September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CDR Behaviour (b_only)
by Ross McKillop 14 Sep '10

14 Sep '10

Hi, I'm curious how others handle CDRs from FreeSWITCH. I need them to end up in a MySQL database and the easiest way I've found of achieving this is to use the mod_xml_cdr module to call a PHP script that inserts the row. Is there a preferred way of achieving this? Also, I only need ONE cdr entry per call, for billing purposes, so what I end up doing is having process_cdr as "b_only" which works great for successful calls. However, a call that is not successfully bridged results in two HTTP POST calls to my script one with is_outbound = true (as I'd expect for the b leg) but also what appears to be the a leg of the CDR (despite requesting b_only) Is this the intended behaviour of "b_only" Regards, Ross

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Invalid Application hash
by Peder 13 Sep '10

13 Sep '10

Does anybody know what this means: 2010-09-13 11:01:50.353264 [ERR] switch_core_session.c:1791 Invalid Application hash Is it saying the application named hash is invalid? Or there is an invalid hash in an application?

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polycoms and alert_info=<http://...>
by Frank Carmickle 13 Sep '10

13 Sep '10

Hello all I am trying to get the polycom 335/650 to play the file specified by alert_info. I see from my http log that it is dowloading the file each time a call is placed to the phone with the alert_info channel variable set. It doesn't actually play them. I have tried with 8k pcm wave file, a 16k pcm wave and a 8k ulaw file and no joy. Anyone know what I may be missing in my polycom configs? Thanks --Frank

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How to check user in directory
by Boris Kovalenko 13 Sep '10

13 Sep '10

Hello! I have a set of local users registered to freeswitch. So, when user is not registered with switch I get USER_NOT_REGISTERED when route call to it. But is there a way to know is user present in directory and not registered, or even isn't in directory? I want to call voicemail in first case and play 'invalid number' message in second. Is this possible? -- Regards, Boris

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SPA942 - SLA - Inbound calls
by Aloysius Lloyd 13 Sep '10

13 Sep '10

Hello All, I have difficulty with Inbound SLA. Outbound SLA working without any issue. Inbound SLA not working. FreeSWITCH Version - FreeSWITCH Version 1.0.head (git-6875a97 2010-09-08 16-33-50 -0500) Linksys SPA942 - 6.1.5(a) Once the inbound call answered the following SQL Error appear on the debug log. Please let me know how to fix this issue. 2010-09-08 23:06:25.086299 [ERR] sofia_presence.c:687 PROCESS PRESENCE EVENT 2010-09-08 23:06:25.086299 [ERR] sofia_presence.c:1827 PRES SQL select sip_from_user,sip_from_host,call_info,call_info_state,uuid from sip_dialogs where hostname='li176-12' and sip_from_user='200' and sip_from_host=' mydomain.com' 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1848 PRES SQL select call_id,expires,sub_to_user,sub_to_host,event from sip_subscriptions where expires > -1 and hostname='li176-12' and sub_to_user='200' and sub_to_host=' mydomain.com' and (event='call-info' or event='line-seize') 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 0 [call_id]=[ 5c0f4043-aaa7bbd0(a)192.168.240.104] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 1 [expires]=[1284008633] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 2 [sub_to_user]=[200] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 3 [sub_to_host]=[mydomain.com] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 4 [event]=[call-info] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 0 [call_id]=[ 696f8b47-50fa4cba(a)192.168.240.110] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 1 [expires]=[1284008691] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 2 [sub_to_user]=[200] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 3 [sub_to_host]=[mydomain.com] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1675 SLA3: 4 [event]=[call-info] 2010-09-08 23:06:25.087545 [ERR] sofia_presence.c:1863 CLEAR SQL delete from sip_dialogs where sip_from_user='200' and sip_from_host='mydomain.com' and call_info_state='seized' 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:680 STATE SQL update sip_dialogs set call_info='appearance-index=1',call_info_state='active' where hostname='li176-12' and uuid='00f17b14-2e0b-4711-b082-a7c5a90621d8' 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:687 PROCESS PRESENCE EVENT 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:1827 PRES SQL select sip_from_user,sip_from_host,call_info,call_info_state,uuid from sip_dialogs where hostname='li176-12' and sip_from_user='200' and sip_from_host=' mydomain.com' 2010-09-08 23:06:25.088541 [ERR] sofia_presence.c:1848 PRES SQL select call_id,expires,sub_to_user,sub_to_host,event from sip_subscriptions where expires > -1 and hostname='li176-12' and sub_to_user='200' and sub_to_host=' mydomain.com' and (event='call-info' or event='line-seize') 2010-09-08 23:06:25.089546 [ERR] sofia_presence.c:1863 CLEAR SQL delete from sip_dialogs where sip_from_user='200' and sip_from_host='mydomain.com' and call_info_state='seized' Thanks Lloyd

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Question about execute_on_answer
by Helmut Kuper 13 Sep '10

