July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

opensips + freeswitch
by Madovsky 01 Jul '10

01 Jul '10

Is there any wiki or link to use Opensips as proxy with FS cluster ? Thanks F

2 4

How to config acl from web GUI ?
by Dome Charoenyost 01 Jul '10

01 Jul '10

I want to make wholesale service. so now i use acl to allow ip. so i want to use web gui. what's best way to do that ? 1. use xml_curl binding configuration to web application. i'm worry about peformance because FS request to web apps every call setup. 2. user web apps direct manage acl.conf.xml and use webapi reload acl Best Reguards. Dome C.

3 3

mod_freetdm is not loading - Saying error
by lakshmanan ganapathy 01 Jul '10

01 Jul '10

Hi all, I've downloaded and compiled wanpipe and freeswitch as it was given in http://wiki.sangoma.com/wanpipe-api-freetdm-linux Here is my freetdm.conf: [span wanpipe wp1] number =>1 trunk_type =>e1 group=>grp1 b-channel => 1:1-15 b-channel => 1:17-31 I started freeswitch In the CLI, I gave "load mod_freetdm" and it gives me the following error. http://pastebin.freeswitch.org/13327 It says "Opening freetdm.conf failed". Can someone tell me how to solve this one.

2 1

groups and g729 codec
by Mark Campbell-Smith 01 Jul '10

01 Jul '10

Hi! I have a group defined to call three extensions. The inbound route uses G729 and I only have one license. When a call comes into FS with a g729 codec chosen, the group call fails with encoding errors. Only one extension in the group supports the g729 codec. 2010-07-01 20:32:17.685087 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0xb6738248 (nil) 2010-07-01 20:32:17.686085 [ERR] switch_core_io.c:352 Codec G.729 decoder error! Is this because FS is trying to set up 3 channels on the b-leg with g729? I thought by activating late-negotiation this might work. No luck. I thought by creating a specific dialplan in the users setting it might work. I couldn't find the correct format. Would this work? <param name="dial-string" value="{absolute_codec_string=GSM}{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> Is there a way I can get this to work, or do I need to force the inbound call to use another codec? Thanks for your help

