- FreeSWITCH-users - FreeSWITCH Mailing Lists

Seg Fault on Mac
by Jonathan Palley 25 Dec '07

25 Dec '07

Hoping someone can help debug this. I'm getting a seg fault (OS X 10.4) whenever the dialplan hits an expression with brackets in it. This happens on a fresh install, latest trunk, no modifications to config files. Here's what the console output looks like if you make the first item of the dialplan include a bracket: 2007-12-25 17:33:59 [DEBUG] sofia_glue.c:1447 sofia_glue_negotiate_sdp () Set 2833 dtmf payload to 101 2007-12-25 17:33:59 [DEBUG] sofia.c:1444 sofia_handle_sip_i_state() sofia/default/1000@192.168.1.4:5060 State Change CS_NEW -> CS_INIT 2007-12-25 17:33:59 [DEBUG] switch_core_session.c:633 switch_core_session_signal_state_change() Kill sofia/default/ 1000@192.168.1.4:5060 [BREAK] 2007-12-25 17:33:59 [DEBUG] switch_core_state_machine.c:402 switch_core_session_run() sofia/default/1000@192.168.1.4:5060 Running State Change CS_INIT 2007-12-25 17:33:59 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/default/1000@192.168.1.4:5060) State INIT 2007-12-25 17:33:59 [DEBUG] mod_sofia.c:87 sofia_on_init() sofia/ default/1000@192.168.1.4:5060 SOFIA INIT 2007-12-25 17:33:59 [DEBUG] mod_sofia.c:100 sofia_on_init() sofia/ default/1000@192.168.1.4:5060 State Change CS_INIT -> CS_RING 2007-12-25 17:33:59 [DEBUG] switch_core_session.c:633 switch_core_session_signal_state_change() Kill sofia/default/ 1000@192.168.1.4:5060 [BREAK] 2007-12-25 17:33:59 [DEBUG] switch_core_state_machine.c:402 switch_core_session_run() sofia/default/1000@192.168.1.4:5060 Running State Change CS_RING 2007-12-25 17:33:59 [DEBUG] switch_core_state_machine.c:417 switch_core_session_run() (sofia/default/1000@192.168.1.4:5060) State RING 2007-12-25 17:33:59 [DEBUG] mod_sofia.c:117 sofia_on_ring() sofia/ default/1000@192.168.1.4:5060 SOFIA RING 2007-12-25 17:33:59 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_ring() Standard RING sofia/default/ 1000@192.168.1.4:5060 2007-12-25 17:33:59 [INFO] mod_dialplan_xml.c:217 dialplan_hunt() Processing test->20! 2007-12-25 17:33:59 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test conditions ${numbering_plan}() =~ /^[0-9]$/ Here's what I get from gdb on the core file: #0 0x00000000 in _mh_dylib_header () No symbol table info available. So not that helpful and I don't know where to go from here... Thanks! JP

1 0

localized dialplans
by trixter aka Bret McDanel 21 Dec '07

21 Dec '07

I have redone the localized dialplan stuff, there is even a readme file. The scope of this is limited, its only to provide a replacement dialplan for what users would be used to in a given country. That way the way they used the PSTN telephone provider they can with freeswitch. Currently there is: NANPA (the 18 or so countries that are country code 1) Netherlands Spain - thanks to David Villasmil Venezuela - thanks to David Villasmil If you want to create a new one for your local area the readme gives full instructions on how. The rumor is that this is going to go into trunk soon so getting this filled out (it only takes 2 minutes) would be nice. look for it in SRC/scripts/contrib/trixter/dialplan in an svn repository near you. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you!

1 0

Conference tones
by Brian West 20 Dec '07

20 Dec '07

I need everyone to help pickout a default tone set for various tones outlined in phrase_en.xml for mod_conference. I'm going to be placing an order for more sound files with gmvoices to fill in any gaps so please try to go over voicemail or anything else that might be missing and let me know what we need to order. Thanks, /b

