- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch for Windows - command history too short?
by Jeff Lenk 04 Apr '10

04 Apr '10

No there is no config setting for that yet - feature does not exist yet -----Original Message----- From: Frank Church Sent: 4/4/2010 1:16:34 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch for Windows - command history too short? I am using svn 17048 for Windows and the command history is too short and doesn't persist between restarts. Are there some configuration settings to fix that? -- Frank Church ======================= http://devblog.brahmancreations.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

1 0

Contrib modules
by mayamatakeshi 04 Apr '10

04 Apr '10

Hi, I have updated (svn) to trunk (r. 17188) and now I don't see the contrib folder anymore (I was using mod_xml_odbc from there). What happened? regards, takeshi

2 1

domain-wise context
by Jayesh Nambiar 04 Apr '10

04 Apr '10

Hi, Sorry for sounding so impatient, the anxiety only grew because before posting it to the list I spent a week on reading all the documentation available on and around this topic FS site and mailing lists. I really appreciate and am thankful for your suggestions. I'll try out the suggestions given by you. Is there a way that we can compare the domain name in the dialplan using regular expressions. Is there a value in condition tag that can be used to compare this, something like "extension_number". Or should I store the domain value in some variable(something like sip_h_<domain_name>) and compare that variable to take the call to a different context? Thanks, --- Jayesh ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sat, 3 Apr 2010 09:11:50 -0600 > Subject: Re: [Freeswitch-users] domain-wise context > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension > > The first one transfers the call to the desired exten/dialplan/context and > the 2nd one executes the specified extension in a similar manner and returns > to the same point in the dp. > > You make one inbound context and use routing logic from there to decide > which context to transfer to. > > Next time, please have a little more patience, I don't like it when people > reply to themselves on the list asking why nobody answered when their > question is only unanswered for 2 days especially during a holiday weekend. > > > > On Sat, Apr 3, 2010 at 1:43 AM, Jayesh Nambiar <jayesh.voip(a)gmail.com>wrote: > >> I expected at east one reply saying that the question is stupid, and the >> solution is simple !! >> Any folks who can help me understand only how to achieve this in FS which >> is acheivable in asterisk as follows: >> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com", >> the call goes to context "mydomain" in dialplan) >> domain = yourdomain.com, yourdomain (if any call has domain as " >> yourdomain.com", the call goes to context "yourdomain" in dialplan) >> >> These calls can come from anywhere, in my case it comes from an Opensips >> instance !! >> >> Thanks for any replies :) >> >> --- Jayesh >> >> >> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip(a)gmail.com>wrote: >> >>> Hi All, >>> I am quite very new to freeswitch and I am kind of playing with it to >>> understand it better. >>> I am primarily using Opensips as registrar and SIP Proxy and intend to >>> use FS as media server handling voicemails, IVR, Announcements, MeetMe etc. >>> My Opensips is a multi-domain setup and I wish to have all the configuration >>> of media-capabilties segregated domain-wise in the FS too. >>> >>> For eg: When a call for user(a)domain1.com needs to go to voicemail, I >>> redirect that call to FS IP address keeping the URI intact. I add the >>> mailbox number as a header as X-Mailbox and have FS extract it and go to >>> appropriate mailbox. Similarly when a call for user(a)domain2.com needs to >>> go to voicemail I do the same thing. >>> The requirement is I want to maintain the dialplans for each domains >>> separately. Thus if call from Opensips comes to FS with domain as domain1, >>> the call should go to dialplan context domain1 and similarly if call from >>> Opensips comes to FS with domain2 the call handling should be mentioned in >>> the domain2 context. >>> >>> The problem is; I am not able to send the calls to respective contexts >>> according to their domains when they come from Opensips. I've read the >>> examples on multi-domain setup and have tried taking some help from that >>> example, but whenever the call comes from Opensips to FS, it tries to go >>> into the context that is defined in the SIP Profile. If i don't mention >>> anything in the SIP Profile, it tries to search for default context. >>> I have tried the following: >>> 1) Created file domain1.xml and domain2.xml in the directory folder. >>> 2) mentioned parameters in domain1.xml as follows: >>> <include> >>> <domain name = "domain1.com" >>> <params> >>> <param name="user_context" value "domain1.com"/> >>> </params> >>> </domain> >>> </include> >>> 3) Similarly done for file domain2.xml. >>> >>> But I am just not able to get the calls to the required context according >>> to the domain value in the r-uri. In asterisk something like this in >>> sip.conf worked fine for me: >>> domain=domain1.com, domain1.com >>> domain=domain2.com, domain2.com >>> Can someone please help me understanding where I am going wrong or have >>> I mis-understood something? >>> >>> Thanks in advance !! >>> >>> --- Jayesh >>> >>