2 1

Re: [Freeswitch-users] create a external FXO Trunk - A200
by Deya M 01 Jul '10

01 Jul '10

Hi, Managed to ge I added the following to the dialplan: <extension name="OutgoingNumber"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bridge" data="openzap/1/4/${dialed_ext}"/> </condition> </extension> No ring. I get the following: 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1317 Connect outbound channel OpenZAP/1:4/ 2010-07-01 11:07:04.627394 [NOTICE] switch_channel.c:776 New Channel OpenZAP/1:4/ [3ca74e5b-7e26-4a0a-b5f7-72d7a2f33fbf] 2010-07-01 11:07:04.627394 [DEBUG] mod_openzap.c:1331 (OpenZAP/1:4/) State Change CS_NEW -> CS_INIT 2010-07-01 11:07:04.627394 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:59 Changing state on 1:4 from DOWN to DIALING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_analog.c:293 ANALOG CHANNEL thread starting. 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_INIT 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:431 (OpenZAP/1:4/) State Change CS_INIT -> CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:4/) State INIT going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_channel.c:1471 (OpenZAP/1:4/) Callstate Change DOWN -> RINGING 2010-07-01 11:07:04.627394 [DEBUG] ozmod_wanpipe.c:608 Enabled DTMF events on chan 1:4 2010-07-01 11:07:04.628392 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DIALING 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING 2010-07-01 11:07:04.628392 [DEBUG] mod_openzap.c:454 OpenZAP/1:4/ CHANNEL ROUTING 2010-07-01 11:07:04.628392 [DEBUG] switch_ivr_originate.c:64 (OpenZAP/1:4/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:4/) State ROUTING going to sleep 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA 2010-07-01 11:07:04.628392 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:4/) State CONSUME_MEDIA going to sleep 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:20.335200 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 2 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1036 1:4 wanpipe returned event 5 2010-07-01 11:07:21.150195 [DEBUG] ozmod_wanpipe.c:1061 1:4 rxhook, state 1 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:802 EVENT [ONHOOK][1:4] STATE [DIALING] 2010-07-01 11:07:21.150195 [DEBUG] ozmod_analog.c:838 Changing state on 1:4 from DIALING to DOWN 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:464 Executing state handler on 1:4 for DOWN 2010-07-01 11:07:21.166199 [DEBUG] mod_openzap.c:1556 got FXO sig 1:4 [STOP] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2261 (OpenZAP/1:4/) Callstate Change RINGING -> HANGUP 2010-07-01 11:07:21.166199 [NOTICE] mod_openzap.c:1577 Hangup OpenZAP/1:4/ [CS_CONSUME_MEDIA] [NONE] 2010-07-01 11:07:21.166199 [DEBUG] switch_channel.c:2277 Send signal OpenZAP/1:4/ [KILL] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.166199 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_HANGUP 2010-07-01 11:07:21.166199 [DEBUG] zap_io.c:1388 channel done 1:4 2010-07-01 11:07:21.166199 [DEBUG] ozmod_analog.c:778 ANALOG CHANNEL 1:4 thread ended. 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP 2010-07-01 11:07:21.168191 [WARNING] mod_openzap.c:517 VETO Changing state on 1:4 from DOWN to HANGUP 2010-07-01 11:07:21.168191 [DEBUG] mod_openzap.c:556 OpenZAP/1:4/ CHANNEL HANGUP 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:4/ Standard HANGUP, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:500 (OpenZAP/1:4/) State HANGUP going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_ivr_originate.c:3369 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-07-01 11:07:21.168191 [INFO] mod_dptools.c:2382 Originate Failed. Cause: NO_ANSWER 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:4/) State Change CS_HANGUP -> CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:4/) Running State Change CS_REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:4/ Standard REPORTING, cause: NONE 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:591 (OpenZAP/1:4/) State REPORTING going to sleep 2010-07-01 11:07:21.168191 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:4/) State Change CS_REPORTING -> CS_DESTROY 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1027 Send signal OpenZAP/1:4/ [BREAK] 2010-07-01 11:07:21.168191 [DEBUG] switch_core_session.c:1175 Session 71 (OpenZAP/1:4/) Locked, Waiting on external entities 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1193 Session 71 (OpenZAP/1:4/) Ended 2010-07-01 11:07:21.168191 [NOTICE] switch_core_session.c:1195 Close Channel OpenZAP/1:4/ [CS_DESTROY] But no ringing is heard. thanks, -:D On Thu, Jul 1, 2010 at 2:49 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Default Dial Plan: action application bridge (Deya M) > > > ---------- Forwarded message ---------- > From: Deya M <deya787(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 1 Jul 2010 14:49:26 +1000 > Subject: Re: [Freeswitch-users] Default Dial Plan: action application > bridge > I generated samples, and started from scratch. > > I also made a copy of the conf after installation, rm it, and copied the > original conf before testing! > > -:D > > > On Thu, Jul 1, 2010 at 1:37 PM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Query !! freeswitch, dingaling, jingle, skypopen >> (Anthony Minessale) >> 2. continue_on_fail and hangup_after_bridge with transfer >> (Vladimir Elizarov) >> 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) >> 4. Default Dial Plan: action application bridge (Deya M) >> 5. Re: Default Dial Plan: action application bridge >> (Anthony Minessale) >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale <anthony.minessale(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Date: Wed, 30 Jun 2010 17:40:27 -0500 >> Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, >> skypopen >> If he didn't like to joke, he'd never survive in telephony. >> >> >> On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan <sameer2k3t(a)gmail.com>wrote: >> >>> no problem. but did't expect from the creator of mod_skypopen, the wonder >>> i would say >>> >>> >>> On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli < >>> gmaruzz(a)celliax.org> wrote: >>> >>>> On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan <sameer2k3t(a)gmail.com> >>>> wrote: >>>> > Mr Giovani what was that ? >>>> >>>> just jokin, nevermind >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> _______________________________________________ >>>> 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 >>>> >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> >> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> >> pstn:+19193869900 >> >> >> ---------- Forwarded message ---------- >> From: Vladimir Elizarov <xengelpublicx(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Date: Thu, 1 Jul 2010 03:29:09 +0400 >> Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with >> transfer >> I'm trying to make a dialplan: >> >> <extension name="Long Dial"> >> <condition field="destination_number" >> >> expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> >> <action application="set" >> data="hangup_after_bridge=true"/> >> <action >> application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> >> <action application="set" data="real_dialed_number=7777777"/> >> </condition> >> >> <condition field="destination_number" expression="^(79|89)(\d{9})$" >> break="on-true"> >> <action application="set" >> data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> >> <action application="bridge" >> >> data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> >> <action application="transfer" data="to_card"/> >> </condition> >> </extension> >> >> <extension name="to_card"> >> <condition field="destination_number" expression="^to_card$"> >> <action application="bridge" >> data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${ >> real_dialed_number}@192.168.50.53:5061<http://real_dialed_number%7D%40192.168.50.53%3A5061> >> "/> >> </condition> >> </extension> >> >> The logic of his work: if unavailable gateway to the next. if not >> available all the gateway to go to the extension transfer. >> A problem in If we get a code busy here, it is not satisfied >> hangup_after_bridge. The call goes to the transfer. Why is this so? >> >> trace: >> http://pastebin.freeswitch.org/13244 >> >> -- >> Best regards, Vladimir Elizarov >> >> >> >> >> ---------- Forwarded message ---------- >> From: "Tim St. Pierre" <fs-list(a)communicatefreely.net> >> To: freeswitch-users(a)lists.freeswitch.org >> Date: Wed, 30 Jun 2010 22:45:19 -0400 >> Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge >> Thanks all, >> >> I read that bit, but didn't fully understand it's implications. >> >> I'll give that a go. >> >> -Tim >> >> Steven Ayre wrote: >> > <action