Id be glad to be that test lab if that is the meaning of your message.
About ring_on_ms, changin this value may not be a good fix. I try to solve my problems in an empiric way. I found out that setting ring_time to -1 could solve my intermittent ringing problem, but that may cause problems with CLIP then (I took time to find the right document where CLIP FSK is defined). I dont know how it is implemented in openzap (which TAS is used ). Id be really happy to help, but I will need some pointers to start. Thanks François De : freeswitch-dev-boun...@lists.freeswitch.org [mailto:freeswitch-dev-boun...@lists.freeswitch.org] De la part de Anthony Minessale Envoyé : lundi 18 janvier 2010 23:24 À : freeswitch-dev@lists.freeswitch.org Objet : Re: [Freeswitch-dev] openzap information + mod_say_fr contribution Hi, Thanks for the assistance. jira.freeswitch.org is where you submit code ring_on_ms is currently hard coded in ozmod_wanpipe.c:1069 We anticipate problems with CLIP in non-US lines since I do not have any to work with. We can work together with Sangoma to get this working right in the future once we have test lab to reproduce it. On Mon, Jan 18, 2010 at 4:01 PM, <de...@thom.fr.eu.org> wrote: Hello, First let me congratulate for this great software. I was trying to diagnose a problem with my analog sangoma card, and FXS port giving intermitent ringing and not sending CLIP. I could see the intermitent ringing comes from this line. zchan->ring_time = zap_current_time_in_ms() + wp_globals.ring_on_ms; Setting zchan->ring_time could solve my problem, so could anybody indicate how to configure ring_on_ms (preferably in an XML configuration) The second point about CLIP is more difficult. First, let me indicate that Im residing in France and therefore using carrier and phones complying to european standard. About CLIP, I could see the V23 standard modulating frequencies are being populated (in fsk_modem_definitions in fsk.c). However, in zap_channel_send_fsk_data in zap_io.c, I could see that the BELL standard is hardcoded. Replacing the modem type with FSK_V23_FORWARD_MODE2 (which by what I could read seem to apply to me), I could see no difference. As I could not find as much relevant information as I would like in V23 (apart from the frequencies for the FSK), I was wondering what were I could find some information that could help to fix this. I dont know exactly what carrier_bits_start, carrier_bits_stop and chan_sieze_bits refer to, but I could find in that document (http://www.cs-strumentazione.com/manuali_pdf/clip.pdf) that « The Channel Seizure is present only in FSK-V23 Protocol. It consists in a series of 90 to 300 alternate SPACE/MARK bits » Considering this, I guess that I should use the else branch in the test « zchan->token_count > 1 ». What does this token_count refer to and where should I set it. Im still trying to figure this out (and will of course contribute the result of all this), so any help appreciated. I also did a contribution to mod_say_fr (to take into account the specific french idioms in saying date and time). Could you please tell me how to contribute this ? Thanks in advance François _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev 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 <mailto:msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com <mailto:paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <mailto:sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> pstn:+19193869900
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