December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Guidelines for setting up contexts per user / gateway
by Nestor A Diaz 21 Dec '11

21 Dec '11

Hello Freeswitch Users List, I am trying to emulate asterisk behavior for user / gateway context under FreeSWITCH. For users inside the 'directory' directory I can set up different contexts using the 'user_context' variable, solved. But when i have to deal with external gateways sip_profiles seems to be the best place to put the configuration, and if i want to use just one sip profile due to firewall restricctions i am limited to just one context. According to FreeSWITCH FAQ: Q: How do I assign endpoints to contexts with different sets of extensions <http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_How_do_I_assign_endpoints…> Here are the different possible approaches: 1. use 1 profile per context you want to route to (each one needs a distinct ip:port) 2. use different domains in the registration data and use the auto context thing 3. send them all to a common context and execute_exten or transfer to somewhere else 4. send them to an IVR to decide where they go 5. use xml_curl to make it dish out a different dialplan based on who they are in the list of data you are fed I don't want to use 1 profile per context, so option 1 discarded. I don't want to use different domains in the registration data, since it have no semantic for a one company, so option 2 discarded, but for gateways it makes sense. I don't want to use an IVR for every user call, so option 4 discarded. So that gives me two choices, use xml_curl, which means i have to set up a webserver but that seems overkill, it seems to be an superset of option 3, so that left me with option 3 using just static XML files. I found this example on the dialplan XML: <extension name="Inbound_external"> <condition regex="any"> <regex field="${sip_from_host}" expression="domainA"/> <regex field="${sip_from_uri}" expression="1234567890@domainB"/> <regex field="${sip_from_uri}" expression="user@domainC"/> <regex field="caller_id_name" expression="^(John Smith)$"/> <regex field="caller_id_number" expression="^(55512341)|(55512342)|(55512343)$"/> <action application="set" data="domain_name=domainZ"/> <action application="transfer" data="${destination_number} XML domainZ"/> </condition> </extension> It seems to be the first place to start adapting to internal extension and transfering to the correct dialplan, anybody have a more refined example ? which variables can i trust / use in order to validate a call from a particular gateway and then set up a context transfer according to them. p.d. anybody knows about a good cdr web stats software for freeswitch ? i have used cdr-stats for asterisk in the past, newer version seems to have freeswitch support on the way, but stable ? any suggestions will be apreciated . Thank you !! -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

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Problem with make current
by lloyd thomas 21 Dec '11

21 Dec '11

Since I am not getting anywhere with my sip registration problem I decided to update FS from git using make current. Unfortunately I am getting the following errors. What can I do to resolve? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or â...â before âtz_tâ make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2

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transfer to conference from conference / caller controls
by Rodney 20 Dec '11

20 Dec '11

I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501@default to <digits>@dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney <group name="default"> <control action="mute" digits="1"/> <control action="hangup" digits="0"/> <control action="execute_application" digits="4" data="playback C:/Program Files/FreeSWITCH/recordings/conf_help.wav"/> <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> <control action="execute_application" digits="#" data="execute_extension 759 xml default"/> - <group name="dynamic"> <control action="mute" digits="1"/> <control action="execute_application" digits="0" data="execute_extension 401 XML default"/> (main ivr menu extension) <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> condition destination_number ^759$ 1 action answer 2 action set conference_user_list=| 11 action play_and_get_digits 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 15 action phrase spell,${target_num} 16 action conference ${target_num}-127.0.0.1@default

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originate via rpc
by Anders 20 Dec '11

20 Dec '11

Hi, I am trying to originate a call, using a mod command through the xml_rpc, but I am doing something wrong - not sure what it is. I post this url: http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}s… (numbers and IPs modified here) The reponse I get is this: -USAGE: <call url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] I can do other commands directly in the browser like this (such as http://1.2.3.4/webapi/show?calls) without any problems. Therefore my conclusion is that I am doing something wrong in this syntax for http. Any ideas? Thanks in advance! //Anders

