September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to get the content of SIP headers
by Jon Bruel 18 Sep '08

18 Sep '08

As I see it, the NOTIFY methods mentioned are used when there is a call, i.e. at the earliest when the phone has responded with a RINGING or a similar response to the INVITE. By the way, the response to the INVITE also includes a list of events (Header: Allow-Events) which the phone responds to. I have not seen other instances where the NOTIFY To-header needs to be tagged, but my experience is limited and not bound to the RFCs, so don't rely on me! On Sep 18, 2008, at 4:53 AM, Jon Bruel wrote: > Thank you for your replies. In checking the SIP messaging sequence, > I saw that you are right about the source of the tag to the To- > header: it is the phone, which as a response to the INVITE, adds the > tag (and not, as I wrote, the FreeSWITCH). Still, we need a way to > read it, and not the least to send NOTIFYs with the right tag. I > have raised two issues on the Jira, maybe the category is wrong. > > One of the tricky things to sort out is how to handle it in the case > of bridging to a list of devices, and bridging to a user with > multiple registrations. The NOTIFY method should include measures to > send a NOTIFY to a specific phone among the many phones responding. > > Thanks again for your excellent responsiveness ? it adds the trust > of the FreeSWITCH. Jon. > So we would only have the to tag in the case of a device with a call in progress, what about for just a registration, how would you determine between the devices? Mike

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Re: [Freeswitch-users] billing platform
by Mitul Limbani 18 Sep '08

18 Sep '08

Hello, We did actually start writing a billing platform for FreeSwitch back when it didnt even have the Socket API as a C module to FreeSwitch (compiled on linux platform as such), but the core developer moved away and the project is in deserted mode. So we do have a lead of it, if someone wants to take it up from there and create a full fledged billing platform, I can provide the work that we had done back then. Any one interested to continue with that particular project, I shall be glad, I can also look at funding the project partially, but the only condition is that it has to be released as Open Source (MPL) providing us capabilities to sub-license it under dual mode. If anyone interested, do let me know. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (R), http://www.enterux.com/ On Thu 18/09/08 21:30 , xbipin <bipin(a)xbipin.com> wrote: > are u interested in a paid implementation as im willing to post a > bounty for > it if there r others like us willing to use it with freeswitch. > wikipbx is > also good for a start but one thing i dont understand is if > freeswitch is > made for cross platform then y r other projects related to it > limiting it to > just linux? > Jair Santos wrote: > > > > I've been trying to install. The problem is the lack of > documentation on > > how to integrate with FS. > > > > Jair Santos > > > > > > > > > > > > > >> -----Original Message----- > >> From: > >> [] On > >> Behalf Of xbipin > >> Sent: Thursday, September 18, 2008 3:25 AM > >> To: > >> Subject: [Freeswitch-users] billing platform > >> > >> > >> > >> has any1 been able to use ASTPP billing with freeswitch and > >> if so then my next question, is it possible to use it on a > >> win2k3 platform and not linux as i have freeswitch running on > >> a win2k3 server since a few days and works fine, i just need > >> a prepaid type of script that basically stops a call once its > >> credit is over. i will add the credit etc to the account > manually. > >> > >> im ready to post a bounty if any1 willing to do it. the > >> script just needs to end the call once it computes that > >> remaining time is 0 based on available time which is based on > >> the cost per minute and available credit and avoid calling > >> again which needs to be done as soon as call is placed so the > >> call is never connected if account has no credit. > >> -- > >> View this message in context: > >> http://www.nabble.com/billing-platform-tp19549891p19549891.html > >> Sent from the Freeswitch-users mailing list archive at > Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > >> itch-users > >> http://www.freeswitch.org > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > View this message in context: > http://www.nabble.com/billing-platform-tp19549891p19556187.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/

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Can't find user (Anonymous calls)
by henkoegema 18 Sep '08

18 Sep '08

When I make a call (from SJphone) to sip:henk@oegema.com than I can see in Freeswitch: 2008-09-18 10:16:19 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [anonymous(a)oegema.com] 2008-09-18 10:16:19 [DEBUG] sofia_reg.c:553 sofia_reg_handle_register() send challange for [henk(a)oegema.com] How to I answer this call for henk(a)oegema.com at extension 2000 ? (also calling 2000 (which is a registered number in FS) gives me: 2008-09-18 10:25:39 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [anonymous(a)oegema.com] 2008-09-18 10:25:39 [DEBUG] sofia_reg.c:553 sofia_reg_handle_register() send challange for [2000(a)oegema.com] ) Henk -- View this message in context: http://www.nabble.com/Can%27t-find-user-%28Anonymous-calls%29-tp19548337p19… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Ringback disappeared
by Ivan C Myrvold 18 Sep '08

