October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference timer synchronization with T1 FreeTDM - how?
by Bob Hartwig 20 Dec '13

20 Dec '13

Looking at the source for mod_conference.c, I see that the timer_name is hard-coded to "soft", i.e. the 1000 Hz. system clock. How can this work with a T1-based FreeTDM system, where the FreeTDM channels are presumably synchronized to the T1 clock source? I'm seeing some weirdness with my FreeTDM / Dahdi system, where only FreeTDM channels are hearing distortion on conference, and I'm almost certain that this is the culprit, but I have no idea how to proceed to fix it. I'm pretty new to FreeTDM, so I'm hoping that I'm missing something simple. Any thoughts? Thanks, Bob

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Gateway Registration Issues
by Aloysius Lloyd 15 Dec '13

15 Dec '13

Hi All, I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have the following gateway configuration <include> <gateway name="wwc_test"> <param name="auth-username" value="username"/> <param name="username" value="14161231234" /> <param name="password" value="password" /> <param name="proxy" value="sbc1.iristel.net" /> <param name="realm" value="iristel.net" /> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> The above configuration is not working. I have the following error in cli Error. *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234 Registration Failed with status Conflict [409]. failure #1* ==== Also carrier saying there is conflict in the contact field.Here is how they receive now. *Contact: <sip:gw+test_iristel@10.20.30.1:5080 ;transport=udp;gw=test_iristel>.* But the contact field should be like below Contact: <14161231234@10.20.30.1:5080; Any suggestions? Thanks in advance. Lloyd

5 6

old calls hanging around
by Peter Hartmann 07 Dec '13

07 Dec '13

Hi, First, thanks for Freeswitch! I'm experiencing an issue where 'show calls' returns several calls that aren't actually happening both inbound and outbound. Has anyone seen this before? Rebooting the handset (Polycom IP 550) associated with that extension has no effect so it seems in FS. freeswitch@internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 15:55:49,1381002949,sofia/external/+1347xxxxxxx@flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 09:59:34,1381154374,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 10:44:28,1381157068,sofia/external/+1646xxxxxxx@flowroute.com,CS_EXECUTE,unknown ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 10:49:05,1381157345,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 12:06:38,1381161998,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 13:37:17,1381167437,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 15:09:34,1381172974,sofia/external/+1347xxxxxxx@flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 15:41:20,1381174880,sofia/external/+1646xxxxxxx@flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 9 total. Running: FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 2013-10-02 21:30:46Z) Thanks much!

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JsSIP and wss
by Peter Steinbach 06 Dec '13

06 Dec '13

I am trying the Jssip demo with freeswitch and it works nicely with jssip 0.3.0 and the following configuration: ws_servers: 'ws://my_domain.com:8088' (My ws port is on 8088) However this only works with 'ws' but not with secure 'wss'. * In the log I can see that freeswitch sets wss-binding[:7443] * wss.pem is also present in the certs directory But with the config: ws_servers: 'wss://my_domain.com:7443' I do not even get a connection and I receive a 1006 error. Anybody has a working wss connection with Jssip and can share his config with me? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Re: [Freeswitch-users] mod_opus codec on Windows
by Jeff Leung 04 Dec '13

04 Dec '13

Perfect, I'll try it out when I have the time to do so. Thanks for going to places where I wouldn't have normally went through ;) Peter Olsson <peter(a)olssononline.se> wrote: I just wanted to notify that there is a patch available for this now (FS-5704). It has not yet been commited to git though. If anyone want to test it out, please apply the patch manually. /Peter 9 jul 2013 kl. 08:09 skrev Peter Olsson <peter(a)olssononline.se>: I think so too. I've started on a complete patch for the Windows build, hopefully I will have some time during this week to complete it. I'll submit this to Jira when finished. 2013/7/8 Michael Jerris <mike(a)jerris.com> > Are you sure symbol visibility is a problem even if these are in different > dll's ? I think this shouldn't be an issue. > > Mike > > On Jul 6, 2013, at 12:10 PM, Jeff Leung <jleung(a)v10networks.ca> wrote: > > The project was within FreeSWITCH, but the problem is because the way how > the opus lib builds, it includes silk and celt and I wasn’t going to have > FS to build all of that in tree because I was afraid the symbols from OPUS > would conflict with the symbols in CELT and SILK.**** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > users-bounces(a)lists.freeswitch.org] *On Behalf Of *Peter Olsson > *Sent:* Saturday, July 6, 2013 2:02 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] mod_opus codec on Windows**** > ** ** > Ah, ok. So you did the build of the lib outside fs, and then just pointed > to the built library?**** > ** ** > I'll start working to get this integrated in the Windows build, by > downloading the lib source files, unpacking them, and then building them.* > *** > ** ** > I will submit a Jira as soon as I have some progress. I might get back to > you with further questions.**** > ** ** > /Peter**** > > ** ** > 2013/7/6 Jeff Leung <jleung(a)v10networks.ca>**** > The project file is nothing more than having mod_opus to point to an out of > tree directory where the actual opus binary libs are built. > > The binaries I did was basically a bunch of hacks, there is quite some work > needed to be done with integrating the build process of opus with > FreeSWITCH's VS solution file > > > -----Original Message----- > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- > > users-bounces(a)lists.freeswitch.org] On Behalf Of Peter Olsson > > Sent: Saturday, July 6, 2013 12:07 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] mod_opus codec on Windows > > > > Good work! Could you also please file a Jira, and upload the project > files > and > > modifications that you've made for this to work? I'd be happy to assist > and > > get this commited. > > > > /Peter > > > > 6 jul 2013 kl. 08:20 skrev Jeff Leung <jleung(a)v10networks.ca>: > > > > > As I heard on last Wednesday's conference call, mod_opus isn't > "available" > > > on Windows, basically meaning that FreeSWITCH users can't fully > > > implement WebRTC on Windows Platforms just yet. > > > > > > I have taken the steps to have the codec successfully build on this > > > platform; However it's a pain in the neck to build as the opus library > > > generates multiple libs which includes CELT and SILK codecs (Go > figure). > > > > > > Shoot me a message off-list if you would like to use the binary I've > > > built for Win64 platforms. > > > > > > Before anybody asks for a 32-bit binary, the short answer is going to > be > no. > > > I won't be providing a 32-bit binary as I intend this binary to be > > > used in a production environment. > > > > > > > > > _________________________________________________________________________ > 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 > >