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conference count minus 1
by Rodney 20 Dec '11

20 Dec '11

jeff, i have tried both expressions. even re tried original file and its same errors. just happen with new version update this week. also i get a bunch of "adding tone descripters" for busy, etc. that weren't there before on reloadxml rodney -Rodney, Do you have any unusual chars embedded in that string? that are not visible? Jeff -

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unexpectedly dial through gateway
by Zhuravlov Sergey 20 Dec '11

20 Dec '11

Hi, in my dialplan I try to call directly to the asterisk server follows: <extension name="ast_NTCOM-711"> <condition field="destination_number" expression="^(711)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1(a)ast.er.isk.ip"/> <action application="hangup"/> </condition> </extension> But it does not work for me! If the profile external has gateways then a call goes through a gateway, and receives forbidden. If in the profile external NOT have gateways - all is good and asterisk recieve call. I do not understand why this is happening? Where to look? -- Zhuravlov Sergey GTALK/JABBER:4orbit@gmail.com

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Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS]
by Stephen Wilde 20 Dec '11

20 Dec '11

I have a problem with a provider where the call are dropped by Freeswitch after receiving a 183 + SDP. I think that the problem is in SDP but I don't know where. The SDP that Freeswitch sends in INVITE is: v=0 o=FreeSWITCH 1324273767 1324273768 IN IP4 62.196.58.2 s=FreeSWITCH c=IN IP4 62.196.58.2 t=0 0 m=audio 52062 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 where the SDP that Freeswitch receives from the provider is: v=0 o=sansay-VSX 10 10 IN IP4 206.172.44.229 s=session controller c=IN IP4 206.172.44.229 t=0 0 m=audio 28364 RTP/AVP 18 150 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:150 telephone-event/8000 a=fmtp:150 0-15 a=ptime:20 After this SDP I see in the log: 2011-12-19 21:17:11.567017 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] Can anyone give me an advice to solve this issue? Stephen

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conf count minus 1 /invalid UTF-8 character
by Rodney 20 Dec '11