18 Sep '08

I have for more than 1 year now sent my inbound calls to an application running on another machine listening to FreeSWITCH's outbound socket. I have this in my dialplan: <action application="socket" data="192.168.207.101:8084 async full"/> In my external application, I look at the called number, and redirects it back to FreeSWITCH with the following messages: myevents sendmsg call-command: execute execute-app-name: set execute-app-arg: ringback=%(400,200,400,450);%(400,2200,400,450) sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/imyrvold.dyndns.org/100%imyrvold.dyndns.org Now when anyone calls me, they don't hear the ringtone before I answer the call. This used to work before, and I am wondering why this suddenly don't work anymore. The only thing which have changed, is that I have changed my voip phone provider. I can also see in the events from FreeSWITCH, in the variable_ringback, that it is set: variable_endpoint_disposition: EARLY%20MEDIA variable_current_application: set variable_ringback: %25(400,200,400,450)%3B%25(400,2200,400,450) Application: bridge Application-Data: sofia/imyrvold.dyndns.org/100%25imyrvold.dyndns.org Event-Name: CHANNEL_EXECUTE Can the voip phone provider have shut this off? Ivan

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Re: [Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?
by Jon Bruel 18 Sep '08

18 Sep '08

Thank you for your replies. In checking the SIP messaging sequence, I saw that you are right about the source of the tag to the To-header: it is the phone, which as a response to the INVITE, adds the tag (and not, as I wrote, the FreeSWITCH). Still, we need a way to read it, and not the least to send NOTIFYs with the right tag. I have raised two issues on the Jira, maybe the category is wrong. One of the tricky things to sort out is how to handle it in the case of bridging to a list of devices, and bridging to a user with multiple registrations. The NOTIFY method should include measures to send a NOTIFY to a specific phone among the many phones responding. Thanks again for your excellent responsiveness - it adds the trust of the FreeSWITCH. Jon.

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Test Proxy Media
by msp 18 Sep '08

18 Sep '08

Hi all, I have enabled proxy media from dialplan. After that, I can make calls same as it done before without enabling proxy media. So, how can i test that my calls are in proxy media mode after enabling "proxy-media" mode ? Thanks, MShehzad

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transcoding / g729 / g723.1
by Cristian Talle 18 Sep '08

18 Sep '08

How hard would it be to add support for g729/g723.1 transcoding using the TC400B card (digium)? It's all part of the zaptel driver. Cristian

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Accessing the variable shown by the application "info"?
by Jon Bruel 18 Sep '08

