Re: [Freeswitch-users] A little Q931 tool
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, ups, main message of me wasn't in my mail. Of course you can have it if someone wants it. regards Helmut On 24.09.2009 19:22, Michael Collins wrote: Excellent, thanks! -MC On Thu, Sep 24, 2009 at 4:50 AM, Helmut Kuper helmut.ku...@ewetel.de mailto:helmut.ku...@ewetel.de wrote: Hello, I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C (linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps and puts them in a .pcap file which is directly readable and decodeable by Wireshark/tshark. The big plus here is, that you are able to get an isdn Q931 trace even of a call in the past (as long as you have FS in DEBUG loglevel). regards Helmut ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKvGWw4tZeNddg3dwRAsUlAJ9E7kyJ7EE6iQVWqimrdAZqU5eZfACfesP6 Fp9F4HelsMbtVoKUNJFvSh0= =SYdk -END PGP SIGNATURE- ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] How do I get freeswitch variable which has DTMF values.
Dear Sir, If I disable the async mode in socket, the playAndGetDigits doesn't exit after getting the DTMF value. It exit after time out seconds. But I need to exit when DTMF digit is got. My subroutine call is, $conn-playAndGetDigits(1,1,1,8000,'#',$play_list,ivr/ivr-please.wav,res,\\d+); Is there any way to overcome this problem? Please help me... On Fri, Sep 25, 2009 at 10:52 AM, Brian West br...@freeswitch.org wrote: Or use the socket without async so that it blocks till the action is complete. /b On Sep 25, 2009, at 12:13 AM, velusamy velu wrote: To get the freeswitch variable I used getVar subroutine which is defined in ESL::IVR.pm file. When I print that digits, Perl program prints empty value while playing the menu itself. If I need to get the DTMF value I need to wait the perl program. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How do I get freeswitch variable which has DTMF values.
I beg to differ on that one.. what distro are you on? I use this all over the place and it works so i'm concerned about this. /b On Sep 25, 2009, at 1:55 AM, velusamy velu wrote: Dear Sir, If I disable the async mode in socket, the playAndGetDigits doesn't exit after getting the DTMF value. It exit after time out seconds. But I need to exit when DTMF digit is got. My subroutine call is, $conn-playAndGetDigits (1,1,1,8000,'#',$play_list,ivr/ivr-please.wav,res,\\d+); Is there any way to overcome this problem? Please help me... ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] How do I get freeswitch variable which has DTMF values.
Dear Sir, Please pardon me for this question. Now I corrected my mistake. It is working fine. Thank you vrey much for your valuable help.. On Fri, Sep 25, 2009 at 12:30 PM, Brian West br...@freeswitch.org wrote: I beg to differ on that one.. what distro are you on? I use this all over the place and it works so i'm concerned about this. /b On Sep 25, 2009, at 1:55 AM, velusamy velu wrote: Dear Sir, If I disable the async mode in socket, the playAndGetDigits doesn't exit after getting the DTMF value. It exit after time out seconds. But I need to exit when DTMF digit is got. My subroutine call is, $conn-playAndGetDigits(1,1,1,8000,'#',$play_list,ivr/ivr-please.wav,res, \\d+); Is there any way to overcome this problem? Please help me... ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages.
On Fri, Sep 25, 2009 at 9:52 AM, Jason White ja...@jasonjgw.net wrote: [Just catching up on this thread.] William King quentus...@gmail.com wrote: I would be more than happy to share the code I use. Here is the git repo: http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/ When you would like your changes to the Debian build infrastructure in FreeSWITCH to be tested on Debian Sid, I'll gladly volunteer, excluding any modules that depend on proprietary software that I don't have and don't want, e.g., Skype. If you don't use mod_skypiax then no Skype client software is required. There is no dependency on the package level. I would also like to see these changes integrated into the FreeSWITCH repository to replace what is currently in the debian directory, once you have a version that is well tested. It would be great if William's changes are committed to FreeSWITCH svn repository. Anyway, debian folder in /trunk is outdated (has not been updated since FreeSWITCH 1.0.3). - Dmitry Bely ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Transfer hangs.
Just tried that, but that simply results in *nothing* happening. It processes the meta digit, but then it just goes on, without transfering at all, nor if the original B-leg hangs up - that just results in the A-leg hanging up as well. 2009/9/25 Anthony Minessale anthony.miness...@gmail.com: in that case, it's probably a delay in the media stream where the app is queued when you press the key try updating to trunk and add the new i flag to the flags param i.e. 1 b ai transfer::ff-transfer XML public On Thu, Sep 24, 2009 at 2:04 PM, Harry Vangberg ha...@vangberg.name wrote: Not exactly, as I said, if the original B-leg doesn't hang up, it will wait 20 second before transfering to the new extension (check the timestamps!) - but if the original B leg hangs up, it gets transfered to the extension immediately. Look at this: 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042 sofia/external/hemmel...@129.142.224.250 Processing meta digit '1' [transfer::ff-transfer XML public] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228 sofia/external/hemmel...@129.142.224.250 receive message [UNBRIDGE] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228 sofia/external/46934488 receive message [UNBRIDGE] 2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540 sofia/external/46934488 Command Execute playback(local_stream://moh) EXECUTE sofia/external/46934488 playback(local_stream://moh) 2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown source moh, trying 'default' 2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source default 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231 sofia/external/46934488 receive message [BRIDGE] 2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540 sofia/external/hemmel...@129.142.224.250 Command Execute transfer(ff-transfer XML public) EXECUTE sofia/external/hemmel...@129.142.224.250 transfer(ff-transfer XML public) From 18:29:48 to 19:30:09 nothing happens - it's first then it's transferred to the new extension, and first after that that the new B-leg will even get called. 2009/9/24 Anthony Minessale anthony.miness...@gmail.com: because it's waiting for the other party to answer if you want to hear ringback or music while you are waiting see: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones specifically transfer_ringback On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg ha...@vangberg.name wrote: Hello My setup is this (I've simplified everything, because a lot of my logic isn't necesarry for showcasing this): A calls in, transfer is bound as meta app, B is bridged. When the meta app is processed, the call is transfered to a new extension, which rebridges A. But! After triggering the meta app, it hangs 20 seconds, until transfering to the new extension, unless the B-leg hangs up manually. It should be noted that I've set dtmf-type=sip-info, as I would like to bypass media–if there's a better solution to get DTMF events while bypassing media, please say so, as I know the SIP INFO solution is kinda havoced. This is my dialplan: include context name=public extension name=ff-ivr condition field=destination_number expression=^(\d+)$ action application=answer / action application=bind_meta_app data=1 b a transfer::ff-transfer XML public / action application=bridge data=sofia/gateway/gw1.fonet.dk/46934488 / /condition /extension extension name=ff-transfer condition field=destination_number expression=^ff-transfer$ action application=bridge data=sofia/gateway/gw1.fonet.dk/31354228 / /condition /extension ... /context /include A full trace of a session with A calling in, B answering, B triggering meta app, waiting for transfer, and finally bridge to C is attached. This is using freeswitch-tr...@14962 ___ FreeSWITCH-users mailing list 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 -- Anthony Minessale II FreeSWITCH
Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo
Anthony, I am embarassed to say that it's already documented in the mod_unimrcp wiki page now I come to look at it ... I can't understand how we didn't see that. Must try harder. I have however added the mod_unimrcp page to the Modules page in the wiki - it wasn't there before, and I have marked the mod_openmrcp page as DEPRECATED. Hope that's ok. If you like we can put together something for the currently 'TBD' javascript section in the next few days. From: freeswitch-users-boun...@lists.freeswitch.org [freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale [anthony.miness...@gmail.com] Sent: 24 September 2009 20:01 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo if you find the time, can you add that to the wiki too? On Thu, Sep 24, 2009 at 12:07 PM, Jim Page jim.p...@redmatter.commailto:jim.p...@redmatter.com wrote: Hi Arsen Thanks for your message – it inspired us to do what we should have done in the first place, and look at the code. The problem we were having was related to grammar files not being available locally. Now we have discovered the “builtin:” keyword we are up and running :) Many thanks Jim From: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Arsen Chaloyan Sent: 24 September 2009 18:24 To: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo Hi Jim, From conceptual viewpoint, mod_unimrcp is just an alternate implementation of an abstract ASR/TTS interface FreeSWITCH provides. Therefore you can use it exactly the same way as other ASR/TTS modules. See scripts/javascript/ps_pizza.js in FS tree for a working example. The only thing you should know and change there is module name var asr = new SpeechDetect(session, pocketsphinx); var asr = new SpeechDetect(session, unimrcp); Typically you can specify any grammar your MRCP server supports. From: Jim Page jim.p...@redmatter.commailto:jim.p...@redmatter.com To: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Sent: Thursday, September 24, 2009 1:25:20 PM Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo Hi All Has anyone had any experience doing ASR with mod_unimrcp in javascript? In particular, how do you deal with grammars? A simple piece of demo code would be massively appreciated - the documentation on mod_unimrcp ASR javascript bindings is TBD, which I assume means 'to be documented' ... unless it means 'to be developed' ... Also is anyone aware of a vlingo integration for freeswitch? All the best Jim ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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@lists.freeswitch.orgmailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.commailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.commailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.nethttp://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgmailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orgmailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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] Ringback when running G729 codec
Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) extension name=inbound_routing condition field=destination_number expression=^4420885767(0\d)$ action application=set data=ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options/ action application=set data=instant_ringback=true/ action application=bridge data=user/10$1/ /condition /extension when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success l...@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1...@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1...@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1...@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that version it
works as a charm.
for some reason i cannot get it right in 1.0.4 and trunk.
Actually, what i'm doing is to subscribe to events (within a custom module)
and try to get timestamps... I started having issues when i moved to trunk.
To be sure that i'm not doing something wrong, i configured mod_cdr_csv to
dump CDRs. Well it turned out this module doesn't work as well in the trunk.
Can it be because of AMD opteron + Debian 5.0 enviorment?
There is something in the 1.0.4/trunk version that is wrong for that kind of
event/CDR.
T.
On Fri, Sep 25, 2009 at 6:44 AM, Michael Jerris m...@jerris.com wrote:
Can you get these same values in xml-cdr? I don't think csv was ever
intended to work with different cdrs for a and b leg, it was more intended
as a more familiar interface for those coming over from asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in
my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both
a and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will always be appended to log-base --
!--param name=log-base value=/var/log/--
param name=default-template value=example/
!-- This is like the info app but after the call is hung up --
!--param name=debug value=true/--
param name=rotate-on-hup value=true/
!-- may be a b or ab --
param name=legs value=ab/
/settings
templates
template name=sqlINSERT INTO cdr VALUES
(${caller_id_name},${caller_id_number},${destination_number},${context},${s
tart_stamp},${answer_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode}
);/template
template
name=example${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec}/template
template
name=snom${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answer_
stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode},${read_codec},${wr
ite_codec},${sip_user_agent},${call_clientcode},${sip_rtp_rxstat},${sip_rtp_txstat},${sofia_record_file}/template
template
name=linksys${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec},${sip_user_agent},${sip_p_rtp_stat}/template
template
name=asterisk${accountcode},${caller_id_number},${destination_number},${context},${caller_id},${channel_
name},${bridge_channel},${last_app},${last_arg},${start_stamp},${answer_stamp},${end_stamp},${duration},${billsec},
${hangup_cause},${amaflags},${uuid},${userfield}/template
/templates
/configuration
call flow is the following:
CALLER = FS = CALLED
FS answers the call from CALLER, plays an announcement and bridges towards
CALLED.
I get different behavior when the call is released by Caller and by Called.
Released by Caller: the CDR is ok having all timestamps
OK CDR:
Outbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:02:48,2009-09-24 12:02:54,2009-09-24
12:03:01,13,7,NORMAL_CLEARING,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Inbound LEG = 016659280,016659280,05000403,public,2009-09-24
12:02:27,2009-09-24 12:02:41,2009-09-24
12:03:01,34,20,NORMAL_CLEARING,5d530192-a8f1-11de-962a-e328afdb9d8d,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
Released by Called: the CDR is NOT OK as timestamps are missing
NOT OK CDR:
Inbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:05:20,2009-09-24 12:05:30,2009-09-24
12:05:39,19,9,NORMAL_CLEARING,c479411a-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Outbound LEG =016659280,016659280,015000403,public,*,,,*
0,0,NORMAL_CLEARING,b82f2046-a8f1-11de-962a-e328afdb9d8d,c479411a-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
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Re: [Freeswitch-users] Transfer hangs.
