October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using SIP outbound leg error
by stephen＠picardogroup.com 07 Oct '12

07 Oct '12

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Absolute Codec String
by Nick Vines 06 Oct '12

06 Oct '12

I am trying to set the codec when I bridge. I have late negotiation set to true, and since I am trying to use G729, I set proxy_media=true as well. I am using this in my dialplan (I have tried with and without "nolocal:" with no change): <action application="bridge" data="{absolute_codec_string=G729}sofia/gateway/gateway/${dialed_number}"/> But in the invite to the gateway, I am seeing the codecs offered by the device: m=audio 24960 RTP/AVP 18 0 2 8 9 96 97 98 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 I am troubleshooting right now, so I am able to specify on the phone "Use Only Preferred", in which case I see only to following offered: a=rtpmap:18 G729/8000 Thanks, Nick

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More recent Skype client for mod_skypeopen
by Emrah 06 Oct '12

06 Oct '12

Hi list, It's great to have a 16khz audio sample when using Skype via FS… However this does not apply for sessions with mobile versions of Skype which only embed SILK, not available in the Skype version currently used by Skypeopen. How complicated is it to use a more recent version of the Skype client? Is it even possible? Cheers and thanks in advance, Emrah

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PLEASE HELP with Callcentric disaster - 48 hours down now
by Mario G 06 Oct '12

06 Oct '12

Callcentric had a DOS attack since Friday that had it down until late Sat, still limping. Their instructions seem to work for everyone but FreeSwitch, I can only get inbound calls not outbound, better than nothing. They don't know FreeSwitch. I used their instructions for Asterisk which are to change the host, and outbound proxy. But outbound calls just sit and timeout. They say the registrar and domain should remain callcentric.com, but the Asterisk instructions say to set host and outboundproxy to srv.callcentric.com. I have been glued to the system since Friday trying to get phones back up, I tried a few combos but no luck, any help greatly appreciated! Mario G <param name="realm" value="srv.callcentric.com"/> <param name="proxy" value="srv.callcentric.com"/> <param name="outbound-proxy" value="srv.callcentric.com"/>

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xml cdr help
by Lloyd Aloysius 06 Oct '12

06 Oct '12

Hi, I am directly writing xml cdr into the database. In my configuration <param name="log-b-leg" value="false"/> But freeswitch post 2 cdr records into the database. 1. How to avoid the b-leg 2. Is there a way to flag the records for a-leg or b-leg Thanks in advanced. Lloyd * *

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Presence Not Working with sofia_contact bridging to multiple registrations
by Brett Wilson 06 Oct '12

06 Oct '12

I am using multiple registrations, and had initially kept the bridge calls using the 'user/' syntax. Presence was working with this, however i noticed that when I used a simultaneous brige separated by commas, some of the extensions would ring one registration and not the other. So I switched to the sofia_contact bridge method and this fixed the issue. However, when using sofia_contact to bridge, I noticed that presence does not seem to operate. I turned on debug presence and the only thing I see that may relate to an error is: 2012-10-06 13:13:50.183942 [WARNING] sofia_presence.c:4222 check_subs: external is passive, skipping 2012-10-06 13:14:11.153694 [ERR] sofia_presence.c:3343 DELTA 120 2012-10-06 13:14:11.153694 [ERR] sofia_presence.c:3395 check subs sql: select count(*) from sip_subscriptions where call_id='4agoj24x@snom' and hostname='euro-pbx' and profile_name='internal' [1] 2012-10-06 13:14:11.153694 [ERR] sofia_presence.c:3417 re-subscribe event message-summary, sql: update sip_subscriptions set expires=1349543771,contact='"Shop" <sip:102@192.168.1.13:3167;transport=udp;line=5r26s2>' where hostname='euro-pbx' and profile_name='internal' and call_id='4agoj24x@snom' and profile_name='internal' 2012-10-06 13:14:12.183897 [ERR] sofia_presence.c:1982 SEND PRES NOTIFY: file[sofia_presence.c] func[sofia_presence_mwi_callback] line[2896] profile[internal] via[] ip[192.168.1.13] port[3167] route[sip:user@192.168.1.13:3167;transport=udp;line=1k5l6w] contact["Front" <sip:101@192.168.1.13:3167;transport=udp;line=1k5l6w>] to["Front" <sip:101@192.168.1.200>;tag=TQK1cNoQcA2H] from["Front" <sip:101@192.168.1.200>;tag=06l8nn] url[sip:101@192.168.1.13:3167;transport=udp;line=1k5l6w] call_id[9mgy45bh@snom] expires_str[] event[message-summary] ct[application/simple-message-summary] pl[Messages-Waiting: no Message-Account: sip:101@192.168.1.200 Voice-Message: 0/4 (0/0) ] -- ******************************************* BRETT WILSON IT DEPARTMENT LAUNCH 3 VENTURES, LLC 134 Myer Street Hackensack, NJ 07601 PHONE: 877.878.9134 FAX: 646.536.3866 EMAIL: Brett.Wilson(a)launch3.net AOL IM: Brett.Wilson(a)launch3.net www.Launch3.net www.Launch3telecom.com *******************************************

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How to disable key ahead
by JP 06 Oct '12

06 Oct '12

Hi, How do I disable "key ahead" while executing "play-and-get-digits". Basically I want the user to hear the whole prompt before typing dtmf. Thanks in advance, JP

3 2

Yes We Know the website is down heh....
by Ken Rice 06 Oct '12

06 Oct '12

BKW and his crew are working on the website... It¹ll be back sooner or later... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Friday Free For All
by Ken Rice 05 Oct '12

