >> >> Kindly help me on how can I modify the SIP header in the re-invite in the below scenario: >> >> >> [A Number] <———>[FreeSWitch]<———>[B Number] >> >> >> 1-A Number Call B Number (Sip Invite sent from A number) >> 2-B number is not registered. >> 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number) >> >> in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To. >> >> >> Your help will be appreciated. >> > >

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CallerID for E911
by Joel White 16 Dec '14

16 Dec '14

Hi everyone, I was wondering what the maximum length a caller id number can be. I need to implement appending a location ID to each extension not assigned a DID. This is for purposes of the E911 system. Does anyone have experience with this? Any help would be greatly appreciated

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How to get the action executed using bind_digit_action ?
by Keith Laaks 16 Dec '14

16 Dec '14

Hi Freeswitchers. (Reposting under new thread - previous went to the wrong place.) I am running "FreeSWITCH Version 1.5.15b+git~20141215T224714Z~0b414a8de8~64bit (git 0b414a8 2014-12-15 22:47:14Z 64bit)” - updated this morning. I have the following actions in an extension being executed: <action application="info"/> <action application="bind_digit_action" data="start_echo,##,exec:info,both,self"/> <action application="digit_action_set_realm" data="start_echo"/> <action application="delay_echo" data="10"/> For the ‘info’ action at the top, I see the normal stuff printed in the console log: EXECUTE sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> info() 2014-12-15 20:52:06.137537 [INFO] mod_dptools.c:1647 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Channel-Name: [sofia/portaprod/27878881000(a)10.17.180.204 <mailto:sofia/portaprod/27878881000@10.17.180.204>] Unique-ID: [8218a229-562d-4706-85a2-988d2bae0041] <cut cut> Then follows the ‘bind_digit_action’ and ‘digit_action_set_realm’ to bind ‘##’ to executing the ‘info’ app. The ‘echo' application at the very bottom works as per normal and I get my audio echoed back. When I send “#9” - I get this in the log: 2014-12-15 21:13:32.777537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:13:33.637537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 9:800 2014-12-15 21:13:33.637537 [DEBUG] mod_dptools.c:132 sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> Digit NOT match binding [#9] So all normal. But here my issue. When I send ‘##’ - I only see this in the log: 2014-12-15 21:14:58.817537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF #:800 2014-12-15 21:14:59.317537 [DEBUG] mod_dptools.c:188 sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> Digit match binding [exec:info][both] 2014-12-15 21:14:59.317537 [DEBUG] switch_core_session.c:1188 Send signal sofia/portaprod/27878881000(a)10.10.10.10 <mailto:sofia/portaprod/27878881000@10.10.10.10> [BREAK] So its finding a match, but the app to be executed never does so. If it had, I would have seen the above ‘info’ stuff printed to the console. I actually want the call transferred to another extension when the user enters ‘##’ - but that similarly does not happen. It seems I can’t get any action to actually be executed. So what do I need to set first, such that the specified ‘action’ gets executed? Hope someone can help me out here. Looking at the code, I see that in mod_dptools.c there seems to be support for “blocking” and “non-blocking” execution modes. I figured out that to ‘activate’ the ’non-blocking mode the above action needs to change to: <action application="bind_digit_action" data="start_echo,##,exec[i]:info,both,self"/> [This sets ‘exec = 2’ such that ‘switch_core_session_execute_application’ is called - otherwise ‘switch_ivr_broadcast_in_thread’ is called.] But when I try this, as soon as I send the ##, I get a core dump. Regards Keith

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Strange xml_curl behavior. Can't see more than 7 variables on POST
by Muhammad Naseer Bhatti 16 Dec '14

16 Dec '14

I am using xml_curl to generate all config dynamically and this is getting very strange, my directory looks like this, <?xml version='1.0' encoding='UTF-8' standalone='no'?> <document type='freeswitch/xml'> <section name='directory'> <domain name='10.211.55.26'> <user id='9401404' cacheable='2000'> <params> <param name='a1-hash' value='3c66e98d4b174b42d3e17669b59165c9'/> <param name='max-registrations-per-extension' value='1'/> <param name='allow-empty-password' value='false'/> <param name='sip-force-expires' value='180'/> </params> <variables> <variable name='customer_id' value='98765'/> <variable name='effective_caller_id_name' value='Mr Bhatti'/> <variable name='effective_caller_id_number' value='9401404'/> <variable name='debug_call' value='0'/> <variable name='debug_sql' value='0'/> <variable name='user_context' value='vbilling'/> <variable name='hangup_after_bridge' value='true'/> <variable name='xbypass' value='google'/> <variable name='account_type_02' value='pre'/> <variable name='account_type_03' value='pre'/> <variable name='account_type' value='pre'/> </variables> </user> </domain> </section> </document> The directory is working fine and user is able to login. I have a bunch of variables defined here. When I look at the POST on the web server, I am only able to see first 7 variable no matter what their names are. If I change order for example account_type which is last variable, and make it on top, it shows there in the XML_REQUEST but then hangup_after_bridge goes away. I see only first 7. Are there any limitation on what channel variables can be POSTed to the web server or am I missing something here. -- Muhammad Naseer Bhatti