2 1

Error in 'svn up'
by Larry Marshall 03 Apr '10

03 Apr '10

The current version of FS I'm using is 17135. I tried to 'make current' and it errored out in the svn up portion: svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a different repository Should I just delete the libs/openzap directory? openzap is commented out in modules.conf.xml. Thanks, Lars

5 8

Conference Dial In Isn't Working (mod_conference)
by Jan Riedinger 03 Apr '10

03 Apr '10

If I dial in to a conference from my Cisco IP Phone 7940 all is working as it should. However, if I dial in via a voip carrier, the calls are disconnect with disconnect cause "facility rejected" by the voip carrier. It seems that Freeswitch tries to change some parameters of the call setup, which is refused. You find below the console log output. Do you have an idea, what I have to change to get it working? I know that it is working in prinicple, because I can setup outgoing conferences via the voip carrier. Thank you very much in advance Jan =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.04.03 16:56:10 =~=~=~=~=~=~=~=~=~=~=~= [36m2010-04-03 16:56:07.593334 [NOTICE] switch_channel.c:669 New Channel sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [27446b50-dc2a-4088-83f2-7b7cc8f014e8] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4153 Channel sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 entering state [received][100] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4164 Remote SDP: v=0 o=- 1270306567 1270306567 IN IP4 XXX.XX.XX.X6 s=- c=IN IP4 XXX.XX.XX.X6 t=0 0 m=audio 19188 RTP/AVP 18 4 8 0 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G729:18:8000:20]/[G7221:115:32000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G729:18:8000:20]/[G7221:107:16000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G729:18:8000:20]/[G722:9:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G729:18:8000:20]/[PCMU:0:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G729:18:8000:20]/[PCMA:8:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G729:18:8000:20]/[GSM:3:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G723:4:8000:20]/[G7221:115:32000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G723:4:8000:20]/[G7221:107:16000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G723:4:8000:20]/[G722:9:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G723:4:8000:20]/[PCMU:0:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G723:4:8000:20]/[PCMA:8:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [G723:4:8000:20]/[GSM:3:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMA:8:8000:20]/[G7221:115:32000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMA:8:8000:20]/[G7221:107:16000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMA:8:8000:20]/[PCMU:0:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:2354 Set Codec sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv payload to 101 [m[33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4310 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Running State Change CS_INIT [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State INIT [m[33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:83 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 SOFIA INIT [m[33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:117 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State Change CS_INIT -> CS_ROUTING [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State INIT going to sleep [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Running State Change CS_ROUTING [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State ROUTING [m[33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:140 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 SOFIA ROUTING [m[33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_state_machine.c:77 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Standard ROUTING [m[32m2010-04-03 16:56:07.593334 [INFO] mod_dialplan_xml.c:418 Processing 49XXXXXXX6->49331YYYYYY in context public [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 parsing [public->unloop] continue=false [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 parsing [public->outside_call] continue=true [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Absolute Condition [outside_call] [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Action set(outside_call=true) [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 parsing [public->call_debug] continue=true [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never ... [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 parsing [public->sns_conference] continue=false [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Regex (PASS) [sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/ break=on-false [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Action answer() [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Action set(conference_enforce_security=false) [m[33mDialplan: sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Action conference(49331YYYYYY-${domain_name}@default) [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_state_machine.c:119 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State Change CS_ROUTING -> CS_EXECUTE [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State ROUTING going to sleep [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Running State Change CS_EXECUTE [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State EXECUTE [m[33m2010-04-03 16:56:07.596335 [DEBUG] mod_sofia.c:226 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 SOFIA EXECUTE [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_state_machine.c:157 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Standard EXECUTE [m[33mEXECUTE sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 set(outside_call=true) [m[33m2010-04-03 16:56:07.596335 [DEBUG] mod_dptools.c:816 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 SET [outside_call]=[true] [m[33mEXECUTE sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 answer() [m[33m2010-04-03 16:56:07.596335 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6] XXX.XX.XX.X7 port 17138 -> XXX.XX.XX.X6 port 19188 codec: 8 ms: 20 [m[33m2010-04-03 16:56:07.596335 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms [m[33m2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send payload to 101 [m[33m2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf receive payload to 101 [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_sofia.c:636 Local SDP sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6: v=0 o=FreeSWITCH 1270289429 1270289430 IN IP4 XXX.XX.XX.X7 s=FreeSWITCH c=IN IP4 XXX.XX.XX.X7 t=0 0 m=audio 17138 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [m[33m2010-04-03 16:56:07.599335 [DEBUG] sofia.c:4153 Channel sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 entering state [completed][200] [m[33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[36m2010-04-03 16:56:07.599335 [NOTICE] mod_dptools.c:719 Channel [sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6] has been answered [m[33mEXECUTE sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 set(conference_enforce_security=false) [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_dptools.c:816 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 SET [conference_enforce_security]=[false] [m[33mEXECUTE sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 conference(49331YYYYYY-XXX.XX.XX.X7@default) [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'mute' bound to '0'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1616 max len 1 [m[33m2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1620 min len 1 [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'deaf mute' bound to '*'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'energy up' bound to '9'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'energy equ' bound to '8'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'energy dn' bound to '7'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'vol talk up' bound to '3'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'vol talk zero' bound to '2'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'vol talk dn' bound to '1'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'vol listen up' bound to '6'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'vol listen zero' bound to '5'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'vol listen dn' bound to '4'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 Installing default caller control action 'hangup' bound to '#'. [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:4990 Raw Codec Activation Success L16@8000hz 1 channel 20ms [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5035 Raw Codec Activation Success L16@8000hz 1 channel 20ms [m[33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_codec.c:122 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Push codec L16:10 [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:994 Setup timer success interval: 20 samples: 160 [m[33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:2202 Setup timer soft success interval: 20 samples: 160 [m[33m2010-04-03 16:56:07.617340 [DEBUG] sofia.c:4153 Channel sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 entering state [ready][200] [m[36m2010-04-03 16:56:07.839332 [NOTICE] sofia.c:481 Hangup sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED] [m[33m2010-04-03 16:56:07.839332 [DEBUG] switch_channel.c:2071 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [KILL] [m[33m2010-04-03 16:56:07.839332 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.848341 [DEBUG] mod_conference.c:2473 Channel leaving conference, cause: FACILITY_REJECTED [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_codec.c:146 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Restore previous codec PCMA:8. [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State EXECUTE going to sleep [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Running State Change CS_HANGUP [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State HANGUP [m[33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:408 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Overriding SIP cause 501 with 200 from the other leg [m[33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:414 Channel sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 hanging up, cause: FACILITY_REJECTED [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:46 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Standard HANGUP, cause: FACILITY_REJECTED [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State HANGUP going to sleep [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:333 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State Change CS_HANGUP -> CS_REPORTING [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Running State Change CS_REPORTING [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State REPORTING [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:53 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Standard REPORTING, cause: FACILITY_REJECTED [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State REPORTING going to sleep [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:327 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State Change CS_REPORTING -> CS_DESTROY [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018 Send signal sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [BREAK] [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1161 Session 7 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Locked, Waiting on external entities [m[36m2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1179 Session 7 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Ended [m[36m2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 [CS_DESTROY] [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:428 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) Running State Change CS_DESTROY [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State DESTROY [m[33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:341 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 SOFIA DESTROY [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:60 sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6 Standard DESTROY [m[33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6(a)XXX.XX.XX.X6) State DESTROY going to sleep [m[33m2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1361 Write Lock ON [m[33m2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1364 Write Lock OFF -- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger(a)sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan(a)sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger AG Charlottenburg - HRB 71973