For good measure, this is with tr...@14973 2009/9/25 Harry Vangberg ha...@vangberg.name: Just tried that, but that simply results in *nothing* happening. It processes the meta digit, but then it just goes on, without transfering at all, nor if the original B-leg hangs up - that just results in the A-leg hanging up as well. 2009/9/25 Anthony Minessale anthony.miness...@gmail.com: in that case, it's probably a delay in the media stream where the app is queued when you press the key try updating to trunk and add the new i flag to the flags param i.e. 1 b ai transfer::ff-transfer XML public On Thu, Sep 24, 2009 at 2:04 PM, Harry Vangberg ha...@vangberg.name wrote: Not exactly, as I said, if the original B-leg doesn't hang up, it will wait 20 second before transfering to the new extension (check the timestamps!) - but if the original B leg hangs up, it gets transfered to the extension immediately. Look at this: 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042 sofia/external/hemmel...@129.142.224.250 Processing meta digit '1' [transfer::ff-transfer XML public] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228 sofia/external/hemmel...@129.142.224.250 receive message [UNBRIDGE] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228 sofia/external/46934488 receive message [UNBRIDGE] 2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540 sofia/external/46934488 Command Execute playback(local_stream://moh) EXECUTE sofia/external/46934488 playback(local_stream://moh) 2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown source moh, trying 'default' 2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source default 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231 sofia/external/46934488 receive message [BRIDGE] 2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540 sofia/external/hemmel...@129.142.224.250 Command Execute transfer(ff-transfer XML public) EXECUTE sofia/external/hemmel...@129.142.224.250 transfer(ff-transfer XML public) From 18:29:48 to 19:30:09 nothing happens - it's first then it's transferred to the new extension, and first after that that the new B-leg will even get called. 2009/9/24 Anthony Minessale anthony.miness...@gmail.com: because it's waiting for the other party to answer if you want to hear ringback or music while you are waiting see: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones specifically transfer_ringback On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg ha...@vangberg.name wrote: Hello My setup is this (I've simplified everything, because a lot of my logic isn't necesarry for showcasing this): A calls in, transfer is bound as meta app, B is bridged. When the meta app is processed, the call is transfered to a new extension, which rebridges A. But! After triggering the meta app, it hangs 20 seconds, until transfering to the new extension, unless the B-leg hangs up manually. It should be noted that I've set dtmf-type=sip-info, as I would like to bypass media–if there's a better solution to get DTMF events while bypassing media, please say so, as I know the SIP INFO solution is kinda havoced. This is my dialplan: include context name=public extension name=ff-ivr condition field=destination_number expression=^(\d+)$ action application=answer / action application=bind_meta_app data=1 b a transfer::ff-transfer XML public / action application=bridge data=sofia/gateway/gw1.fonet.dk/46934488 / /condition /extension extension name=ff-transfer condition field=destination_number expression=^ff-transfer$ action application=bridge data=sofia/gateway/gw1.fonet.dk/31354228 / /condition /extension ... /context /include A full trace of a session with A calling in, B answering, B triggering meta app, waiting for transfer, and finally bridge to C is attached. This is using freeswitch-tr...@14962 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages.
Dmitry Bely ha scritto: It would be great if William's changes are committed to FreeSWITCH svn repository. Anyway, debian folder in /trunk is outdated (has not been updated since FreeSWITCH 1.0.3). Just updated to build and package some new modules MAx ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG
nothing I can think of, set up a test box that is not in production
and lets figure out what is wrong.
Mike
On Sep 25, 2009, at 7:22 AM, Tihomir Culjaga wrote:
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that
version it works as a charm.
for some reason i cannot get it right in 1.0.4 and trunk.
Actually, what i'm doing is to subscribe to events (within a custom
module) and try to get timestamps... I started having issues when i
moved to trunk. To be sure that i'm not doing something wrong, i
configured mod_cdr_csv to dump CDRs. Well it turned out this module
doesn't work as well in the trunk.
Can it be because of AMD opteron + Debian 5.0 enviorment?
There is something in the 1.0.4/trunk version that is wrong for that
kind of event/CDR.
T.
On Fri, Sep 25, 2009 at 6:44 AM, Michael Jerris m...@jerris.com
wrote:
Can you get these same values in xml-cdr? I don't think csv was
ever intended to work with different cdrs for a and b leg, it was
more intended as a more familiar interface for those coming over
from asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on latest trunk and for some reason i cannot get timestamps
dumped in my cdrs. I use mod_cdr_csv with default settings plus i
enabled to get both a and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will always be appended to log-base --
!--param name=log-base value=/var/log/--
param name=default-template value=example/
!-- This is like the info app but after the call is hung up --
!--param name=debug value=true/--
param name=rotate-on-hup value=true/
!-- may be a b or ab --
param name=legs value=ab/
/settings
templates
template name=sqlINSERT INTO cdr VALUES ($
{caller_id_name},${caller_id_number},${destination_number},$
{context},${s
tart_stamp},${answer_stamp},${end_stamp},${duration},$
{billsec},${hangup_cause},${uuid},${bleg_uuid}, $
{accountcode}
);/template
template name=example${caller_id_name},$
{caller_id_number},${destination_number},${context},$
{start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},$
{hangup_cause},${uuid},${bleg_uuid},${accountcode},$
{read_codec},${
write_codec}/template
template name=snom${caller_id_name},$
{caller_id_number},${destination_number},${context},$
{start_stamp},${answer_
stamp},${end_stamp},${duration},${billsec},$
{hangup_cause},${uuid},${bleg_uuid}, ${accountcode},$
{read_codec},${wr
ite_codec},${sip_user_agent},${call_clientcode},$
{sip_rtp_rxstat},${sip_rtp_txstat},${sofia_record_file}/
template
template name=linksys${caller_id_name},$
{caller_id_number},${destination_number},${context},$
{start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},$
{hangup_cause},${uuid},${bleg_uuid},${accountcode},$
{read_codec},${
write_codec},${sip_user_agent},${sip_p_rtp_stat}/template
template name=asterisk${accountcode},$
{caller_id_number},${destination_number},${context},$
{caller_id},${channel_
name},${bridge_channel},${last_app},${last_arg},$
{start_stamp},${answer_stamp},${end_stamp},${duration},$
{billsec},
${hangup_cause},${amaflags},${uuid},${userfield}/template
/templates
/configuration
call flow is the following:
CALLER = FS = CALLED
FS answers the call from CALLER, plays an announcement and bridges
towards CALLED.