application="bridge" >> > >> data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/> >> > >> > >> > >> > On 28 June 2010 19:45, Tim St. Pierre <fs-list(a)communicatefreely.net >> > <mailto:fs-list@communicatefreely.net>> wrote: >> > >> > Hello list, >> > >> > I would like to bridge a call to multiple SIP endpoints, but add >> > different headers to each. >> > >> > I'm not entirely sure how to do this. I have no problem exporting a >> > SIP header that does what I >> > want for one destination, but I'm not sure how to set it for two. >> > >> > My application is that I want two IP phones to ring - one with the >> > internal ring-ring splash, the >> > others with a group-answer (single ring, then lamp flash only), for >> > administrative assistants, etc. >> > >> > How do I export different variables to each branch? >> > >> > Thanks! >> > >> > -Tim >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org >> > <mailto:FreeSWITCH-users@lists.freeswitch.org> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > 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 >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Deya M <deya787(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Date: Thu, 1 Jul 2010 06:19:51 +0300 >> Subject: [Freeswitch-users] Default Dial Plan: action application bridge >> Hi, >> >> In the default dial plan, with two extensions defined, 1000 and 1001, when >> I call from 1000 to 1001, I always get Voicemail, using the default config >> files: conf/dialplan/default.xml >> >> I changed the following from : >> >> * <action application="bridge" data="user/${dialed_extension}@ >> ${domain_name}"/>* >> TO >> *<action application="bridge" >> data="sofia/internal/${dialed_extension}%${domain_name}"/>* >> >> and it did work! Not sure if the first / default one, * <action >> application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should >> work, but something is missing ? >> >> >> New: >> <extension name="Local_Extension"> >> <condition field="destination_number" expression="^(10[01][0-9])$"> >> <action application="set" data="dialed_extension=$1"/> >> <action application="export" data="dialed_extension=$1"/> >>  >> <action application="bind_meta_app" data="1 b s >> execute_extension::dx XML features"/> >> <action application="bind_meta_app" data="2 b s >> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >> <action application="bind_meta_app" data="3 b s >> execute_extension::cf XML features"/> >> <action application="set" data="ringback=${us-ring}"/> >> <action application="set" >> data="transfer_ringback=$${hold_music}"/> >> <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="hash" >> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> <action application="hash" >> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> <action application="set" >> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >>  >> <action application="hash" >> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> <action application="bridge" >> data="sofia/internal/${dialed_extension}%${domain_name}"/> >> <action application="answer"/> >> <action application="sleep" data="1000"/> >> <action application="bridge" data="loopback/app=voicemail:default >> ${domain_name} ${dialed_extension}"/> >> </condition> >> </extension> >> >> >> Old: >> >> <extension name="Local_Extension"> >> <condition field="destination_number" expression="^(10[01][0-9])$"> >> <action application="set" data="dialed_extension=$1"/> >> <action application="export" data="dialed_extension=$1"/> >>  >> <action application="bind_meta_app" data="1 b s >> execute_extension::dx XML features"/> >> <action application="bind_meta_app" data="2 b s >> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >> <action application="bind_meta_app" data="3 b s >> execute_extension::cf XML features"/> >> <action application="set" data="ringback=${us-ring}"/> >> <action application="set" >> data="transfer_ringback=$${hold_music}"/> >> <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="hash" >> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> <action application="hash" >> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> <action application="set" >> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >>  >> <action application="hash" >> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> <action application="bridge" data="user/${dialed_extension}@ >> ${domain_name}"/> >> <action application="answer"/> >> <action application="sleep" data="1000"/> >> <action application="bridge" data="loopback/app=voicemail:default >> ${domain_name} ${dialed_extension}"/> >> </condition> >> </extension> >> >> :-D >> >> >> >> ---------- Forwarded message ---------- >> From: Anthony Minessale <anthony.minessale(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Date: Wed, 30 Jun 2010 22:37:15 -0500 >> Subject: Re: [Freeswitch-users] Default Dial Plan: action application >> bridge >> you must have changed more than you think. >> >> you might want to diff you configs against the in-tree ones. >> >> >> On Wed, Jun 30, 2010 at 10:19 PM, Deya M <deya787(a)gmail.com> wrote: >> >>> Hi, >>> >>> In the default dial plan, with two extensions defined, 1000 and 1001, >>> when I call from 1000 to 1001, I always get Voicemail, using the default >>> config files: conf/dialplan/default.xml >>> >>> I changed the following from : >>> >>> * <action application="bridge" data="user/${dialed_extension}@ >>> ${domain_name}"/>* >>> TO >>> *<action application="bridge" >>> data="sofia/internal/${dialed_extension}%${domain_name}"/>* >>> >>> and it did work! Not sure if the first / default one, * <action >>> application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should >>> work, but something is missing ? >>> >>> >>> New: >>> <extension name="Local_Extension"> >>> <condition field="destination_number" expression="^(10[01][0-9])$"> >>> <action application="set" data="dialed_extension=$1"/> >>> <action application="export" data="dialed_extension=$1"/> >>>  >>> <action application="bind_meta_app" data="1 b s >>> execute_extension::dx XML features"/> >>> <action application="bind_meta_app" data="2 b s >>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >>> <action application="bind_meta_app" data="3 b s >>> execute_extension::cf XML features"/> >>> <action application="set" data="ringback=${us-ring}"/> >>> <action application="set" >>> data="transfer_ringback=$${hold_music}"/> >>> <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="hash" >>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >>> <action application="hash" >>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >>> <action application="set" >>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >>> var callgroup)}"/> >>>  >>> <action application="hash" >>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >>> <action application="bridge" >>> data="sofia/internal/${dialed_extension}%${domain_name}"/> >>> <action application="answer"/> >>> <action application="sleep" data="1000"/> >>> <action application="bridge" data="loopback/app=voicemail:default >>> ${domain_name} ${dialed_extension}"/> >>> </condition> >>> </extension> >>> >>> >>> Old: >>> >>> <extension name="Local_Extension"> >>> <condition field="destination_number" expression="^(10[01][0-9])$"> >>> <action application="set" data="dialed_extension=$1"/> >>> <action application="export" data="dialed_extension=$1"/> >>>  >>> <action application="bind_meta_app" data="1 b s >>> execute_extension::dx XML features"/> >>> <action application="bind_meta_app" data="2 b s >>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >>> <action application="bind_meta_app" data="3 b s >>> execute_extension::cf XML features"/> >>> <action application="set" data="ringback=${us-ring}"/> >>> <action application="set" >>> data="transfer_ringback=$${hold_music}"/> >>> <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="hash" >>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >>> <action application="hash" >>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >>> <action application="set" >>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >>> var callgroup)}"/> >>>  >>> <action application="hash" >>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >>> <action application="bridge" data="user/${dialed_extension}@ >>> ${domain_name}"/> >>> <action application="answer"/> >>> <action application="sleep" data="1000"/> >>> <action application="bridge" data="loopback/app=voicemail:default >>> ${domain_name} ${dialed_extension}"/> >>> </condition> >>> </extension> >>> >>> :-D >>> >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> >> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> >> pstn:+19193869900 >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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

