July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Different REGISTER and INVITE From:
by dave9876＠gmx.com 21 Jul '11

21 Jul '11

Hi, Thanks! I thought I had tried that, but apparently not properly. The next problem was that I needed P-Asserted-Identity set to "12345_01". I added "{sip_h_P-Asserted-Identity='sip:12345_01@gw.example.com',..." and it works fine. Thanks a lot! Dave ----- Original Message ----- From: Steven Ayre Sent: 07/20/11 05:55 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Different REGISTER and INVITE From: Try setting this in the gateway config:    <param name="caller-id-in-from" value="true"/> You'll need to do 'sofia profile NAME killgw NAME', 'sofia profile NAME rescan'. It'll then use the caller id in the From for the INVITE, so this should then work: originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 -Steve On 20 July 2011 08:34, <dave9876(a)gmx.com> wrote: > Hi, > I'm trying to test a gateway with FS. > > The GW requires me to register with "From: 12345_01(a)gw.example.com", but when calling I need to have "From: +12345(a)gw.example.com" (with "+" and without "_01") > > If I try from the console: > originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 >ld > I still get "From: 12345_01(a)gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile. > > Any ideas on how to force a different "From" on the INVITE? > > I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead. > > Any help is much appreciated. > > > Dave > _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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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Google voice terminating outbound calls
by Rob Hill 20 Jul '11

20 Jul '11

I'm using mod dingaling to connect via Google Voice. Incoming and outgoing calls were working fine until just a few days ago, now outgoing calls are terminated after ~15 seconds. Incoming calls are still fine. Here is the disconnect info : 2011-07-20 17:36:20.485777 [INFO] libdingaling.c:1371 SecRECV: ------------------------------------------------------------------------------- <iq from="+1312xxxxxxx(a)voice.google.com/srvres-MTAuMTc3LjEyMi4xNDo5ODgz" to="r(a)gmail.com/talkCC63ED4F" id="jingle:10.177.122.14-2416924:1:3F7BB3A2" type="set"> <ses:session type="terminate" id="1514126677" initiator="r(a)gmail.com/talkCC63ED4F" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:timeout></ses:timeout> <ses:text>Session timed out</ses:text> </ses:reason> <pho:recipient-unavailable xmlns:pho="http://www.google.com/session/phone"></pho:recipient-unavailable> </ses:session> </iq> To make matters more confusing I've tested my configuration on two different hosts and outgoing calls work fine on the second host (using the exact same configuration files). Any ideas where this might be going wrong? a O

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ClueCon 2011 - Sharing rooms
by Michael Collins 20 Jul '11

20 Jul '11

Hello all! It seems that the Sofitel is getting booked up rapidly and it's becoming difficult to find rooms. If you are looking for someone with whom you may share a room, or you have a room that you would like to share, please email me off list. Thanks, Michael

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FreeTDM viewer
by Valery Kalinin 20 Jul '11

20 Jul '11

I'm write small web utility for FreeTDM channel statuses view: FreeTDM viewer Web-based online view FreeTDM spans & channels statuses. Download from: https://sites.google.com/site/freeswitched/home/downloads Installation guide: Upload files (ftdmv.php & jquery-1.6.1.min.js) to your web-server with PHP support. Check modules.conf and event_socket.conf for enable module mod_event_socket Change IP address, port, password for FreeSWITCH server and FreeTDM spans in ftdmv.php file Enter in browser http://you-server-name/ftdmv.php

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SIP Presence Subscription Management
by Dunkel, Christopher 20 Jul '11

20 Jul '11

I'm having an issue with SIP presence through Freeswitch and I'm hoping that someone might be able to help me determine if my problem is with Freeswitch or with the client(s) I'm using. The system I'm using includes Freeswitch (version 1.0.6) running on a Linux box and two soft phone clients running on iPhone and Android (PJSip on iPhone and Linphone on Android). The problem that I'm having appears the same on both clients, which leads me to believe that this may be a Freeswitch config problem. The clients are able to send a SUBSCRIBE message and receive a NOTIFY immediately when one of the subscribed users registers with Freeswitch. They also receive the offline NOTIFY when the user unregisters from Freeswitch. However, an online NOTIFY message is not sent when the user comes back online. It seems as if Freeswitch is clearing all presence subscriptions for a particular user when they unregister from the server. Is there any way to prevent Freeswitch from clearing these subscribes unless specifically directed to do so? Also, on a possibly related subject, I am only able to receive "online" and "offline" presence notifications. I only receive "offline" presence notifications when a user unregisters from Freeswitch. Attempting to send any other presence value from one of the clients (i.e. "busy", "be right back", "offline", etc...) only results in an "online" notification being sent. Has anyone else seen similar behavior? Thanks, Chris

