Re: [Freeswitch-users] CLIP on FXS channels with mod_openzap
Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? François On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale@hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [12] iax:guest@conference.freeswitch.org/888 [13] googletalk:conf+888@conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel@thom.fr.eu.org [2] mailto:FreeSWITCH-users@lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale@hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale@gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888@conference.freeswitch.org [13] http://iax:guest@conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org
upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, François Legal <devel@thom.fr.eu.org> wrote:
Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org
I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ?
François
On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote:
can you test svn trunk or latest pre release of 1.0.5
On Mon, Nov 30, 2009 at 9:36 AM, François Legal <devel@thom.fr.eu.org>wrote:
Hello,
I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports.
When I ring on FXS ports, the connected phone does not display callerid/callerid-name.
I tried turning the stuff of in openzap.conf.xml () but it did not help.
As a side note, turning this on on the FXO ports drops the callerid information on incoming calls.
Running freeswitch 1.0.4 on linux 2.6.27.
François
_______________________________________________ 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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. François On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, François Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [2] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? François On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [4] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [5] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [6] http://www.freeswitch.org [7] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [8] ClueCon http://www.cluecon.com/ [9] Twitter: http://twitter.com/FreeSWITCH_wire [10] AIM: anthm MSN:anthony_minessale@hotmail.com [11] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [12] IRC: irc.freenode.net [13] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [14] iax:guest@conference.freeswitch.org/888 [15] googletalk:conf+888@conference.freeswitch.org [16] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [17] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [19] http://www.freeswitch.org [20] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [21] ClueCon http://www.cluecon.com/ [22] Twitter: http://twitter.com/FreeSWITCH_wire [23] AIM: anthm MSN:anthony_minessale@hotmail.com [24] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [25] IRC: irc.freenode.net [26] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [27] iax:guest@conference.freeswitch.org/888 [28] googletalk:conf+888@conference.freeswitch.org [29] pstn:213-799-1400 Links: ------ [1] mailto:devel@thom.fr.eu.org [2] http://files.freeswitch.org [3] mailto:devel@thom.fr.eu.org [4] mailto:FreeSWITCH-users@lists.freeswitch.org [5] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6] http://lists.freeswitch.org/mailman/options/freeswitch-users [7] http://www.freeswitch.org [8] http://www.freeswitch.org/ [9] http://www.cluecon.com/ [10] http://twitter.com/FreeSWITCH_wire [11] mailto:MSN%3Aanthony_minessale@hotmail.com [12] mailto:PAYPAL%3Aanthony.minessale@gmail.com [13] http://irc.freenode.net [14] mailto:sip%3A888@conference.freeswitch.org [15] http://iax:guest@conference.freeswitch.org/888 [16] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org [17] mailto:FreeSWITCH-users@lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] http://www.freeswitch.org/ [22] http://www.cluecon.com/ [23] http://twitter.com/FreeSWITCH_wire [24] mailto:MSN%3Aanthony_minessale@hotmail.com [25] mailto:PAYPAL%3Aanthony.minessale@gmail.com [26] http://irc.freenode.net [27] mailto:sip%3A888@conference.freeswitch.org [28] http://iax:guest@conference.freeswitch.org/888 [29] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org
Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, François Legal <devel@thom.fr.eu.org> wrote:
So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too.
On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working.
However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release.
François
On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale < anthony.minessale@gmail.com> wrote:
upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot.
On Tue, Dec 1, 2009 at 2:37 AM, François Legal <devel@thom.fr.eu.org>wrote:
Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org
I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ?
François
On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote:
can you test svn trunk or latest pre release of 1.0.5
On Mon, Nov 30, 2009 at 9:36 AM, François Legal <devel@thom.fr.eu.org>wrote:
Hello,
I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports.
When I ring on FXS ports, the connected phone does not display callerid/callerid-name.
I tried turning the stuff of in openzap.conf.xml () but it did not help.
As a side note, turning this on on the FXO ports drops the callerid information on incoming calls.
Running freeswitch 1.0.4 on linux 2.6.27.