ESL: Question about BACKGROUND_JOB events
by Helmut Kuper 01 Jul '10

01 Jul '10

Hello, today I found that filtering events for a Job-UUID returned by a bgapi command doesn't work well. What I do is this: $con->events("plain", "CHANNEL_HANGUP"); $con->events("plain", "CHANNEL_ANSWER"); $con->events("plain", "BACKGROUND_JOB"); $con->filter("Unique-ID", $a_uuid); $con->filter("Caller-Unique-ID", $a_uuid); $jobid=$con->bgapi("blahblah", ...)->getHeader("Job-UUID"); $con->filter("Job-UUID", $jobid); Then waiting for events. But no BACKGROUND_JOB event is received. Perhaps the BACKGROUND_JOB event is fired before the filter can be applied. When I execute "$con->filter("Event-Name", "BACKGROUND_JOB");" before "$jobid=..." then I'm able to receive it ... and all other but unwanted BACKGROUND_JOB events as well. Any trick to avoid this? regards Helmut

5 9

mod_erlang_event problem
by Timur Irmatov 01 Jul '10

01 Jul '10

Hi! I am trying to use mod_erlang_event with freeswitch 1.0.6 on Red Hat Enterprise Linux Server release 5.4, 64-bit server. Erlang version is R13B04. Calls are sent to my erlang application via: <extension name="phonebooth"> <condition field="destination_number" expression="^000([0-9]+)$"> <action application="erlang" data="phonebooth:launch test@test-server" /> </condition> </extension> My application is very simple: it just prints all events received from freeswitch. The problem is, that call is being terminated immidiately. As far as I can see, phonebooth:launch is called successfully, it returns a pid of a new process. This new process is still alive after the call is finished, and it does not receive any events from freeswitch (if it would, it would print them to screen). Freeswitch log tells me that erlang_outbound_function exits as soon as it gets new pid: 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1316 got pid! 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1446 exit erlang_outbound_function What is wrong? As this is not my first try with erlang and freeswitch (previous were successful) I think there's nothing wrong with configuration, but may be the problem is the box itself, old Red Hat? Here's full log from freeswitch: 2010-06-24 17:46:58.265274 [NOTICE] switch_channel.c:669 New Channel sofia/external/1220139(a)10.1.1.1 [93975342-7f8e-11df-bae2-7b19ef829fd0] 2010-06-24 17:46:58.265274 [DEBUG] sofia.c:4153 Channel sofia/external/1220139(a)10.1.1.1 entering state [received][100] 2010-06-24 17:46:58.265274 [DEBUG] sofia.c:4164 Remote SDP: v=0 o=HuaweiSoftX3000 282382 282382 IN IP4 10.1.1.1 s=Sip Call c=IN IP4 172.16.12.1 t=0 0 m=audio 50000 RTP/AVP 8 0 18 4 2 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 2010-06-24 17:46:58.265274 [DEBUG] sofia.c:4273 (sofia/external/1220139(a)10.1.1.1) State Change CS_NEW -> CS_INIT 2010-06-24 17:46:58.265274 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139(a)10.1.1.1 [BREAK] 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139(a)10.1.1.1) Running State Change CS_INIT 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1220139(a)10.1.1.1) State INIT 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:83 sofia/external/1220139(a)10.1.1.1 SOFIA INIT 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:117 (sofia/external/1220139(a)10.1.1.1) State Change CS_INIT -> CS_ROUTING 2010-06-24 17:46:58.265274 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139(a)10.1.1.1 [BREAK] 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:338 (sofia/external/1220139(a)10.1.1.1) State INIT going to sleep 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139(a)10.1.1.1) Running State Change CS_ROUTING 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1220139(a)10.1.1.1) State ROUTING 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:140 sofia/external/1220139(a)10.1.1.1 SOFIA ROUTING 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:77 sofia/external/1220139(a)10.1.1.1 Standard ROUTING 2010-06-24 17:46:58.265274 [INFO] mod_dialplan_xml.c:418 Processing 1220139->0003707057 in context public Dialplan: sofia/external/1220139(a)10.1.1.1 parsing [public->unloop] continue=false Dialplan: sofia/external/1220139(a)10.1.1.1 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1220139(a)10.1.1.1 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1220139(a)10.1.1.1 parsing [public->outside_call] continue=true Dialplan: sofia/external/1220139(a)10.1.1.1 Absolute Condition [outside_call] Dialplan: sofia/external/1220139(a)10.1.1.1 Action set(outside_call=true) Dialplan: sofia/external/1220139(a)10.1.1.1 parsing [public->call_debug] continue=true Dialplan: sofia/external/1220139(a)10.1.1.1 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/1220139(a)10.1.1.1 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1220139(a)10.1.1.1 Regex (FAIL) [public_extensions] destination_number(0003707057) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/1220139(a)10.1.1.1 parsing [public->public_did] continue=false Dialplan: sofia/external/1220139(a)10.1.1.1 Regex (FAIL) [public_did] destination_number(0003707057) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/1220139(a)10.1.1.1 parsing [public->phonebooth] continue=false Dialplan: sofia/external/1220139(a)10.1.1.1 Regex (PASS) [phonebooth] destination_number(0003707057) =~ /^000([0-9]+)$/ break=on-false Dialplan: sofia/external/1220139(a)10.1.1.1 Action erlang(phonebooth:launch test@test-server) 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1220139(a)10.1.1.1) State Change CS_ROUTING -> CS_EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139(a)10.1.1.1 [BREAK] 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:341 (sofia/external/1220139(a)10.1.1.1) State ROUTING