4 5

Re: [Freeswitch-users] Some question about mod_fifo ??
by Nguyễn Mạnh Hùng 03 Apr '10

03 Apr '10

Hi Seven Du. Thanks to yours suggetion. I have an ideal, it is: when the call between caller and agent is set, the caller_id is determined. So, i want to edit code to sent the agent information (the call_id and call_id_number) which will be displayed againt in the agent's softphone (as Xlite..) when the call is happening. I read some documents but i still can't determine: It's maybe yes or maybe to do this and where to do this. Can you give me some comments. Best Regard. Seven Du [dujinfang(a)gmail.com] ��As discussed in the list, it's not a freeswitch problem but a reality of life. Think about customer A and B calls in one after another, then if FreeSWITCH call agent X with caller id A and Y with caller id B, and angent Y answers before X, then 1) if bridge Y with A with the FIFO rule, then the caller id is wrong 2) if bridge Y with B, the caller id is right but it breaks the rule of FIFO - A should be served before B!! And what even worse is that if X never answer A then A never can be served which is really unfair!! Of course you don't want 1), and you don't need mod_fifo if you want behavior 2), you just need some dialplan trick or some simple Lua script I think. Also FreeSWITCH is designed to be easily extended with almost any languages so feel free to implement anything. 2010/3/31 Nguy�n Mạnh Hùng : > Hi Mike and Seven Du. > Thanks to yours help. > I known the mechanism of mod_fifo. >>>http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175… ml. > What a pity, It can't solve this problem. I can't use freeswitch for my call > center. > Hope new version can solve this !!! _______________________________________________ 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

2 1

domain-wise context
by Jayesh Nambiar 03 Apr '10

03 Apr '10

Hi All, I am quite very new to freeswitch and I am kind of playing with it to understand it better. I am primarily using Opensips as registrar and SIP Proxy and intend to use FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My Opensips is a multi-domain setup and I wish to have all the configuration of media-capabilties segregated domain-wise in the FS too. For eg: When a call for user(a)domain1.com needs to go to voicemail, I redirect that call to FS IP address keeping the URI intact. I add the mailbox number as a header as X-Mailbox and have FS extract it and go to appropriate mailbox. Similarly when a call for user(a)domain2.com needs to go to voicemail I do the same thing. The requirement is I want to maintain the dialplans for each domains separately. Thus if call from Opensips comes to FS with domain as domain1, the call should go to dialplan context domain1 and similarly if call from Opensips comes to FS with domain2 the call handling should be mentioned in the domain2 context. The problem is; I am not able to send the calls to respective contexts according to their domains when they come from Opensips. I've read the examples on multi-domain setup and have tried taking some help from that example, but whenever the call comes from Opensips to FS, it tries to go into the context that is defined in the SIP Profile. If i don't mention anything in the SIP Profile, it tries to search for default context. I have tried the following: 1) Created file domain1.xml and domain2.xml in the directory folder. 2) mentioned parameters in domain1.xml as follows: <include> <domain name = "domain1.com" <params> <param name="user_context" value "domain1.com"/> </params> </domain> </include> 3) Similarly done for file domain2.xml. But I am just not able to get the calls to the required context according to the domain value in the r-uri. In asterisk something like this in sip.conf worked fine for me: domain=domain1.com, domain1.com domain=domain2.com, domain2.com Can someone please help me understanding where I am going wrong or have I mis-understood something? Thanks in advance !! --- Jayesh