I get different behavior when the call is released by Caller and by
Called.
Released by Caller: the CDR is ok having all timestamps
OK CDR:
Outbound LEG =
016659280,016659280,0914392122,public,2009-09-24
12:02:48,2009-09-24 12:02:54,2009-09-24
12:03:01,13,7,NORMAL_CLEARING,699cc2d0-a8f1-11de-962a-
e328afdb9d8d,,,PCMA,PCMA
Inbound LEG =
016659280,016659280,05000403,public,2009-09-24
12:02:27,2009-09-24 12:02:41,2009-09-24
12:03:01,34,20,NORMAL_CLEARING,5d530192-a8f1-11de-962a-
e328afdb9d8d,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
Released by Called: the CDR is NOT OK as timestamps are missing
NOT OK CDR:
Inbound LEG =
016659280,016659280,0914392122,public,2009-09-24
12:05:20,2009-09-24 12:05:30,2009-09-24
12:05:39,19,9,NORMAL_CLEARING,c479411a-a8f1-11de-962a-
e328afdb9d8d,,,PCMA,PCMA
Outbound LEG
=
016659280
,016659280
,015000403,public0,0,NORMAL_CLEARING,b82f2046-
a8f1-11de-962a-e328afdb9d8d,c479411a-a8f1-11de-962a-
e328afdb9d8d,,PCMA,PCMA
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Re: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG
should i move this to the DEV mailing list ?
T.
On Fri, Sep 25, 2009 at 4:12 PM, Michael Jerris m...@jerris.com wrote:
nothing I can think of, set up a test box that is not in production and
lets figure out what is wrong.
Mike
On Sep 25, 2009, at 7:22 AM, Tihomir Culjaga wrote:
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that version
it works as a charm.
for some reason i cannot get it right in 1.0.4 and trunk.
Actually, what i'm doing is to subscribe to events (within a custom module)
and try to get timestamps... I started having issues when i moved to trunk.
To be sure that i'm not doing something wrong, i configured mod_cdr_csv to
dump CDRs. Well it turned out this module doesn't work as well in the trunk.
Can it be because of AMD opteron + Debian 5.0 enviorment?
There is something in the 1.0.4/trunk version that is wrong for that kind
of event/CDR.
T.
On Fri, Sep 25, 2009 at 6:44 AM, Michael Jerris m...@jerris.com wrote:
Can you get these same values in xml-cdr? I don't think csv was ever
intended to work with different cdrs for a and b leg, it was more intended
as a more familiar interface for those coming over from asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in
my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both
a and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will always be appended to log-base --
!--param name=log-base value=/var/log/--
param name=default-template value=example/
!-- This is like the info app but after the call is hung up --
!--param name=debug value=true/--
param name=rotate-on-hup value=true/
!-- may be a b or ab --
param name=legs value=ab/
/settings
templates
template name=sqlINSERT INTO cdr VALUES
(${caller_id_name},${caller_id_number},${destination_number},${context},${s
tart_stamp},${answer_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode}
);/template
template
name=example${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec}/template
template
name=snom${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answer_
stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode},${read_codec},${wr
ite_codec},${sip_user_agent},${call_clientcode},${sip_rtp_rxstat},${sip_rtp_txstat},${sofia_record_file}/template
template
name=linksys${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec},${sip_user_agent},${sip_p_rtp_stat}/template
template
name=asterisk${accountcode},${caller_id_number},${destination_number},${context},${caller_id},${channel_
name},${bridge_channel},${last_app},${last_arg},${start_stamp},${answer_stamp},${end_stamp},${duration},${billsec},
${hangup_cause},${amaflags},${uuid},${userfield}/template
/templates
/configuration
call flow is the following:
CALLER = FS = CALLED
FS answers the call from CALLER, plays an announcement and bridges towards
CALLED.
I get different behavior when the call is released by Caller and by
Called.
Released by Caller: the CDR is ok having all timestamps
OK CDR:
Outbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:02:48,2009-09-24 12:02:54,2009-09-24
12:03:01,13,7,NORMAL_CLEARING,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Inbound LEG = 016659280,016659280,05000403,public,2009-09-24
12:02:27,2009-09-24 12:02:41,2009-09-24
12:03:01,34,20,NORMAL_CLEARING,5d530192-a8f1-11de-962a-e328afdb9d8d,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
Released by Called: the CDR is NOT OK as timestamps are missing
NOT OK CDR:
Inbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:05:20,2009-09-24 12:05:30,2009-09-24
12:05:39,19,9,NORMAL_CLEARING,c479411a-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Outbound LEG =016659280,016659280,015000403,public,*,,,*
0,0,NORMAL_CLEARING,b82f2046-a8f1-11de-962a-e328afdb9d8d,c479411a-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
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Re: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages.