going to sleep 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139(a)10.1.1.1) Running State Change CS_EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:348 (sofia/external/1220139(a)10.1.1.1) State EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] mod_sofia.c:226 sofia/external/1220139(a)10.1.1.1 SOFIA EXECUTE 2010-06-24 17:46:58.265274 [DEBUG] switch_core_state_machine.c:157 sofia/external/1220139(a)10.1.1.1 Standard EXECUTE EXECUTE sofia/external/1220139(a)10.1.1.1 set(outside_call=true) 2010-06-24 17:46:58.265274 [DEBUG] mod_dptools.c:816 sofia/external/1220139(a)10.1.1.1 SET [outside_call]=[true] EXECUTE sofia/external/1220139(a)10.1.1.1 erlang(phonebooth:launch test@test-server) 2010-06-24 17:46:58.265274 [DEBUG] mod_erlang_event.c:1403 enter erlang_outbound_function phonebooth:launch test@test-server 2010-06-24 17:46:58.265274 [DEBUG] mod_erlang_event.c:1409 Creating new listener for session 2010-06-24 17:46:58.269275 [DEBUG] mod_erlang_event.c:1418 Launching new listener 2010-06-24 17:46:58.269275 [DEBUG] mod_erlang_event.c:1424 Creating new spawned session for listener 2010-06-24 17:46:58.271280 [DEBUG] mod_erlang_event.c:961 Connection Open 2010-06-24 17:46:58.271280 [DEBUG] mod_erlang_event.c:1291 rpc call: phonebooth:launch(Ref) 2010-06-24 17:46:58.271280 [DEBUG] handle_msg.c:776 Hashed ref to 4.0.0@freeswitch@test-server 2010-06-24 17:46:58.271280 [DEBUG] handle_msg.c:787 Found waiting slot for 4.0.0@freeswitch@test-server 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1316 got pid! 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:1446 exit erlang_outbound_function 2010-06-24 17:46:58.281273 [NOTICE] switch_core_state_machine.c:185 sofia/external/1220139(a)10.1.1.1 has executed the last dialplan instruction, hanging up. 2010-06-24 17:46:58.281273 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/external/1220139(a)10.1.1.1 [CS_EXECUTE] [NORMAL_CLEARING] 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_EXECUTE_COMPLETE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_channel.c:2102 Send signal sofia/external/1220139(a)10.1.1.1 [KILL] 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139(a)10.1.1.1 [BREAK] 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_HANGUP to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:348 (sofia/external/1220139(a)10.1.1.1) State EXECUTE going to sleep 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139(a)10.1.1.1) Running State Change CS_HANGUP 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_STATE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1220139(a)10.1.1.1) State HANGUP 2010-06-24 17:46:58.281273 [DEBUG] mod_sofia.c:414 Channel sofia/external/1220139(a)10.1.1.1 hanging up, cause: NORMAL_CLEARING 2010-06-24 17:46:58.281273 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 480 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:46 sofia/external/1220139(a)10.1.1.1 Standard HANGUP, cause: NORMAL_CLEARING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:499 (sofia/external/1220139(a)10.1.1.1) State HANGUP going to sleep 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:333 (sofia/external/1220139(a)10.1.1.1) State Change CS_HANGUP -> CS_REPORTING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139(a)10.1.1.1 [BREAK] 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_HANGUP_COMPLETE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:314 (sofia/external/1220139(a)10.1.1.1) Running State Change CS_REPORTING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1220139(a)10.1.1.1) State REPORTING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:53 sofia/external/1220139(a)10.1.1.1 Standard REPORTING, cause: NORMAL_CLEARING 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:590 (sofia/external/1220139(a)10.1.1.1) State REPORTING going to sleep 2010-06-24 17:46:58.281273 [DEBUG] switch_core_state_machine.c:327 (sofia/external/1220139(a)10.1.1.1) State Change CS_REPORTING -> CS_DESTROY 2010-06-24 17:46:58.281273 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_STATE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1021 Send signal sofia/external/1220139(a)10.1.1.1 [BREAK] 2010-06-24 17:46:58.281273 [DEBUG] switch_core_session.c:1164 Session 4 (sofia/external/1220139(a)10.1.1.1) Locked, Waiting on external entities 2010-06-24 17:46:58.283274 [NOTICE] switch_core_session.c:1182 Session 4 (sofia/external/1220139(a)10.1.1.1) Ended 2010-06-24 17:46:58.283274 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/1220139(a)10.1.1.1 [CS_DESTROY] 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:428 (sofia/external/1220139(a)10.1.1.1) Running State Change CS_DESTROY 2010-06-24 17:46:58.283274 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_DESTROY to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1220139(a)10.1.1.1) State DESTROY 2010-06-24 17:46:58.283274 [DEBUG] mod_sofia.c:341 sofia/external/1220139(a)10.1.1.1 SOFIA DESTROY 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:60 sofia/external/1220139(a)10.1.1.1 Standard DESTROY 2010-06-24 17:46:58.283274 [DEBUG] switch_core_state_machine.c:439 (sofia/external/1220139(a)10.1.1.1) State DESTROY going to sleep 2010-06-24 17:46:58.283274 [DEBUG] mod_erlang_event.c:153 Sending event CHANNEL_STATE to attached session for 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.371269 [WARNING] mod_erlang_event.c:531 Can't locate session 93975342-7f8e-11df-bae2-7b19ef829fd0 2010-06-24 17:46:58.371269 [DEBUG] mod_erlang_event.c:586 Notifying new session failed 2010-06-24 17:46:58.371269 [DEBUG] mod_erlang_event.c:323 Removing session element for 93975342-7f8e-11df-bae2-7b19ef829fd0 -- Timur Irmatov, xmpp:irmatov@jabber.ru