05 Oct '12

The Don¹t Forget the Friday Free For All at sip:888@conference.freeswitch.org Whens it start? When do you want it to start? Officially 4PM EST... However you can join and leave at a whim... There were complaints about the moderation feature being busted when turned off those have been resolved... Seems the conference monitoring bots were arguing over whats going to happen when someone pressed 0... Come on and Join us, invite your friends, invite voip users from around the world... And of course all the usual means for joining on Wed worth all the time K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

1 0

Could not locate channel type sofia
by Lloyd Aloysius 05 Oct '12

05 Oct '12

FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) I have trouble dialing between users. is this a bug? *Dial plan* <document type="freeswitch/xml"> <section name="dialplan" description="FreeSWITCH Dialplan"> <context name="5ele.mydomain.ca"> <extension name="Local_Extension"> <condition field="destination_number" expression="^(24)$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="{sip_invite_domain=5ele.mydomain.ca}user/ dick(a)5ele.mydomain.ca"/> </condition> </extension> </context> </section> *Error...* 2012-10-04 21:57:31.143580 [DEBUG] switch_core_state_machine.c:209 sofia/sipinterface_1/klaus(a)5ele.mydomain.ca Standard EXECUTE EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.ca set(dialed_extension=24) 2012-10-04 21:57:31.143580 [DEBUG] mod_dptools.c:1319 sofia/sipinterface_1/ klaus(a)5ele.mydomain.ca SET [dialed_extension]=[24] EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.caexport(dialed_extension=24) 2012-10-04 21:57:31.143580 [DEBUG] switch_channel.c:1118 EXPORT (export_vars) [dialed_extension]=[24] EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.caset(ringback=%(2000,4000,440,480)) 2012-10-04 21:57:31.143580 [DEBUG] mod_dptools.c:1319 sofia/sipinterface_1/ klaus(a)5ele.mydomain.ca SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.ca set(call_timeout=30) 2012-10-04 21:57:31.143580 [DEBUG] mod_dptools.c:1319 sofia/sipinterface_1/ klaus(a)5ele.mydomain.ca SET [call_timeout]=[30] EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.caset(hangup_after_bridge=true) 2012-10-04 21:57:31.143580 [DEBUG] mod_dptools.c:1319 sofia/sipinterface_1/ klaus(a)5ele.mydomain.ca SET [hangup_after_bridge]=[true] EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.caset(continue_on_fail=true) 2012-10-04 21:57:31.143580 [DEBUG] mod_dptools.c:1319 sofia/sipinterface_1/ klaus(a)5ele.mydomain.ca SET [continue_on_fail]=[true] EXECUTE sofia/sipinterface_1/klaus(a)5ele.mydomain.cabridge({sip_invite_domain= 5ele.mydomain.ca}user/dick(a)5ele.mydomain.ca) 2012-10-04 21:57:31.143580 [DEBUG] switch_channel.c:1072 sofia/sipinterface_1/klaus(a)5ele.mydomain.ca EXPORTING[export_vars] [dialed_extension]=[24] to event 2012-10-04 21:57:31.143580 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables 2012-10-04 21:57:31.143580 [DEBUG] switch_event.c:1569 Parsing variable [sip_invite_domain]=[5ele.mydomain.ca] 2012-10-04 21:57:31.163577 [DEBUG] switch_channel.c:1072 sofia/sipinterface_1/klaus(a)5ele.mydomain.ca EXPORTING[export_vars] [dialed_extension]=[24] to event 2012-10-04 21:57:31.163577 [DEBUG] switch_ivr_originate.c:2005 Parsing global variables 2012-10-04 21:57:31.163577 [DEBUG] switch_event.c:1569 Parsing variable [presence_id]=[dick(a)5ele.mydomain.ca] 2012-10-04 21:57:31.163577 [ERR] switch_core_session.c:408 Could not locate channel type sofia 2012-10-04 21:57:31.163577 [NOTICE] switch_ivr_originate.c:2591 Cannot create outgoing channel of type [ sofia] cause: [CHAN_NOT_IMPLEMENTED] 2012-10-04 21:57:31.163577 [DEBUG] switch_ivr_originate.c:3508 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] 2012-10-04 21:57:31.163577 [NOTICE] switch_ivr_originate.c:2591 Cannot create outgoing channel of type [user] cause: [CHAN_NOT_IMPLEMENTED] 2012-10-04 21:57:31.163577 [DEBUG] switch_ivr_originate.c:3508 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] 2012-10-04 21:57:31.163577 [INFO] mod_dptools.c:3027 Originate Failed. Cause: CHAN_NOT_IMPLEMENTED 2012-10-04 21:57:31.163577 [NOTICE] switch_core_state_machine.c:262 sofia/sipinterface_1/klaus(a)5ele.mydomain.ca has executed the last dialplan instruction, hanging up. 2012-10-04 21:57:31.163577 [DEBUG] switch_channel.c:2950 (sofia/sipinterface_1/klaus(a)5ele.mydomain.ca) Callstate Change RINGING -> HANGUP 2012-10-04 21:57:31.163577 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/sipinterface_1/klaus(a)5ele.mydomain.ca [CS_EXECUTE] [NORMAL_CLEARING] 2012-10-04 21:57:31.163577 [DEBUG] switch_channel.c:2973 Send signal sofia/sipinterface_1/klaus(a)5ele.mydomain.ca [KILL] 2012-10-04 21:57:31.163577 [DEBUG] switch_core_session.c:1210 Send signal sofia/sipinterface_1/klaus(a)5ele.mydomain.ca [BREAK] 2012-10-04 21:57:31.163577 [DEBUG] switch_core_state_machine.c:453 (sofia/sipinterface_1/klaus(a)5ele.mydomain.ca) State EXECUTE going to sleep

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