20 Dec '11

both the following expressions work. but since i went to the new version msi this week i get the following error echo'd in console about 20 times at the time of pressing digit to initiate the call count extension: {warning} switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it <action application="say" data="en number pronounced ${expr ${conf_count} - 1}"/> marius <action application="say" data="en number pronounced ${expr(${conf_count}-1)}"/> mc ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, December 19, 2011 2:57 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 103 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: PCI Compliance Over Telephone for Credit Cards- how? (Avi Marcus) 2. Re: announce conf count (total callers minus 1)? (Michael Collins) 3. originate via rpc (Anders) 4. Re: originate via rpc (Michael Collins) Encrypting yourself only helps if you have a T1/BRI whatever private link to the telco. I don't.. what are my options? -Avi On Mon, Dec 19, 2011 at 9:28 PM, Elliott Vogel <elliott(a)zoogmedia.com> wrote: I haven’t seen a company yet and I have searched – none of the big origination providers do and many of the smaller ones use the big providers – we are force to do our own encoding > >From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus >Sent: Monday, December 19, 2011 12:03 PM >To: FreeSWITCH Users Help >Subject: Re: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? > >So is there a provider for USA who takes T1 and encrypts it, so I can buy origination from them? > > >-Avi > >On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel <elliott(a)zoogmedia.com> wrote: >Well, I have worked a lot with PCI compliance in the past and I don’t think you can meet the requirements of encryption if you’re not doing encoding yourself because most voip service providers aren’t encrypting the calls. Also dtmf has the same for requirements and for T1 not being encrypted this is true but because the network is considered secured(funny)/private it’s doesn’t need to be – now if you would encapsulate t1 traffic to send it over the internet without encrypting it this would be unsecured. > >From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus >Sent: Monday, December 19, 2011 5:52 AM >To: FreeSWITCH Users Help >Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? > >I'm planning on an IVR to accept credit card information for signing up and renewal of my services. >Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. > >But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_p… on page 9: >Call centers will need to ensure that transmission of cardholder data across public networks is encrypted. >>This is part of PCI DSS Requirement 4 and includes: >> * ... >> * Voice or data streams over Voice over IP (VoIP) telephone systems, whenever sent over an open or public network. Note that only those consumer or enterprise VoIP systems that provide strong cryptography should be used. >> * Requiring agents to use analog telephone lines when a VoIP telephone system does not provide strong cryptography. >I'm doing dtmf, not voice, but I can't imagine that's LESS strict. > >I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? > > >-Avi Marcus > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > Try this: <action application="say" data="en number pronounced ${expr ${conf_count} - 1}"/>That's just off the top of my head, so if it doesn't work then tinker with it until you get it working. -MC On Sun, Dec 18, 2011 at 10:22 AM, Rodney <notlikeme75(a)yahoo.com> wrote: this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? > > > > >conf/dialplan/default/01_Announce_Conf_Count.xml: ><include>  <extension name="Announce Conf Count (Inline)"> <condition field="destination_number" expression="^ANNOUNCE_CONF_COUNT_PRIVATE$"> <action application="set" data="conf_count=${conference ${conference_name} list count}"/> <action application="log" data="INFO ${conf_count}"/> <action application="say" data="en number pronounced ${conf_count}"/> </condition> </extension> </include> >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > > Hi, I am trying to originate a call, using a mod command through the xml_rpc, but I am doing something wrong - not sure what it is. I post this url: http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}s… (numbers and IPs modified here) The reponse I get is this: -USAGE: <call url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] I can do other commands directly in the browser like this (such as http://1.2.3.4/webapi/show?calls) without any problems. Therefore my conclusion is that I am doing something wrong in this syntax for http. Any ideas? Thanks in advance! //Anders Don't you have to URL encode the & character? -MC On Mon, Dec 19, 2011 at 11:14 AM, Anders <vaerge(a)gmail.com> wrote: Hi, > >I am trying to originate a call, using a mod command through the >xml_rpc, but I am doing something wrong - not sure what it is. I post >this url: > >http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}s… > >(numbers and IPs modified here) > >The reponse I get is this: -USAGE: <call url> ><exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] >[<cid_name>] [<cid_num>] [<timeout_sec>] > >I can do other commands directly in the browser like this (such as >http://1.2.3.4/webapi/show?calls) without any problems. Therefore my >conclusion is that I am doing something wrong in this syntax for http. > >Any ideas? > >Thanks in advance! > >//Anders > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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

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RFC-3841 : Insert Accept-contact in sip INVITE
by Beppe Grillo 20 Dec '11

20 Dec '11

Hi, I need insert Accept-Contact in sip INVITE . I saw that RFC-3841 is supported . Do I configure something in the profile of gw to enter the Accept-contact in Sip Invite ? Regards, Giuseppe

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xml dialplan explenation
by Miha Zoubek 20 Dec '11

20 Dec '11

Hi, I need a help about understanding freeswitch. From behaviour I can see that dialplan is only for B-leg. If I make call and disable radius cdr from dialplan, this only disable b lag (a leg start and stop freeswitch still send). If I disable radius cdr in vars.xml, and than make disable_radius_stop=false, it enable packets for B leg, (a is missing, which is ok for me). My quastione is if dialplan is only for b leg? Other one, I have put disable_radius_start=false and disable_radius_stop=false in dial plan. Only stop packet is send. Does anyone know why? From my log I can see this :Action set(disable_radius_start=false) Dialplan: sofia/internal/018108500(a)xxx.xxx.xxx.xxx Action set(disable_radius_stop=false). But further than 2011-12-20 12:19:54.357167 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing 2011-12-20 12:19:54.357167 [DEBUG] mod_radius_cdr.c:165 [mod_radius_cdr] Not Sending RADIUS Start Can anyone know why? Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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