2 7

Re: [Freeswitch-users] Default Dial Plan: action application bridge
by Deya M 01 Jul '10

01 Jul '10

I generated samples, and started from scratch. I also made a copy of the conf after installation, rm it, and copied the original conf before testing! -:D On Thu, Jul 1, 2010 at 1:37 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Query !! freeswitch, dingaling, jingle, skypopen > (Anthony Minessale) > 2. continue_on_fail and hangup_after_bridge with transfer > (Vladimir Elizarov) > 3. Re: SIP header on only one fork of a bridge (Tim St. Pierre) > 4. Default Dial Plan: action application bridge (Deya M) > 5. Re: Default Dial Plan: action application bridge > (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 30 Jun 2010 17:40:27 -0500 > Subject: Re: [Freeswitch-users] Query !! freeswitch, dingaling, jingle, > skypopen > If he didn't like to joke, he'd never survive in telephony. > > > On Wed, Jun 30, 2010 at 3:39 PM, Sameer Khan <sameer2k3t(a)gmail.com> wrote: > >> no problem. but did't expect from the creator of mod_skypopen, the wonder >> i would say >> >> >> On Wed, Jun 30, 2010 at 9:19 PM, Giovanni Maruzzelli <gmaruzz(a)celliax.org >> > wrote: >> >>> On Wed, Jun 30, 2010 at 6:02 PM, Sameer Khan <sameer2k3t(a)gmail.com> >>> wrote: >>> > Mr Giovani what was that ? >>> >>> just jokin, nevermind >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _______________________________________________ >>> 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 >>> >> >> >> _______________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > pstn:+19193869900 > > > ---------- Forwarded message ---------- > From: Vladimir Elizarov <xengelpublicx(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 1 Jul 2010 03:29:09 +0400 > Subject: [Freeswitch-users] continue_on_fail and hangup_after_bridge with > transfer > I'm trying to make a dialplan: > > <extension name="Long Dial"> > <condition field="destination_number" > expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> > <action application="set" > data="hangup_after_bridge=true"/> > <action > application="set"data="continue_on_fail=NO_ROUTE_DESTINATION"/> > <action application="set" data="real_dialed_number=7777777"/> > </condition> > > <condition field="destination_number" expression="^(79|89)(\d{9})$" > break="on-true"> > <action application="set" > data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> > <action application="bridge" > > data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> > <action application="transfer" data="to_card"/> > </condition> > </extension> > > <extension name="to_card"> > <condition field="destination_number" expression="^to_card$"> > <action application="bridge" > data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${ > real_dialed_number}@192.168.50.53:5061"/> > </condition> > </extension> > > The logic of his work: if unavailable gateway to the next. if not > available all the gateway to go to the extension transfer. > A problem in If we get a code busy here, it is not satisfied > hangup_after_bridge. The call goes to the transfer. Why is this so? > > trace: > http://pastebin.freeswitch.org/13244 > > -- > Best regards, Vladimir Elizarov > > > > > ---------- Forwarded message ---------- > From: "Tim St. Pierre" <fs-list(a)communicatefreely.net> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 30 Jun 2010 22:45:19 -0400 > Subject: Re: [Freeswitch-users] SIP header on only one fork of a bridge > Thanks all, > > I read that bit, but didn't fully understand it's implications. > > I'll give that a go. > > -Tim > > Steven Ayre wrote: > > <action application="bridge" > > > data="{this_is_global=true}[this_is_gw1_only=true]sofia/gateway/gw1/$1|[this_is_gw2_only=true]sofia/gateway/gw2/$1"/> > > > > > > > > On 28 June 2010 19:45, Tim St. Pierre <fs-list(a)communicatefreely.net > > <mailto:fs-list@communicatefreely.net>> wrote: > > > > Hello list, > > > > I would like to bridge a call to multiple SIP endpoints, but add > > different headers to each. > > > > I'm not entirely sure how to do this. I have no problem exporting a > > SIP header that does what I > > want for one destination, but I'm not sure how to set it for two. > > > > My application is that I want two IP phones to ring - one with the > > internal ring-ring splash, the > > others with a group-answer (single ring, then lamp flash only), for > > administrative assistants, etc. > > > > How do I export different variables to each branch? > > > > Thanks! > > > > -Tim > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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 > > > > > > ---------- Forwarded message ---------- > From: Deya M <deya787(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 1 Jul 2010 06:19:51 +0300 > Subject: [Freeswitch-users] Default Dial Plan: action application bridge > Hi, > > In the default dial plan, with two extensions defined, 1000 and 1001, when > I call from 1000 to 1001, I always get Voicemail, using the default config > files: conf/dialplan/default.xml > > I changed the following from : > > * <action application="bridge" data="user/${dialed_extension}@ > ${domain_name}"/>* > TO > *<action application="bridge" > data="sofia/internal/${dialed_extension}%${domain_name}"/>* > > and it did work! Not sure if the first / default one, * <action > application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should > work, but something is missing ? > > > New: > <extension name="Local_Extension"> > <condition field="destination_number" expression="^(10[01][0-9])$"> > <action application="set" data="dialed_extension=$1"/> > <action application="export" data="dialed_extension=$1"/> >  > <action application="bind_meta_app" data="1 b s > execute_extension::dx XML features"/> > <action application="bind_meta_app" data="2 b s > record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > <action application="bind_meta_app" data="3 b s > execute_extension::cf XML features"/> > <action application="set" data="ringback=${us-ring}"/> > <action