3 4

CLI status?
by Lon Baker 03 Apr '10

03 Apr '10

Hi gang! I'm trying to determine the best way to periodically check the status of production Freeswitch deployments. While Freeswitch has been incredibly stable in production, I would like to ping it periodically to check that its not hung in someway. Has anyone considered or done this yet? My thought it so issue a command via fs_cli periodically and compare the out. But am trying to determine what a good command would be. thinking of using show channels or show calls, in addition to status. Any thoughts? Is this a wrong-headed approach? Thanks for any feedback. Lon

6 8

make fails
by Troy Anderson 02 Apr '10

02 Apr '10

Help. I've compiled FS successfully on this box many times, but now, it's failing. It must be something I've done as it even fails when I check out an svn revision that compiled in the past on this box. I'm hopeful someone here can point me in the right direction. Here's the output: make make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive Making all in . gcc -I/home/chronos_client/freeswitch.svn/src/include -I/home/chronos_client/freeswitch.svn/src/include -I/home/chronos_client/freeswitch.svn/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/home/chronos_client/freeswitch.svn/libs/apr/include -I/home/chronos_client/freeswitch.svn/libs/apr-util/include -I/home/chronos_client/freeswitch.svn/libs/apr-util/xml/expat/lib -I/home/chronos_client/freeswitch.svn/libs/stfu -I/home/chronos_client/freeswitch.svn/libs/sqlite -I/home/chronos_client/freeswitch.svn/libs/pcre -I/home/chronos_client/freeswitch.svn/libs/speex/include -Ilibs/speex/include -I/home/chronos_client/freeswitch.svn/libs/srtp/include -I/home/chronos_client/freeswitch.svn/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -DSWITCH_HAVE_ODBC -I/usr/include -I/home/chronos_client/freeswitch.svn/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch freeswitch-switch.o -pthread -lm ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a libs/libedit/src/.libs/libedit.a -L/home/chronos_client/freeswitch.svn/libs/apr-util/xml/expat/lib -lrt -ldl -lcrypt -lpthread -lssl -lcrypto -lncurses -Wl,--rpath -Wl,/usr/local/freeswitch/lib ./.libs/libfreeswitch.so: undefined reference to `XML_ErrorString' ./.libs/libfreeswitch.so: undefined reference to `XML_SetUserData' ./.libs/libfreeswitch.so: undefined reference to `XML_ParserFree' ./.libs/libfreeswitch.so: undefined reference to `XML_SetElementHandler' ./.libs/libfreeswitch.so: undefined reference to `XML_SetCharacterDataHandler' ./.libs/libfreeswitch.so: undefined reference to `XML_GetErrorCode' ./.libs/libfreeswitch.so: undefined reference to `XML_ParserCreate' ./.libs/libfreeswitch.so: undefined reference to `XML_Parse' collect2: ld returned 1 exit status make[2]: *** [freeswitch] Error 1 Thanks! Troy

1 0

Calling in context public instead of default, mod_xml_curl
by Vitalii Colosov 02 Apr '10

02 Apr '10

Hi, I am using mod_xml_curl to generate user directory data. It produces data like this: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="$${domain}"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <groups> <group name="default"> <users> <user id="20001"> <params> <param name="a1-hash" value="d8c32787fd9f00b5ad4246e93f757360"/> <param name="vm-a1-hash" value="a53b5e8b5edf34208274cbaa00412097"/> </params> <variables> *<variable name="user_context" value="**default**"/>* <variable name="accountcode" value="20001"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> Both internal and external sofia profiles have context = "public". But it should be overridden here in the user settings to DEFAULT. Wwhen I call to any number using this subscriber 20001 it calls in *context PUBLIC instead of DEFAULT.* 2010-04-02 17:27:32.384472 [INFO] mod_dialplan_xml.c:418 Processing 20001->001...[removed]... in context *public* I need it to call in context default. When I change sofia profile settings to context="default", it works fine, but shouldn't it work like this as well? Please any hints where else should I change the context? Thank you, vIT

1 0