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Get caller IP address
by Snabel Kabiya 16 Dec '14

16 Dec '14

Hi, How can i get the caller IP from session? In sip trace I've: Call-ID: 7XHCXNSJABE2NGOBAHSXRZIGPA(a)1.2.3.4 i want to get the IP 1.2.3.4 i my Lua script, anyone knows how? Thanks, Snabel

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Originate call SIP call-id
by Doron Kruh 16 Dec '14

16 Dec '14

Hi All, How can I make sure the SIP Call-Id header sent by Freeswitch when originating a new call is in the form of: uuid@host ? Thank you, Doron

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[ERR][mod_json_cdr.c][Open of json_cdr.conf failed]
by Bilal Abbasi 16 Dec '14

16 Dec '14

Hi All, I have successfully compiled mod_json_cdr but when i go to fs_cli and load mod_json_cdr i get the below error please help. Plus there is no file named json_cdr.conf in autoconfig directory. I think FS is unable to locate that file. freeswitch@internal> load mod_json_cdr +OK Reloading XML -ERR [module load file routine returned an error] freeswitch@internal> 2014-12-16 06:37:25.377189 [INFO] switch_time.c:1369 Timezone reloaded 530 definitions 2014-12-16 06:37:25.377189 [ERR] mod_json_cdr.c:598 Open of json_cdr.conf failed 2014-12-16 06:37:25.377189 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_json_cdr.so **Module load routine returned an error** Regards Bilal Abbasi

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Hoot get variables in a LUA Hook Event.
by Jaime 16 Dec '14

16 Dec '14

Hello, I¹m trying to use the Hook event¹s with LUA script¹s when the channel Hangup a call, but I¹m unable to get the variables, for example uuid, ani or billed seconds for the call, I¹m using freeswitch 1.4.13 on Centos 6.6 and my configuration is: In /autoload_configs/lua.conf.xml [] <hook event="CHANNEL_HANGUP_COMPLETE" script="catch-event-cdr1.lua"/> In my catch-event-cdr1.lua script I wrote: ses = freeswitch.Session(); my_uuid = ses:getVariable("uuid"); freeswitch.consoleLog("notice"," uuid=("..my_uuid..")\n") But I¹m get the following error: 2014-12-15 23:51:46.100755 [ERR] switch_cpp.cpp:724 session is not initalized 2014-12-15 23:51:46.100755 [NOTICE] switch_cpp.cpp:1328 uuid=() Also, in the wiki doc I see a reference for the ³env² Object, (https://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) but if I call it as the example in the same lua script my result is: How I call the env variable: dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n²) And I got the following error: 2014-12-16 00:00:17.440782 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/catch-event-reg6.lua:27: attempt to index global 'env' (a nil value) Your help will be appreciated. Jamie Jaimecm(a)gmail.com

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Freeswitch based door communication
by Michael Traut 16 Dec '14

16 Dec '14

Further experimenting with another client was successful. So, the example from my previous post is working, thank you all. But: As there's still the possibility that my two hardphones have the same strange behavior - is there a way to get protocol information to detect WHY some machines behave differently? May be there are some options with the session bind to make the other phone happy to answer? The non-working software was CSipSimple version 1.02.03, Android. The working kind was Zoiper 1.19.7 Android.

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Freeswitch based door communication, 2nd
by Michael Traut 15 Dec '14

15 Dec '14

With your help i managed understand some basics and to do some setup with two android softphones. The softphones register as 1000 and 1001 respectively. Calls between them are working. Now i try to implement the "door comm" scenario: -> originate sofia/internal/1000%192.168.0.170 2001 XML default TEST TEST will as expected call softphone 1000. When i answer, my dialplan extension <extension name="test"> <condition field="destination_number" expression="^(2001)$"> <action application="bridge" data="user/1001"/> </condition> </extension> seems to be found and executed - but the softphone 1001 is not ringing now. After a while the call is discarded. Here's some Log output freeswitch@MIT-T540-W8> originate sofia/internal/1000%192.168.0.170 2001 XML default TEST TEST 2014-12-15 23:12:21.265437 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1000 [9e94d0e6-80cb-42fd-af72-22140bfaa0e2] 2014-12-15 23:12:21.385007 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/1000! 2014-12-15 23:12:26.104794 [INFO] switch_core_media.c:5328 Activating RTCP PORT 4031 2014-12-15 23:12:26.104794 [NOTICE] sofia.c:7475 Channel [sofia/internal/1000] has been answered +OK 9e94d0e6-80cb-42fd-af72-22140bfaa0e2 2014-12-15 23:12:26.104794 [NOTICE] switch_ivr.c:1854 Transfer sofia/internal/1000 to XML[2001@default] 2014-12-15 23:12:26.104794 [INFO] mod_dialplan_xml.c:635 Processing TEST <TEST>->2001 in context default 2014-12-15 23:12:26.104794 [INFO] switch_channel.c:3062 sofia/internal/1000 Flipping CID from "TEST" <TEST> to "Outbound Call" <1000%192.168.0.170> 2014-12-15 23:12:26.125865 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:1001@192.168.0.135:42990 [c49443c5-7bc0-4482-8851-8797b5e9102d] freeswitch@MIT-T540-W8>

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