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Different REGISTER and INVITE From:
by dave9876＠gmx.com 20 Jul '11

20 Jul '11

Hi, I'm trying to test a gateway with FS. The GW requires me to register with "From: 12345_01(a)gw.example.com", but when calling I need to have "From: +12345(a)gw.example.com" (with "+" and without "_01") If I try from the console: originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 I still get "From: 12345_01(a)gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile. Any ideas on how to force a different "From" on the INVITE? I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead. Any help is much appreciated. Dave

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FreeSWITCH Conference Call Today
by Michael Collins 20 Jul '11

20 Jul '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_20 We have a special guest: Alfred from the baresip project is coming in to talk about his cool new SIP client! Come join us today at 1PM eastern/10AM pacific/1700 UTC. -Michael

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API conference invite group
by Fabio Bigliardi 20 Jul '11

20 Jul '11

Hi all, I would like to be able to invite a group to a conference through an API command. In xml dialplan this can be achieved through the following action: <action application="conference_set_auto_outcall" data="${group_call(my_group}"/> In the API, I suppose the right command is "dial": api conference my_conf dial <group dial string> But how can I specify group dial string? Thank you for your support. Best regards, Fabio Bigliardi

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absolute_codec_string just process one codec
by Saeed Ahmed 20 Jul '11

20 Jul '11

Hi, If i set absoluteye_codec_string in dialplan with multiple codecs, like G729,PCMA,PCMU, then just first codec will be processed, rest will be ignored. Example: {absolute_codec_string=${ep_codec_string}, i even tried using it like: absolute_codec_string='${ep_codec_string}' or absolute_codec_string=\'${ep_codec_string}\' or absolute_codec_string=\\'${ep_codec_string}\\' (as suggested by bkw_ on IRC) Console log shows like that: EXECUTE sofia/profile/1234567(a)1.2.2.3 bridge({absolute_codec_string=PCMA@8000h@20i@64000b,PCMU@8000h@20i@64000b,G729@8000h@20i@8000b,leg_time_out= 10,process_cdr=b_only}[SupplierPrefix=]sofia/fstest/456789(a)7.2.1.4) 2011-07-20 11:03:59.448455 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable [absolute_codec_string]=[PCMA@8000h@20i@64000b] 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable [leg_time_out]=[10] 2011-07-20 11:03:59.448455 [DEBUG] switch_event.c:1170 Parsing variable [process_cdr]=[b_only] Am i doing something wrong or should i open a JIRA? -- Kind Regards Saeed Ahmed

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X-PRE-PROCESS and Dialplan XML Module
by Daniel Knoll 20 Jul '11

20 Jul '11

Hi FreeSwitch Group, i'm a little bit confused about the X-PRE-PROCESS and XML Dialplan Module. I'm using a dynamic dialplan to include in FreeSwitch with <X-PRE-PROCESS cmd="exec" data="wget -qO - http://www.domain.com/dialplan.php" /> This don't work if i place X-PRE-PROCESS in at the top from public.xml outsite the <include> tag <include> <context name="public"> <extension name="123456789"> <condition field="destination_number" expression="123456789"> <action application="set" data="hangup_after_bridge=true"/> <action application="javascript" data="app.js"/> </condition> </extension> </context> </include> but this works if X-PRE-PROCESS is in the <include> tag, but it's not clean XML, because the root is missing <extension name="123456789"> <condition field="destination_number" expression="123456789"> <action application="set" data="hangup_after_bridge=true"/> <action application="javascript" data="app.js"/> </condition> </extension> My Question is what is the right anatomy of a XML to include my dialplan in the public.xml with X-PRE-PROCESS and exec wget ? Thanks for your answers. Daniel

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