13 Sep '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have a xml dialplan which arms a session with execute_on_answer "lua on_answer.lua" on_anwer.lua is this: direction = session:getVariable("Call-Direction") uuid = session:get_uuid() session:execute("log", "NOTICE INSERT "..direction..", "..uuid.."\n") session:execute("info") When a call is answered the script is executed, but direction is empty. Top of info is this: 2010-09-13 19:09:23.583688 [INFO] mod_dptools.c:1003 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Unique-ID: [40e982cd-052c-4718-9567-7d64a937ba12] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [2850(a)85.16.246.6] Channel-Call-UUID: [40e982cd-052c-4718-9567-7d64a937ba12] Answer-State: [answered] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Direction: [inbound] Caller-Username: [2850] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [] Caller-Caller-ID-Number: [2850] Caller-ANI: [2850] Caller-Destination-Number: [4919] Caller-Unique-ID: [40e982cd-052c-4718-9567-7d64a937ba12] Caller-Source: [mod_sofia] Caller-Context: [et_internal_error] Caller-RDNIS: [4919] Caller-Profile-Index: [2] Caller-Profile-Created-Time: [1284397763581765] Caller-Channel-Created-Time: [1284397711332733] Caller-Channel-Answered-Time: [1284397713461801] Caller-Channel-Progress-Time: [1284397711963763] Caller-Channel-Progress-Media-Time: [1284397713461801] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] variable_direction: [inbound] When I try to get "direction:" (last in info) the I got a value. So are not all variables listed in info are readable in execute_on_answer script? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMjl0d4tZeNddg3dwRArjBAJ99nO1xHcIciCt2OGhE8Qi4iFHglwCeOtxG vIpIFn2l1A6bsnXa9jWmzBE= =V/E6 -----END PGP SIGNATURE-----

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FS sigfaults when client connects
by Ben Sanford 13 Sep '10

13 Sep '10

I've got a Freeswitch server running on a Seagate Dockstar segfaulting when any client connects to it. Core dump is way too large to post as an attachment (54MB), you can grab it from mediafire: http://www.mediafire.com/?0197kd8zhaqbq2b. I've got the latest build that "make current" gives me yet this still happens. Turning on console debugging is worthless as the server crashes as soon as a client attempts to connect. Thanks for the help in advance. - Ben

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Re: [Freeswitch-users] get lcr channel variable from esl
by Nigel Kent 12 Sep '10

12 Sep '10

My code is set to receiving events as type plain. Changing the event type to xml makes the existing code not work . Is there a way I can use lcr with is scenario? --- On Fri, 10/9/10, Rupa Schomaker <rupa(a)rupa.com> wrote: From: Rupa Schomaker <rupa(a)rupa.com> Subject: Re: [Freeswitch-users] get lcr channel variable from esl To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Friday, 10 September, 2010, 13:13 It is streamed to you over ESL. On Fri, Sep 10, 2010 at 6:54 AM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: thanks I meant how does Freeswitch save "as xml" data and what is the way to access it --- On Fri, 10/9/10, Rupa Schomaker <rupa(a)rupa.com> wrote: From: Rupa Schomaker <rupa(a)rupa.com> Subject: Re: [Freeswitch-users] get lcr channel variable from esl To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Friday, 10 September, 2010, 11:19 On Fri, Sep 10, 2010 at 5:27 AM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: Hi I am not clear when you say api version because I am calling lcr through the esl @con.execute method, do you mean when used from event socket. API version is the version that is used via event socket and is what you get when called from the fs_cli or console. If that is the case then I think I would need to look into the as xml option because I am doing everthing from event socket. Right. What is does 'default' set and which module is needed to handle xml because I have not used xml for any other part of my code lcr 15555551212 default as xml the "default" is the lcr profile you are using. If you don't specify one then default is the profile used. However, in order to specify "as xml" a profile name must be specified. In this case I suggested default. As for module to process the xml, that depends on what language you are using on the client side, no? --- On Thu, 9/9/10, Rupa Schomaker <rupa(a)rupa.com> wrote: From: Rupa Schomaker <rupa(a)rupa.com> Subject: Re: [Freeswitch-users] get lcr channel variable from esl To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Thursday, 9 September, 2010, 23:21 The api version of mod_lcr doesn't give you lcr_auto_route. You can do lcr 15555551212 default as xml and get the lcr info as xml which can then be parsed on the client side. To build the equivalent of lcr_auto_route you can concatenate all the dialstrings separated by "|". I'd suggest transfering to the lcr/ endpoint instead. lcr/15555551212 The only way to get the channel var set is to use the application version of lcr. On Thu, Sep 9, 2010 at 2:49 PM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: I have used mod_lcr application in my esl script, digits are passed to mod_lcr and a dial string is found successfully as seen on the debug log. My question is how do I acces the variable ${lcr_auto_route} from esl. Previously in my script I have used this method to get variables which was set earlier in the script @con.api("uuid_getvar", "#{@uuid} myvarible").getBody However this is not working for lcr_auto_route. If I do(a)con.api("uuid_getvar", "#{@uuid} lcr_auto_route").getBody I get a _undef_ error -- -Rupa -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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what happened to iax
by covici＠ccs.covici.com 12 Sep '10

12 Sep '10

I have a need to connect to an asterisk box running iax -- what happened to mod_iax I can't find it anywhere? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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call rejected?
by covici＠ccs.covici.com 12 Sep '10

12 Sep '10

Hi. I was trying to use a gateway to galaxyvoice, and I have registered correctly. However I get a 403 response from them. It does work in asterisk with the following additional parameters from-user=<user name>, auth-user=<user-name> and auth=md5. I tried putting the from-user and auth-username explicitly in the gateway parameters, but no joy. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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