18 Sep '08

I have tried to access the variables shown by the application info, and the variable name to be used is generally not the same as the name listed by info. I have listed nearly 100 variables below, where I guessed the "real" names, but a number are still missing, and I would like to know what the "real" names are or how I derive the right names. Some of the problem-variables, though, for instance Channel-Read-Codec-Name are also found further down in the list as variable_read_codec, which can be read alright, and the presence of two variable names which may have the same value seems confusing for a dummy as me. The info-names have been taken - as an example - from the info result of a call that has been diverted by the phone. The list below shows my guess (via a javascript), and when the third column shows FALSE, the guess need to be corrected. In some cases, the reason for the "FALSE" may be related to the fact that the variable is unassigned: Info-name---Guessed name---Result Channel-State---Channel_State---FALSE Channel-State-Number---Channel_State_Number---FALSE Channel-Name---Channel_Name---OK Unique-ID---Unique_ID---FALSE Call-Direction---Call_Direction---FALSE Answer-State---Answer_State---FALSE Channel-Read-Codec-Name---Channel_Read_Codec_Name---FALSE Channel-Read-Codec-Rate---Channel_Read_Codec_Rate---FALSE Channel-Write-Codec-Name---Channel_Write_Codec_Name---FALSE Channel-Write-Codec-Rate---Channel_Write_Codec_Rate---FALSE Caller-Username---Username---OK Caller-Dialplan---Dialplan---OK Caller-Caller-ID-Name---Caller_ID_Name---OK Caller-Caller-ID-Number---Caller_ID_Number---OK Caller-Network-Addr---Network_Addr---OK Caller-Destination-Number---Destination_Number---OK Caller-Unique-ID---Unique_ID---FALSE Caller-Source---Source---OK Caller-Context---Context---OK Caller-RDNIS---RDNIS---OK Caller-Channel-Name---Channel_Name---OK Caller-Profile-Index---Profile_Index---OK Caller-Channel-Created-Time---Channel_Created_Time---FALSE Caller-Channel-Answered-Time---Channel_Answered_Time---FALSE Caller-Channel-Hangup-Time---Channel_Hangup_Time---FALSE Caller-Channel-Transfer-Time---Channel_Transfer_Time---FALSE Caller-Screen-Bit---Screen_Bit---FALSE Caller-Privacy-Hide-Name---Privacy_Hide_Name---FALSE Caller-Privacy-Hide-Number---Privacy_Hide_Number---FALSE variable_sip_received_ip---sip_received_ip---OK variable_sip_received_port---sip_received_port---OK variable_sip_authorized---sip_authorized---OK variable_sip_mailbox---sip_mailbox---OK variable_sip_auth_username---sip_auth_username---OK variable_sip_auth_realm---sip_auth_realm---OK variable_mailbox---mailbox---OK variable_user_name---user_name---OK variable_domain_name---domain_name---OK variable_record_stereo---record_stereo---OK variable_accountcode---accountcode---OK variable_user_context---user_context---OK variable_effective_caller_id_name---effective_caller_id_name---OK variable_effective_caller_id_number---effective_caller_id_number---OK variable_caller_domain---caller_domain---OK variable_sip_from_user---sip_from_user---OK variable_sip_from_uri---sip_from_uri---OK variable_sip_from_host---sip_from_host---OK variable_sip_from_user_stripped---sip_from_user_stripped---OK variable_sip_from_tag---sip_from_tag---OK variable_sofia_profile_name---sofia_profile_name---OK variable_sofia_profile_domain_name---sofia_profile_domain_name---OK variable_sip_req_params---sip_req_params---OK variable_sip_req_user---sip_req_user---OK variable_sip_req_uri---sip_req_uri---OK variable_sip_req_host---sip_req_host---OK variable_sip_to_params---sip_to_params---OK variable_sip_to_user---sip_to_user---OK variable_sip_to_uri---sip_to_uri---OK variable_sip_to_host---sip_to_host---OK variable_sip_contact_params---sip_contact_params---OK variable_sip_contact_user---sip_contact_user---OK variable_sip_contact_port---sip_contact_port---OK variable_sip_contact_uri---sip_contact_uri---OK variable_sip_contact_host---sip_contact_host---OK variable_channel_name---channel_name---OK variable_sip_call_id---sip_call_id---OK variable_sip_user_agent---sip_user_agent---OK variable_sip_via_host---sip_via_host---OK variable_sip_via_port---sip_via_port---OK variable_sip_via_rport---sip_via_rport---OK variable_presence_id---presence_id---OK variable_sip_h_P-Key-Flags---sip_h_P_Key_Flags---FALSE variable_switch_r_sdp---switch_r_sdp---OK variable_remote_media_ip---remote_media_ip---OK variable_remote_media_port---remote_media_port---OK variable_write_codec---write_codec---OK variable_write_rate---write_rate---OK variable_endpoint_disposition---endpoint_disposition---OK variable_dialed_ext---dialed_ext---OK variable_transfer_ringback---transfer_ringback---OK variable_call_timeout---call_timeout---OK variable_hangup_after_bridge---hangup_after_bridge---OK variable_continue_on_fail---continue_on_fail---OK variable_dialed_user---dialed_user---OK variable_dialed_domain---dialed_domain---OK variable_sip_redirect_contact_user_0---sip_redirect_contact_user_0---OK variable_sip_redirect_contact_host_0---sip_redirect_contact_host_0---OK variable_sip_h_Referred-By---sip_h_Referred_By---FALSE variable_sip_refer_to---sip_refer_to---OK variable_max_forwards---max_forwards---OK variable_originate_disposition---originate_disposition---OK variable_read_codec---read_codec---OK variable_read_rate---read_rate---OK variable_open---open---OK variable_use_profile---use_profile---OK variable_current_application---current_application---OK

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Turn on CN packet generation
by Tamas 18 Sep '08

18 Sep '08

Hello, Is there a way to let FS send RTP CN packets by itself? What I mean is for example when I turn on VAD for an outbound call and allow CN (suppress_cng=false) than lets FS sends out CN packets even when party A is not sending them. <action application="bridge" data="{rtp_enable_vad_out=true,suppress_cng=false}sofia/.... Regards, Tamas

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mod_openzap
by Evgeniy Zolotov 18 Sep '08

18 Sep '08

In a directory libs/openzap/src/isdn [olej@opos8 isdn]$ ls *.c 5ESSmes.c DMSStateNT.c mfifo.c Q921.c Q931mes.c 5ESSStateNT.c DMSStateTE.c nationalmes.c Q931api.c Q931StateNT.c 5ESSStateTE.c EuroISDNStateNT.c nationalStateNT.c Q931.c Q931StateTE.c DMSmes.c EuroISDNStateTE.c nationalStateTE.c Q931ie.c Q932mes.c - we see 2 files with definitions of responses to PRI messages : Q931StateNT.c & Q931StateTE.c - "net side" & "local side" 1. If we connect 2 FreeSWITCHs with PRI channel we should specify (into mod_openzap configurations) NT for one of them and TE for another. In which way we are able to do this??? 2. Is it a "span" parameter : <param name="mode" value="user" /> in openzap.conf.xml? What significance needs to be written in a "value" for the "net side" ?

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