On Fri, Sep 25, 2009 at 3:38 PM, Massimo CtRiX Cetra ctrix...@navynet.it wrote: Dmitry Bely ha scritto: It would be great if William's changes are committed to FreeSWITCH svn repository. Anyway, debian folder in /trunk is outdated (has not been updated since FreeSWITCH 1.0.3). Just updated to build and package some new modules Can you also add the tts_commandline module? Patch attached Mathieu Parent debian-tts_commandline.diff Description: Binary data ___ FreeSWITCH-users mailing list 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] Transfer oddity
Hi guys,
I've got a strange situation that I'm at a loss to explain. With all
callers, I go through a dialplan where I check to see if they should
be a moderator, then transfer them to another which puts them into a
conference accordingly. This worked great on one server, but when I
copied the code to another server (both running CentOS), the transfer
no longer works properly. Here's a log snippet from the incorrectly
working server:
Dialplan: sofia/internal/14258291...@10.10.67.190 Regex (FAIL)
[hold_music] destination_number(7001) =~ /^$/ break=on-false
Dialplan: sofia/internal/14258291...@10.10.67.190 Regex (FAIL)
[hold_music] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=on-false
Dialplan: sofia/internal/14258291...@10.10.67.190 ANTI-Action
set(zrtp_enrollment=true)
Dialplan: sofia/internal/14258291...@10.10.67.190 ANTI-Action answer()
Dialplan: sofia/internal/14258291...@10.10.67.190 ANTI-Action
playback(/usr/local/freeswitch/sounds/vpbx/moh.wav)
2009-09-25 07:51:31.204920 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/14258291...@10.10.67.190) State Change CS_ROUTING -
CS_EXECUTE
2
(note that the 7001 in the first line is the number I chose for my dialplan)
On the working server, the first line is still there, but the second
(and further) is replaced by further checks to see if it might be my
conference dialplans, which is what I would expect. I looked into
dialplans/default.xml, and the code for the above is there, but let me
copy it here again to discuss:
extension name=hold_music
condition field=destination_number expression=^$/
condition field=${sip_has_crypto}
expression=^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$
action application=answer/
action application=execute_extension data=is_secure XML features/
action application=playback data=$${hold_music}/
!-- This really should be an IVR for zrtp enrollment but this
is just a demo--
anti-action application=set data=zrtp_enrollment=true/
anti-action application=answer/
anti-action application=playback data=$${hold_music}/
/condition
/extension
Now, the way I understand this, it says that if the number is , it
should check the 2nd condition (which says to play hold music in a
couple of different flavors), but if the number is NOT , it should
go past, not even checking the 2nd condition. This understanding is
corroborated by the working server, which does indeed skip past and
not check the 2nd condition. Does anyone know why a server might be
going into a conditional that it knows it failed on?
For what it's worth, both servers are running on the current trunk,
with the only change being the addition of flite.
BB
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[Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!
Come on in! sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org or via the good old PSTN at +1-213-799-1400 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!
as always, you can call skype the skypeuser skypiax5, then press 1 On Fri, Sep 25, 2009 at 6:15 PM, Michael Collins m...@freeswitch.org wrote: Come on in! sip:8...@conference.freeswitch.org or via the good old PSTN at +1-213-799-1400 ___ FreeSWITCH-users mailing list 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Bind to more than one ethernet interface
On Thu, Sep 24, 2009 at 12:29 PM, Anthony Minessale anthony.miness...@gmail.com wrote: moral of the story is, it's unwise to bind multiple ip to a server interface that uses UDP signalling and the SIP spec requires a UA to have one specific URL Yeah, I see that now. I wasn't thinking about it as carefully as I should have been. Thanks for the clarification. Best, Gabe ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Status of ubuntu/debian packages.
William King wrote: It has been removed from the dependencies. Thanks go to the reporter for finding the extra depends. A new round of builds just went out and built. Let me know if you find something else. Also mod_skypiax should be available. -William King Michael Jerris wrote: I can confirm you should not need the swig dependency at all for anything. Mike On Sep 24, 2009, at 1:49 PM, William King wrote: Hmm... That is interesting... swig is needed I believe only for the mod_perl or the esl modules. I'll find out more information and put it on the correct package. I will also update the mod_skypiax config files in the *.install files. -William King Dmitry Bely wrote: On Thu, Sep 24, 2009 at 2:14 AM, William King quentus...@gmail.com wrote: Sure, post it here and I'll add it in the next build in a few hours. See attached file. Unfortunately mod_skypiax author did not placed config files (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they are not included into freeswitch-config and should be added manually. BTW, why swig is a dependency for the source package? I recall Brian's post where he insists that swig is never needed to build Freeswitch. -William King Dmitry Bely wrote: On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle fr...@carmickle.com wrote: On Wed, Sep 23, Dmitry Bely wrote: Can you enable mod_skypiax in your debian package? We will be enabling as much as we can cleanly build on debian/ ubuntu. There will be a lot more to come. We will be breaking the mods and end points in to different packages so that you can install what you like. If you have something you would like to see in the package let us know. Also patches are welcome. Well, mod_skypiax just requires trivial one-line addition to debian/rules and debian/freeswitch.install. It builds OK. If the patch is required I can post it here. - Dmitry Bely ___ FreeSWITCH-users mailing list 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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Unfortunately the launchpad.net distribution doesn't seem to be able to mirror your builds, so there is nothing to download and install just yet. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Status of ubuntu/debian packages.
Which launchpad address are you using? I'm uploading to the freeswitch nightlies right now. And will post to the freeswitch tagged release ppa when 1.05 comes out, or when the pre* tags come out. -William King ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Status of ubuntu/debian packages.
William King wrote: Which launchpad address are you using? I'm uploading to the freeswitch nightlies right now. And will post to the freeswitch tagged release ppa when 1.05 comes out, or when the pre* tags come out. -William King Ah, perhaps that's why the keyserver update isn't working (times out.) I now see some files uploaded to the repository. They weren't there a couple hours ago. :) BTW, we're running Hardy 8.04. I'm hoping we can get our production version updated from that, but we've got a lot of clients fielded with 8.04 so at this point we're still running it for dev as well. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Status of ubuntu/debian packages.
This is for everyone using the ubuntu packages. DO NOT UPGRADE PAST HARDY FOR PRODUCTION. There is a major issue with all the newer systems. It has something to do with the threads. In my tests I have seen a box sit at 20% cpu usage with no calls on it at all. This bug isn't in hardy. I will be trying to get a package that will auto add the launchpad key to your apt keys. -William King Mark Sobkow wrote: William King wrote: Which launchpad address are you using? I'm uploading to the freeswitch nightlies right now. And will post to the freeswitch tagged release ppa when 1.05 comes out, or when the pre* tags come out. -William King Ah, perhaps that's why the keyserver update isn't working (times out.) I now see some files uploaded to the repository. They weren't there a couple hours ago. :) BTW, we're running Hardy 8.04. I'm hoping we can get our production version updated from that, but we've got a lot of clients fielded with 8.04 so at this point we're still running it for dev as well. ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Ringback when running G729 codec
fixed in latest trunk, please test thank you On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog d_ho...@ymail.com wrote: Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) extension name=inbound_routing condition field=destination_number expression=^4420885767(0\d)$ action application=set data=ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options/ action application=set data=instant_ringback=true/ action application=bridge data=user/10$1/ /condition /extension when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/ 442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success l...@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/ sip:1...@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/ 442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/ sip:1...@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/ sip:1...@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori ___ FreeSWITCH-users mailing list 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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] No ring tone while recording incoming call. Please help.