1 0

Voicemail
by Peder ＠ NetworkOblivion 20 Dec '07

20 Dec '07

Am I to assume that voicemail requires Cepstral? I setup a dialplan to test and I get the following debug when the call rolls to voicemail: 2007-12-04 21:33:33 [DEBUG] switch_ivr_play_say.c:252 switch_ivr_phrase_macro() Handle speak-text:[4211 is not available.] (en:en) 2007-12-04 21:33:33 [ERR] switch_core_speech.c:60 switch_core_speech_open() invalid speech module [cepstral]! 2007-12-04 21:33:33 [ERR] switch_ivr_play_say.c:1532 switch_ivr_speak_text() Invalid TTS module! 2007-12-04 21:33:33 [DEBUG] switch_ivr_play_say.c:117 switch_ivr_phrase_macro() No language specified - Using [en] 2007-12-04 21:33:33 [DEBUG] switch_ivr_play_say.c:252 switch_ivr_phrase_macro() Handle speak-text:[goodbye.] (en:en) 2007-12-04 21:33:33 [ERR] switch_core_speech.c:60 switch_core_speech_open() invalid speech module [cepstral]! 2007-12-04 21:33:33 [ERR] switch_ivr_play_say.c:1532 switch_ivr_speak_text() Invalid TTS module! I uncommented the cepstral line in modules.conf.xml and I get the following: 2007-12-04 21:52:08 [CRIT] switch_loadable_module.c:726 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_cepstral.so **/usr/local/freeswitch/mod/mod_cepstral.so: cannot open shared object file: No such file or directory** So I am guessing that Cepstral isn't included as part of freeswitch and I have to get it elsewhere and compile it to actually use voicemail. Is that correct? Peder

4 7

How to enable Call Waiting on Freeswitch UA???
by Irshad Ahamed 20 Dec '07

20 Dec '07

Hi, I couldn't hear any beeps when there is a second incoming call. I was of the notion that Call Waiting is enabled by default. If not, how should I enable it?? Any pointers buddies..... Thanks in advance... Best Regards, Irshad.

2 1

Instant Message Gateway
by Alois Komenda 20 Dec '07

20 Dec '07

Hello, how can I configure FreeSWITCH to act as a gateway for Instant Messages. For example it should translate a SIP Instant Message into an XMPP Instant Message. What I also want to know: Is there any way to react on an received IM with actions in the Dialplan? Thank you -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

2 3

How to make Attended Call Transfer as Transferor??
by Irshad Ahamed 20 Dec '07

20 Dec '07

Hi, I have installed Freeswitch successfully on my machine. I was trying to perform an Attended Call transfer to other endpoint in the same domain as a Transferor. But I couldn't find how exactly I should do that. It would be really great if someone helps me out in telling the procedures to follow. I have tried out Call Transfer as Transferee and it works fine. Thanks in Advance... Best Regards, Irshad.

1 0

localized dialplans
by trixter aka Bret McDanel 18 Dec '07

18 Dec '07

I wrote a generic dialplan that can be localized via user variables and would like people to customize it by creating variable definitions for their respective countries/regions. r6852 contains SRC/scripts/contrib/trixter/dialplan dialplan.nanpa.xml dialplan.nl.xml localized.dialplan.xml localized.dialplan.xml is a dialplan include dialplan.*.xml is a localized USER DIRECTORY variable definition To use, include localized.dialplan.xml into your dialplan. For each user (or global if you want) include the dialplan.*.xml appropriate for your locality into the user variable section. This should give you the ability to call international, domestic out of area, local - same area, and emergency. In the case on NANPA I have made the emergency regexp a little looser so that all X11 services are direct route to the provider (eg 411, 611, 911, etc). Most of the EU/EEA countries only need to adjust the country code variable and the local area (do not prefix 0 eg amsterdam NL is 20 not 020, London england is 20[7|8] not 020[7|8]). This is so that ENUM routing will work properly. If anyone finds a situation where this does not work properly I would appreciate it if they would let me know so that I can try to make something better that does work universally everywhere, however I think it should generally work most places. I see this more for multi-national providers to allow their customers to dial in a way they are more familiar with, giving the users easy choice in how their profile is built, as well as end users dealing with providers that generally have a single fixed dialplan but as a user you want to be able to dial like you would with the PSTN. It would be a nice touch for a provider to allow their users to dial as though they were on the pstn without requiring the UA to do a lot of the work, especially when the provider doesnt exactly control the UA. This is a feature I see missing from a great many providers on multiple continents. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you!

3 3

Outbound socket events in XML
by Ivan C Myrvold 17 Dec '07

17 Dec '07

Is it possible to get the events on outbound socket in XML now? When I use outbound socket, I get something like this: Application = bridge; "Application-Data" = "sofia/imyrvold.dyndns.org/110%25192.168.207.203"; "Caller-Caller-ID-Name" = "Ivan%20C%20Myrvold"; "Caller-Caller-ID-Number" = 100; "Caller-Channel-Answered-Time" = 0; "Caller-Channel-Created-Time" = 1197923559764562; .... .... ....

1 0

Call for Help
by Brian West 15 Dec '07

15 Dec '07

I need someone to do some work on this one: At the cli type help... verify each item listed has the proper decription and correct args. For the most part they should be correct on args but the description might not be enough for someone to know what it does. Anyone up for this task? I'm still testing and tweaking the default configuration and have a few more things to add to it. /b

1 0