application="set" data="transfer_ringback=$${hold_music}"/> > <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="hash" > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > <action application="hash" > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > <action application="set" > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> >  > <action application="hash" > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > <action application="bridge" > data="sofia/internal/${dialed_extension}%${domain_name}"/> > <action application="answer"/> > <action application="sleep" data="1000"/> > <action application="bridge" data="loopback/app=voicemail:default > ${domain_name} ${dialed_extension}"/> > </condition> > </extension> > > > Old: > > <extension name="Local_Extension"> > <condition field="destination_number" expression="^(10[01][0-9])$"> > <action application="set" data="dialed_extension=$1"/> > <action application="export" data="dialed_extension=$1"/> >  > <action application="bind_meta_app" data="1 b s > execute_extension::dx XML features"/> > <action application="bind_meta_app" data="2 b s > record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > <action application="bind_meta_app" data="3 b s > execute_extension::cf XML features"/> > <action application="set" data="ringback=${us-ring}"/> > <action application="set" data="transfer_ringback=$${hold_music}"/> > <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="hash" > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> > <action application="hash" > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > <action application="set" > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> >  > <action application="hash" > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > <action application="bridge" data="user/${dialed_extension}@ > ${domain_name}"/> > <action application="answer"/> > <action application="sleep" data="1000"/> > <action application="bridge" data="loopback/app=voicemail:default > ${domain_name} ${dialed_extension}"/> > </condition> > </extension> > > :-D > > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 30 Jun 2010 22:37:15 -0500 > Subject: Re: [Freeswitch-users] Default Dial Plan: action application > bridge > you must have changed more than you think. > > you might want to diff you configs against the in-tree ones. > > > On Wed, Jun 30, 2010 at 10:19 PM, Deya M <deya787(a)gmail.com> wrote: > >> Hi, >> >> In the default dial plan, with two extensions defined, 1000 and 1001, when >> I call from 1000 to 1001, I always get Voicemail, using the default config >> files: conf/dialplan/default.xml >> >> I changed the following from : >> >> * <action application="bridge" data="user/${dialed_extension}@ >> ${domain_name}"/>* >> TO >> *<action application="bridge" >> data="sofia/internal/${dialed_extension}%${domain_name}"/>* >> >> and it did work! Not sure if the first / default one, * <action >> application="bridge" data="user/${dialed_extension}@${domain_name}"/> *should >> work, but something is missing ? >> >> >> New: >> <extension name="Local_Extension"> >> <condition field="destination_number" expression="^(10[01][0-9])$"> >> <action application="set" data="dialed_extension=$1"/> >> <action application="export" data="dialed_extension=$1"/> >>  >> <action application="bind_meta_app" data="1 b s >> execute_extension::dx XML features"/> >> <action application="bind_meta_app" data="2 b s >> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >> <action application="bind_meta_app" data="3 b s >> execute_extension::cf XML features"/> >> <action application="set" data="ringback=${us-ring}"/> >> <action application="set" >> data="transfer_ringback=$${hold_music}"/> >> <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="hash" >> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> <action application="hash" >> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> <action application="set" >> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >>  >> <action application="hash" >> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> <action application="bridge" >> data="sofia/internal/${dialed_extension}%${domain_name}"/> >> <action application="answer"/> >> <action application="sleep" data="1000"/> >> <action application="bridge" data="loopback/app=voicemail:default >> ${domain_name} ${dialed_extension}"/> >> </condition> >> </extension> >> >> >> Old: >> >> <extension name="Local_Extension"> >> <condition field="destination_number" expression="^(10[01][0-9])$"> >> <action application="set" data="dialed_extension=$1"/> >> <action application="export" data="dialed_extension=$1"/> >>  >> <action application="bind_meta_app" data="1 b s >> execute_extension::dx XML features"/> >> <action application="bind_meta_app" data="2 b s >> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> >> <action application="bind_meta_app" data="3 b s >> execute_extension::cf XML features"/> >> <action application="set" data="ringback=${us-ring}"/> >> <action application="set" >> data="transfer_ringback=$${hold_music}"/> >> <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="hash" >> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> >> <action application="hash" >> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> >> <action application="set" >> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}"/> >>  >> <action application="hash" >> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> <action application="bridge" data="user/${dialed_extension}@ >> ${domain_name}"/> >> <action application="answer"/> >> <action application="sleep" data="1000"/> >> <action application="bridge" data="loopback/app=voicemail:default >> ${domain_name} ${dialed_extension}"/> >> </condition> >> </extension> >> >> :-D >> >> >> _______________________________________________ >> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> > pstn:+19193869900 > > _______________________________________________ > 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