Hi guys,
I have solve my problem by adding
action application=ring_ready /
before
action application=set data=ringback=${us-ring}/
I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example
and noticed this line. Tried it and it works like a charm.
Thanks everybody, especially Brian and MC.
Igor
Brian,
Thank you very much for your reply.
I have tried to add transfer_ringback action, but it did not solve my
problem.
Destination phone is ringing, but the person who is calling does not
hear ringing tone in hte handset.
Is there anything in the logfile that can help you to identify the
problem?
What kind of system is the calling party connected to? It looks like a
180 is sent out by FS:
2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/
sip:main at 192.168.0.121:5060 entering state [proceeding][180]
At that point the server at the originating side *should* generate
pretend ringing for the calling phone. If that is not happening then
you need to see what's going on at the originating side. Is it a SIP
provider?
-MC
Closest I can see is:
2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1738 Raw
Codec Activation Success L16 at 8000hz 1 channel 20ms
2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1797 Play
Ringback Tone [%(2000,4000,440.0,480.0)]
2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232
sofia/external/
4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY]
2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel
sofia/internal/ sip:main at 192.168.0.121:5060 entering state
[proceeding][180]
2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready
sofia/internal/ sip:main at 192.168.0.121:5060!
2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel
sofia/external/
4163641113 at 67.205.74.164 entering state [terminated][487]
2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup
sofia/external/
4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL]
Thank you,
Igor
set ringback before record_session and also set transfer_ringback
because record_session causes an pre-answer.
/b
On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:
Hi,
I have trouble recording incoming calls with FreeSwitch.
I have followed the instruction from Misc. Dialplan Tools record
session
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_sessi
on) It works well for outgoing calls, but I have the problem with
incoming calls.
The person who is calling does not hear ring tone, he hears just
the silence until I pick up the phone. Everything else is working,
we can talk, conversation is recorded.
Here is a copy of my dialplan for incoming calls
/usr/local/freeswitch/conf/dialplan/public/voipms.xml
include
extension name=voipms !-- your provider or any name you'd
like to call it --
condition field=destination_number
expression=XX !-- your DID for this gateway--
action application=set data=RECORD_TITLE=Recording
$ {destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:
%M)}/
action application=set data=RECORD_COPYRIGHT=(c)
2009/
action application=set
data=RECORD_SOFTWARE=FreeSwitch/
action application=set
data=RECORD_ARTIST=FreeSwitch/
action application=set
data=RECORD_COMMENT=FreeSwitch/
action application=set data=RECORD_DATE=${strftime
(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_STEREO=true/
action application=set data=RECORD_ANSWER_REQ=true/
action application=set data=ringback=${us-ring}/
action application=record_session
data=$${base_dir}/
recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$
{destination_number}_${caller_id_number}.wav/
action application=bridge data=user/us...@$
{domain_name}/
/condition
/include
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Re: [Freeswitch-users] No ring tone while recording incoming call. Please help.
Doesn't make sense because ring_ready sends only a 180 set
ringback then pre_answer would make use do 183.
/b
On Sep 25, 2009, at 1:27 PM, Svetik VOIP wrote:
Hi guys,
I have solve my problem by adding
action application=ring_ready /
before
action application=set data=ringback=${us-ring}/
I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example
and noticed this line. Tried it and it works like a charm.
Thanks everybody, especially Brian and MC.
Igor
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[Freeswitch-users] Solved!. No ring tone while recording incoming call. Please help. Sorry for possible duplication.
Sorry for possible duplication, I am trying to figure how to reply to
the list properly.
Hi guys,
I have solve my problem by adding
action application=ring_ready /
before
action application=set data=ringback=${us-ring}/
I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example
and noticed this line. Tried it and it works like a charm.
Thanks everybody, especially Brian and MC.
Igor
Brian,
Thank you very much for your reply.
I have tried to add transfer_ringback action, but it did not solve my
problem.
Destination phone is ringing, but the person who is calling does not
hear ringing tone in hte handset.
Is there anything in the logfile that can help you to identify the
problem?
What kind of system is the calling party connected to? It looks like a
180 is sent out by FS:
2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/
sip:main at 192.168.0.121:5060 entering state [proceeding][180]
At that point the server at the originating side *should* generate
pretend ringing for the calling phone. If that is not happening then
you need to see what's going on at the originating side. Is it a SIP
provider?
-MC
Closest I can see is:
2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1738 Raw
Codec Activation Success L16 at 8000hz 1 channel 20ms
2009-09-22 17:18:05.02 [DEBUG] switch_ivr_originate.c:1797 Play
Ringback Tone [%(2000,4000,440.0,480.0)]
2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232
sofia/external/
4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY]
2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel
sofia/internal/ sip:main at 192.168.0.121:5060 entering state
[proceeding][180]
2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready
sofia/internal/ sip:main at 192.168.0.121:5060!
2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel
sofia/external/
4163641113 at 67.205.74.164 entering state [terminated][487]
2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup
sofia/external/
4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL]
Thank you,
Igor
set ringback before record_session and also set transfer_ringback
because record_session causes an pre-answer.
/b
On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:
Hi,
I have trouble recording incoming calls with FreeSwitch.
I have followed the instruction from Misc. Dialplan Tools record
session
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_sessi
on) It works well for outgoing calls, but I have the problem with
incoming calls.
The person who is calling does not hear ring tone, he hears just
the silence until I pick up the phone. Everything else is working,
we can talk, conversation is recorded.