François
_______________________________________________ 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_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I'm already using the latest wanpipe drivers (3.5.8), so yes. François On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote: Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, François Legal wrote: So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. François On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, François Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [4] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? François On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. François _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [8] http://www.freeswitch.org [9] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [10] ClueCon http://www.cluecon.com/ [11] Twitter: http://twitter.com/FreeSWITCH_wire [12] AIM: anthm MSN:anthony_minessale@hotmail.com [13] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [14] IRC: irc.freenode.net [15] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [16] iax:guest@conference.freeswitch.org/888 [17] googletalk:conf+888@conference.freeswitch.org [18] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [23] ClueCon http://www.cluecon.com/ [24] Twitter: http://twitter.com/FreeSWITCH_wire [25] AIM: anthm MSN:anthony_minessale@hotmail.com [26] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [27] IRC: irc.freenode.net [28] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [29] iax:guest@conference.freeswitch.org/888 [30] googletalk:conf+888@conference.freeswitch.org [31] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org [32] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [33] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [34] http://www.freeswitch.org [35] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [36] ClueCon http://www.cluecon.com/ [37] Twitter: http://twitter.com/FreeSWITCH_wire [38] AIM: anthm MSN:anthony_minessale@hotmail.com [39] GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com [40] IRC: irc.freenode.net [41] #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org [42] iax:guest@conference.freeswitch.org/888 [43] googletalk:conf+888@conference.freeswitch.org [44] pstn:213-799-1400 Links: ------ [1] mailto:devel@thom.fr.eu.org [2] mailto:anthony.minessale@gmail.com [3] mailto:devel@thom.fr.eu.org [4] http://files.freeswitch.org [5] mailto:devel@thom.fr.eu.org [6] mailto:FreeSWITCH-users@lists.freeswitch.org [7] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] http://lists.freeswitch.org/mailman/options/freeswitch-users [9] http://www.freeswitch.org [10] http://www.freeswitch.org/ [11] http://www.cluecon.com/ [12] http://twitter.com/FreeSWITCH_wire [13] mailto:MSN%3Aanthony_minessale@hotmail.com [14] mailto:PAYPAL%3Aanthony.minessale@gmail.com [15] http://irc.freenode.net [16] mailto:sip%3A888@conference.freeswitch.org [17] http://iax:guest@conference.freeswitch.org/888 [18] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org [19] mailto:FreeSWITCH-users@lists.freeswitch.org [20] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [21] http://lists.freeswitch.org/mailman/options/freeswitch-users [22] http://www.freeswitch.org [23] http://www.freeswitch.org/ [24] http://www.cluecon.com/ [25] http://twitter.com/FreeSWITCH_wire [26] mailto:MSN%3Aanthony_minessale@hotmail.com [27] mailto:PAYPAL%3Aanthony.minessale@gmail.com [28] http://irc.freenode.net [29] mailto:sip%3A888@conference.freeswitch.org [30] http://iax:guest@conference.freeswitch.org/888 [31] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org [32] mailto:FreeSWITCH-users@lists.freeswitch.org [33] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [34] http://lists.freeswitch.org/mailman/options/freeswitch-users [35] http://www.freeswitch.org [36] http://www.freeswitch.org/ [37] http://www.cluecon.com/ [38] http://twitter.com/FreeSWITCH_wire [39] mailto:MSN%3Aanthony_minessale@hotmail.com [40] mailto:PAYPAL%3Aanthony.minessale@gmail.com [41] http://irc.freenode.net [42] mailto:sip%3A888@conference.freeswitch.org [43] http://iax:guest@conference.freeswitch.org/888 [44] mailto:googletalk%3Aconf%2B888@conference.freeswitch.org
I'm still working on this issue, and decided to take a look at the openzap code. First, I figured out that the parameter name for callerid is enable_callerid rather than enable-callerid. I also figured out that this parameter defaults to TRUE (which is coherent with the observed behaviour on my FXO span) By further checking the code, I figured out that presenting the callerid on an FXS port might not be implemented yet. I could see the code for retrieving the callerid from FXO but nothing to send it. Is my asumption (feature not implemented) correct ? François On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote: Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, François Legal wrote: So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. François On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, François Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [4] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? François On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, François Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. 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I recall implementing that back when we released openzap, it should be in there unless someone chopped it out for some reason. Look for "zap_channel_send_fsk_data" Mike On Dec 9, 2009, at 6:01 AM, François Legal wrote:
I'm still working on this issue, and decided to take a look at the openzap code.
First, I figured out that the parameter name for callerid is enable_callerid rather than enable-callerid.
I also figured out that this parameter defaults to TRUE (which is coherent with the observed behaviour on my FXO span)
By further checking the code, I figured out that presenting the callerid on an FXS port might not be implemented yet. I could see the code for retrieving the callerid from FXO but nothing to send it.
Is my asumption (feature not implemented) correct ?
François
participants (3)
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Anthony Minessale
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François Legal
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Michael Jerris