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FreeSWITCH not sending P-Asserted-Identity
by Vik Killa 29 Nov '13

29 Nov '13

I cannot get FS to send the P-Asserted-Identity header instead it always sends the P-Preferred-Identity header. According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I must set the privacy flag in order for P-Asserted-Identity to be used, so here is my dialplan entry and the SIP INVITE that is sent: <extension name="outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="lcr" data="$1"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="privacy=yes"/> <action application="privacy" data="yes"/> <action application="set" data="sip_h_Privacy=id"/> <action application="bridge" data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> </condition> </extension> ------------------------------------------------------------------------ INVITE sip:+17165555555@YYY.YYY.YYY.YYY:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp Max-Forwards: 68 From: "1000" <sip:7165551001@XX.XX.XX.XX>;tag=8U2rQ1pmXXaDQ To: <sip:+17165555555@YYY.YYY.YYY.YYY:5060> Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d CSeq: 27864844 INVITE Contact: <sip:mod_sofia@XX.XX.XX.XX:5060> User-Agent: FSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 209 X-FS-Support: update_display,send_info P-Preferred-Identity: "1000" <sip:7165551001@XX.XX.XX.XX> v=0 o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 17984 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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Sound files for mod_say_he
by Zev Spitz 28 Nov '13

28 Nov '13

According to the link at http://wiki.freeswitch.org/wiki/Mod_say_he, the sound files for mod_say_he will be available soon. The page was last updated 4 April 2011. Has anything happened since? Thanks in advance, Zev Spitz

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recommendations for Wifi SIP phones?
by Nathan Neulinger 26 Nov '13

26 Nov '13

Anyone got any recommendations for decent quality wifi sip phones? Obviously the 'android+soft client' model will work, but what about actual handsets built for the task. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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SDReporter Integration with FS
by Deon Vermeulen 19 Nov '13

19 Nov '13

Hi I've successfully setup SDReporter, FreeSWITCH to export CDRs using cdr_csv with the Asterisk template. I've setup a cron to "automatically" push the csv file(s) to SDReporter $HOME_DIR/SDReporter/CDRConverter-4.0.8/data/INPUT/Asterisk/1/ I've tried different options in the format of the CDRs ,but no matter what I do I keep getting an " CDRFieldChannelAsterisk(String): wrong Asterisk channel. " ERROR. I've removed the "${channel_name}","${bridge_channel}" variables, but still I get the above error. I currently run FreeSWITCH Version 1.5.6b+git~20131018T052734Z~e0054af96f. Has anyone successfully got SDReporter working with FS? Thanks for any assistance. -- Kind Regards

3 5

No ringback tone on bridge
by mbo 19 Nov '13

19 Nov '13

Hello, we receive calls from gateway A and bridge them using gateway B to the destination number . <extension name="test"> <action application="bridge" data="sofia/gateway/B/+49123456789" /> </extension> The problem is, that we do not hear a ringtone at all. If we put a "ring_ready" before the bridge, we can hear a ringtone immediately, but then the callers hear a ringtone even if the bridged destination is not ringing yet. <extension name="test"> <action application=ring_ready /> <action application="bridge" data="sofia/gateway/B/+49123456789" /> </extension> Is there a solution that the callers hears a ring tone as soon as the destination is ringing, but not before ? Thanks Markus

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