Here is a copy of my dialplan for incoming calls
/usr/local/freeswitch/conf/dialplan/public/voipms.xml
include
extension name=voipms !-- your provider or any name you'd
like to call it --
condition field=destination_number
expression=XX !-- your DID for this gateway--
action application=set data=RECORD_TITLE=Recording
$ {destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:
%M)}/
action application=set data=RECORD_COPYRIGHT=(c)
2009/
action application=set
data=RECORD_SOFTWARE=FreeSwitch/
action application=set
data=RECORD_ARTIST=FreeSwitch/
action application=set
data=RECORD_COMMENT=FreeSwitch/
action application=set data=RECORD_DATE=${strftime
(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_STEREO=true/
action application=set data=RECORD_ANSWER_REQ=true/
action application=set data=ringback=${us-ring}/
action application=record_session
data=$${base_dir}/
recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$
{destination_number}_${caller_id_number}.wav/
action application=bridge data=user/us...@$
{domain_name}/
/condition
/include
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Re: [Freeswitch-users] Status of ubuntu/debian packages - keyserver timeout
Has the keyserver.ubuntu.com been uploaded with the key for the builds? I keep getting a timeout trying to download the key to our ring. apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 451AE93C ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Ringback when running G729 codec
will start testing soon thank you very much Ori From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Friday, September 25, 2009 7:30:34 PM Subject: Re: [Freeswitch-users] Ringback when running G729 codec fixed in latest trunk, please test thank you On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog d_ho...@ymail.com wrote: Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) extension name=inbound_routing condition field=destination_number expression=^4420885767(0\d)$ action application=set data=ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options/ action application=set data=instant_ringback=true/ action application=bridge data=user/10$1/ /condition /extension when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success l...@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1...@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1...@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1...@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori ___ FreeSWITCH-users mailing list 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Status of ubuntu/debian packages - keyserver timeout
keyserver.ubuntu.com is down. I have not seen any official word about it yet. -William King Mark Sobkow wrote: Has the keyserver.ubuntu.com been uploaded with the key for the builds? I keep getting a timeout trying to download the key to our ring. apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 451AE93C ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Transfer hangs.
If anybody else cares, this was fixed by 14983 (http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=14983) Thanks to Anthony. 2009/9/25 Harry Vangberg ha...@vangberg.name: For good measure, this is with tr...@14973 2009/9/25 Harry Vangberg ha...@vangberg.name: Just tried that, but that simply results in *nothing* happening. It processes the meta digit, but then it just goes on, without transfering at all, nor if the original B-leg hangs up - that just results in the A-leg hanging up as well. 2009/9/25 Anthony Minessale anthony.miness...@gmail.com: in that case, it's probably a delay in the media stream where the app is queued when you press the key try updating to trunk and add the new i flag to the flags param i.e. 1 b ai transfer::ff-transfer XML public On Thu, Sep 24, 2009 at 2:04 PM, Harry Vangberg ha...@vangberg.name wrote: Not exactly, as I said, if the original B-leg doesn't hang up, it will wait 20 second before transfering to the new extension (check the timestamps!) - but if the original B leg hangs up, it gets transfered to the extension immediately. Look at this: 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042 sofia/external/hemmel...@129.142.224.250 Processing meta digit '1' [transfer::ff-transfer XML public] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228 sofia/external/hemmel...@129.142.224.250 receive message [UNBRIDGE] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228 sofia/external/46934488 receive message [UNBRIDGE] 2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540 sofia/external/46934488 Command Execute playback(local_stream://moh) EXECUTE sofia/external/46934488 playback(local_stream://moh) 2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown source moh, trying 'default' 2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source default 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231 sofia/external/46934488 receive message [BRIDGE] 2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal sofia/external/hemmel...@129.142.224.250 [BREAK] 2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540 sofia/external/hemmel...@129.142.224.250 Command Execute transfer(ff-transfer XML public) EXECUTE sofia/external/hemmel...@129.142.224.250 transfer(ff-transfer XML public) From 18:29:48 to 19:30:09 nothing happens - it's first then it's transferred to the new extension, and first after that that the new B-leg will even get called. 2009/9/24 Anthony Minessale anthony.miness...@gmail.com: because it's waiting for the other party to answer if you want to hear ringback or music while you are waiting see: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones specifically transfer_ringback On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg ha...@vangberg.name wrote: Hello My setup is this (I've simplified everything, because a lot of my logic isn't necesarry for showcasing this): A calls in, transfer is bound as meta app, B is bridged. When the meta app is processed, the call is transfered to a new extension, which rebridges A. But! After triggering the meta app, it hangs 20 seconds, until transfering to the new extension, unless the B-leg hangs up manually. It should be noted that I've set dtmf-type=sip-info, as I would like to bypass media–if there's a better solution to get DTMF events while bypassing media, please say so, as I know the SIP INFO solution is kinda havoced. This is my dialplan: include context name=public extension name=ff-ivr condition field=destination_number expression=^(\d+)$ action application=answer / action application=bind_meta_app data=1 b a transfer::ff-transfer XML public / action application=bridge data=sofia/gateway/gw1.fonet.dk/46934488 / /condition /extension extension name=ff-transfer condition field=destination_number expression=^ff-transfer$ action application=bridge data=sofia/gateway/gw1.fonet.dk/31354228 / /condition /extension ... /context /include A full trace of a session with A calling in, B answering, B triggering meta app, waiting for transfer, and finally bridge to C is attached. This is using freeswitch-tr...@14962 ___ FreeSWITCH-users mailing
Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
There is a new function I checked in a little bit ago that lets you create any of the SWIGTYPE_p_xxx types - all you need is a pointer to the memory to represent whatever it is in native land. So with that, it's actually possible to call most or all of the functions. (Yes DRK, you can now go do XML binding.) But sure, it'd be nice to make a real .NET-ish layer. Async events seems like it wouldn't be hard, assuming FreeSWITCH delivers them that way? -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, September 24, 2009 10:26 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET There are a few other things I can think would be nice additions to mod_managed. Maybe an event handler that does not require a thread to be sitting and waiting for events trying in a loop would be nice, instead something that is triggered each time there is a certain event class triggered. Also, there has been some interest in doing full endpoint modules in mod_managed. exposing all the state handlers in .net like ways and having that all work would be quite interesting, but probably requires someone specific actually ready to write a module like that to be worthwhile. Mike On Sep 24, 2009, at 4:01 AM, Michael Giagnocavo wrote: Great - hopefully we'll meet on IRC or the conference sometime on Friday. Email me when you're on. A few questions I have: Clarity - I agree with you there, and thanks! Testability - is this even remotely practical? Looking at our FS code plugins, there's simply no way any amount of test environment code would get us to anything testable. We make tons of direct P/Invoke calls, and the whole model for what variables are set when, the state machine progression, etc. does not seem like something that we can hope to possibly model right. And it's subject to many external influences (all the modules you have loaded in FS). Logging is a pretty simple case, sure, we can make it not call FS for testing. But in a real app, it just seems that there are way too many dependencies, no? Maybe others who have apps written can chime in? Modularity - I agree there are two parts. But, I think they are pretty tightly coupled. The FS interface into unmanaged code is done via unmanaged code and is really clear: App, Api, ApiBackground. The other ways I can think of are FS-specific, such as XML binding interface and so on. But those are things we should just add to the mod_managed core and be done with. I'm thinking maybe we are talking about different things? Can you provide some user stories that we want to cover with a pluggable loader/executor/etc.? Thanks for putting up with me! -Michael From: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Thursday, September 24, 2009 12:32 AM To: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo m...