Create a external FXO Trunk - A200
by Deya M 01 Jul '10

01 Jul '10

Hi, San A200 FXO is woking with wanpipe, and freeswitch detects and init : 2010-07-01 09:07:15.427539 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'loopback' 2010-07-01 09:07:15.428061 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/modules.conf. 2010-07-01 09:07:15.428125 [NOTICE] zap_io.c:3058 Modules configured: 1 2010-07-01 09:07:15.428137 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/openzap.conf. 2010-07-01 09:07:15.428161 [DEBUG] zap_io.c:2588 found config for span 2010-07-01 09:07:15.428404 [INFO] zap_io.c:2859 Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] 2010-07-01 09:07:15.428418 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/wanpipe.conf. 2010-07-01 09:07:15.428467 [INFO] zap_io.c:2605 auto-loaded 'wanpipe' 2010-07-01 09:07:15.428494 [DEBUG] zap_io.c:2626 created span 1 (FXO) of type wanpipe 2010-07-01 09:07:15.428503 [DEBUG] zap_io.c:2639 span 1 [name]=[OpenZAP] 2010-07-01 09:07:15.428510 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:1] 2010-07-01 09:07:15.428518 [DEBUG] zap_io.c:2668 setting trunk type to 'FXO' start(KEWL) 2010-07-01 09:07:15.428570 [INFO] ozmod_wanpipe.c:330 configuring device s1c1 as OpenZAP device 1:1 fd:41 DTMF: hardware 2010-07-01 09:07:15.428587 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:2] 2010-07-01 09:07:15.428632 [INFO] ozmod_wanpipe.c:330 configuring device s1c2 as OpenZAP device 1:2 fd:42 DTMF: hardware 2010-07-01 09:07:15.428643 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:3] 2010-07-01 09:07:15.428695 [INFO] ozmod_wanpipe.c:330 configuring device s1c3 as OpenZAP device 1:3 fd:43 DTMF: hardware 2010-07-01 09:07:15.428707 [DEBUG] zap_io.c:2639 span 1 [fxo-channel]=[1:4] 2010-07-01 09:07:15.428748 [INFO] ozmod_wanpipe.c:330 configuring device s1c4 as OpenZAP device 1:4 fd:44 DTMF: hardware 2010-07-01 09:07:15.428774 [INFO] zap_io.c:2783 Configured 4 channel(s) 2010-07-01 09:07:15.428920 [INFO] zap_io.c:2876 Loading SIG from /usr/local/freeswitch/mod/ozmod_analog.so 2010-07-01 09:07:15.428932 [INFO] zap_io.c:2992 auto-loaded 'analog' 2010-07-01 09:07:15.428943 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:1 2010-07-01 09:07:15.428957 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:2 2010-07-01 09:07:15.428963 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:3 2010-07-01 09:07:15.428970 [DEBUG] ozmod_analog.c:184 Enabling call waiting for channel 1:4 2010-07-01 09:07:15.428977 [DEBUG] zap_config.c:56 Configuration file is /usr/local/freeswitch/conf/tones.conf. 2010-07-01 09:07:15.429003 [DEBUG] zap_io.c:551 added tone generation [dial] = [v=-7;%(1000,0,350,440)] 2010-07-01 09:07:15.429028 [DEBUG] zap_io.c:549 added tone detect [dial] = [350,440] 2010-07-01 09:07:15.429037 [DEBUG] zap_io.c:551 added tone generation [ring] = [v=-7;%(2000,4000,440,480)] 2010-07-01 09:07:15.429045 [DEBUG] zap_io.c:549 added tone detect [ring] = [440,480] 2010-07-01 09:07:15.429053 [DEBUG] zap_io.c:551 added tone generation [busy] = [v=-7;%(500,500,480,620)] 2010-07-01 09:07:15.429062 [DEBUG] zap_io.c:549 added tone detect [busy] = [480,620] 2010-07-01 09:07:15.429070 [DEBUG] zap_io.c:551 added tone generation [attn] = [v=0;%(100,100,1400,2060,2450,2600)] 2010-07-01 09:07:15.429078 [DEBUG] zap_io.c:549 added tone detect [attn] = [1400,2060,2450,2600] 2010-07-01 09:07:15.429086 [DEBUG] zap_io.c:551 added tone generation [callwaiting-sas] = [v=0;%(300,0,440)] 2010-07-01 09:07:15.429095 [DEBUG] zap_io.c:549 added tone detect [callwaiting-sas] = [440] 2010-07-01 09:07:15.429104 [DEBUG] zap_io.c:551 added tone generation [callwaiting-cas] = [v=0;%(80,0,2750,2130)] 2010-07-01 09:07:15.429113 [DEBUG] zap_io.c:549 added tone detect [callwaiting-cas] = [2750,2130] 2010-07-01 09:07:15.429125 [DEBUG] zap_io.c:549 added tone detect [fail1] = [913.8] 2010-07-01 09:07:15.429134 [DEBUG] zap_io.c:549 added tone detect [fail2] = [1370.6] 2010-07-01 09:07:15.429141 [DEBUG] zap_io.c:549 added tone detect [fail3] = [1776.7] 2010-07-01 09:07:15.429219 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'openzap' 2010-07-01 09:07:15.429254 [NOTICE] switch_loadable_module.c:251 Adding Application 'disable_ec' 2010-07-01 09:07:15.429290 [NOTICE] switch_loadable_module.c:251 Adding Application 'disable_dtmf' 2010-07-01 09:07:15.429332 [NOTICE] switch_loadable_module.c:251 Adding Application 'enable_dtmf' 2010-07-01 09:07:15.429381 [NOTICE] switch_loadable_module.c:273 Adding API Function 'oz' openzap.conf: [span wanpipe FXO] name => OpenZAP fxo-channel => 1:1 fxo-channel => 1:2 fxo-channel => 1:3 fxo-channel => 1:4 openzap.conf.xml: configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <boost_spans> </boost_spans> <analog_spans> <span id="FXO">  <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> </analog_spans> </configuration> wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1 | N/A | A200/A400/B600/B700| 177 | 4 | 1 | N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status | wanpipe1 | A-ANALOG | N/A | Connected | My questions, is how can I use this trunk, line 1-4, to dial ? I couldn't find a similar example using A200 Analog, could you please send me an example? Thanks, -:D

1 0

Using fifo_orbit_announce For On-hook Agent
by afshin afzali 01 Jul '10

01 Jul '10

Hi FreeSWITCH, I want to use fifo_orbit_announce to play specific agent greeting to his caller (can not insert it to caller dialplan). As my agents are on-hook (they use extensions to login / logout of queues) , I don't know if I could set this variable in login extension! If I can not use this way, is there any other way? appreciate all comments, BEST, -- afshin

4 6