@giagnocavo.netmailto:m...@giagnocavo.net wrote: Right off the bat: there can be tons of cleanup and refactoring, no doubt about that. Much of the current code is to satisfy my needs in production, which it does very well. The current base doesn't have anything wrong with it for sure, in fact, I learned a good bit about PInvoke. AppDomains, and In-Process Remoting in the last week. My refactoring had the following goals (in no particular order) - Testability - I'd really like to see a decent unit test suite on the more module so that we can change it with confidence. Also, it's been drilled into me that a testable design is a good design. - Clarity - Where possible, I extracted blocks of code that served a particular purpose so that purpose could be self-documenting in the method calls rather than mixed in. - Modularity - I wanted to make it easy to remove or add alternative behavior to the managed.dll. I'm a bit hesitant to go too far from the FreeSWITCH core as far as architecture goes. For instance, I'm not quite sure why'd we have our own managed logging subsystem that allows them to plug in other things that aren't part of FS. Either they should use the FS logging system, or use their own such as log4net. Or perhaps I don't see why we'd want this behavior. I completely agree, with the following caveats: 1) I'd like to see things testable. It's very hard to do isolation testing with classes making direct calls out to a static Log class that in turn pinvokes out to unmanaged code. 2) I'd like to allow folk to make changes to the default behavior (optimally) without recompiling managed.dll. One thing at issue here is that there are two principal purposes
Re: [Freeswitch-users] Ringback when running G729 codec
does it mean, if i encode my voice files in g729 i can use mod_nativefile to playback to a call using 729 codec? T. On Fri, Sep 25, 2009 at 8:30 PM, Anthony Minessale anthony.miness...@gmail.com wrote: fixed in latest trunk, please test thank you On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog d_ho...@ymail.com wrote: Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) extension name=inbound_routing condition field=destination_number expression=^4420885767(0\d)$ action application=set data=ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options/ action application=set data=instant_ringback=true/ action application=bridge data=user/10$1/ /condition /extension when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/ 442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success l...@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/ sip:1...@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/ 442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/ sip:1...@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1...@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori ___ FreeSWITCH-users mailing list 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Ringback when running G729 codec
yes /b On Sep 25, 2009, at 3:22 PM, Tihomir Culjaga wrote: does it mean, if i encode my voice files in g729 i can use mod_nativefile to playback to a call using 729 codec? T. ___ FreeSWITCH-users mailing list 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] Documentation Effort: Lost the Post
I had posted that I was interested in writing some documentation for FS or improving upon what I've found. I'd come up with a for dummies guide to get my crew started on FS and drew from various sources. Probably not worth much for engineers who are working on FS module code, but it might help someone wanting to get started using FS (getting off that other VoIP system). There was a recent couple of posts about this in last week or so, about some kind of effort to coordinate documentation creation and modification. I've deleted that thread, absentmindedly. Was that Brian? Michael? None of the above? Mike G. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Documentation Effort: Lost the Post
http://article.gmane.org/gmane.comp.telephony.freeswitch.user/16841 =) Michael Collins is pushing the documentation efforts. I need to get writing some more too. -- William ___ FreeSWITCH-users mailing list 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] XMPP (mod_dingaling) for Event s/Messaging
Hello all, I was wondering if anyone has used mod_dingaling for messaging rather than voice/video. Specifically, I would like to have FS send an XMPP message to an ActiveMQ server when it records a voicemail. Additionally I would like like to have CDR entries posted into the ActiveMQ server as calls are completed.It seems all the pieces are there, just not necessarily used for this task. My fall-back would be to use Event Socket to receive events and then forward them on to the MQ server. But if I could directly integrate it, that would be one less system to worry about.Any help would be appreciated.-pete ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Documentation Effort: Lost the Post
Thanks much, William. I'll post Michael off-line and see what's what. Regards, Mike G. On Fri, Sep 25, 2009 at 4:12 PM, William Suffill william.suff...@gmail.comwrote: http://article.gmane.org/gmane.comp.telephony.freeswitch.user/16841 =) Michael Collins is pushing the documentation efforts. I need to get writing some more too. -- William ___ FreeSWITCH-users mailing list 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@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] Sofia regiter to provider with multiple account
Dear All, How to config freeswitch for support this case ? 1. FS register to provider about 50 user account. (Each account can't support multiple call in same time) 2. FS Check account not inuse before call out. 3. User account should be round-robin BG Dome C. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] XMPP (mod_dingaling) for Events/Messaging
see chat_send api command and api_hangup_hook. In combination that might work. Mike On Sep 25, 2009, at 6:07 PM, Pete Mueller wrote: Hello all, I was wondering if anyone has used mod_dingaling for messaging rather than voice/video. Specifically, I would like to have FS send an XMPP message to an ActiveMQ server when it records a voicemail. Additionally I would like like to have CDR entries posted into the ActiveMQ server as calls are completed. It seems all the pieces are there, just not necessarily used for this task. My fall-back would be to use Event Socket to receive events and then forward them on to the MQ server. But if I could directly integrate it, that would be one less system to worry about. Any help would be appreciated. -pete ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sofia regiter to provider with multiple account
On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote: Dear All, How to config freeswitch for support this case ? 1. FS register to provider about 50 user account. (Each account can't support multiple call in same time) Sofia gateways 2. FS Check account not inuse before call out. mod_limit 3. User account should be round-robin what does this mean? ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Sofia regiter to provider with multiple account
2009/9/26 Michael Jerris m...@jerris.com: On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote: Dear All, How to config freeswitch for support this case ? 1. FS register to provider about 50 user account. (Each account can't support multiple call in same time) Sofia gateways 2. FS Check account not inuse before call out. mod_limit 3. User account should be round-robin what does this mean? Each user account have own balance (in provider). So i want to use all user ___ FreeSWITCH-users mailing list 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
