From m.sobkow at marketelsystems.com Wed Apr 6 20:41:08 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 06 Apr 2011 10:41:08 -0600 Subject: [Freeswitch-dev] Automatic hangups? Message-ID: <4D9C97A4.3060508@marketelsystems.com> Has someone over the past few months added some sort of automatic hangup feature for parked calls? Our application lets an operator log in, and parks their call until there is a customer available for them to bridge to. Previously this worked just fine -- operators could sit for minutes or hours (much to the annoyance of managers), waiting for a call. However, something has changed. Freeswitch now seems to automatically unpark the call and hang it up, even though it hasn't been told to do _anything_ by the application. -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From anthony.minessale at gmail.com Wed Apr 6 21:15:57 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 6 Apr 2011 12:15:57 -0500 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: <4D9C97A4.3060508@marketelsystems.com> References: <4D9C97A4.3060508@marketelsystems.com> Message-ID: no you are probably getting a session timer timeout. On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow wrote: > Has someone over the past few months added some sort of automatic hangup > feature for parked calls? > > Our application lets an operator log in, and parks their call until > there is a customer available for them to bridge to. ?Previously this > worked just fine -- operators could sit for minutes or hours (much to > the annoyance of managers), waiting for a call. > > However, something has changed. ?Freeswitch now seems to automatically > unpark the call and hang it up, even though it hasn't been told to do > _anything_ by the application. > > -- > Mark Sobkow > Senior Developer > MarkeTel Multi-Line Dialing Systems LTD. > 428 Victoria Ave > Regina, SK S4N-0P6 > Toll-Free: 800-289-8616-X533 > Local: 306-359-6893-X533 > Fax: 306-359-6879 > Email: m.sobkow at marketelsystems.com > Web: http://www.marketelsystems.com > > Visit our Blog for industry related information. > http://marketel-systems.blogspot.com/ > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From m.sobkow at marketelsystems.com Wed Apr 6 22:11:54 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 06 Apr 2011 12:11:54 -0600 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: References: <4D9C97A4.3060508@marketelsystems.com> Message-ID: <4D9CACEA.1020107@marketelsystems.com> I've never even heard of the session timers, much less programmed any into the software. What should I look for? I'm suspecting there may be a default timer that needs to be reset to disabled/no-timeout to get the behaviour we used to have. On 06/04/2011 11:15 AM, Anthony Minessale wrote: > no you are probably getting a session timer timeout. > > > On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow > wrote: >> Has someone over the past few months added some sort of automatic hangup >> feature for parked calls? >> >> Our application lets an operator log in, and parks their call until >> there is a customer available for them to bridge to. Previously this >> worked just fine -- operators could sit for minutes or hours (much to >> the annoyance of managers), waiting for a call. >> >> However, something has changed. Freeswitch now seems to automatically >> unpark the call and hang it up, even though it hasn't been told to do >> _anything_ by the application. >> >> -- >> Mark Sobkow >> Senior Developer >> MarkeTel Multi-Line Dialing Systems LTD. >> 428 Victoria Ave >> Regina, SK S4N-0P6 >> Toll-Free: 800-289-8616-X533 >> Local: 306-359-6893-X533 >> Fax: 306-359-6879 >> Email: m.sobkow at marketelsystems.com >> Web: http://www.marketelsystems.com >> >> Visit our Blog for industry related information. >> http://marketel-systems.blogspot.com/ >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From anthony.minessale at gmail.com Wed Apr 6 22:28:52 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 6 Apr 2011 13:28:52 -0500 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: <4D9CACEA.1020107@marketelsystems.com> References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> Message-ID: issue these 2 commands from cli console loglevel debug (/log debug if in fs_cli) sofia global siptrace on reproduce look at the logs or post them to pastebin and let us have a look. On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow wrote: > I've never even heard of the session timers, much less programmed any > into the software. ?What should I look for? ?I'm suspecting there may be > a default timer that needs to be reset to disabled/no-timeout to get the > behaviour we used to have. > > On 06/04/2011 11:15 AM, Anthony Minessale wrote: >> no you are probably getting a session timer timeout. >> >> >> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >> ?wrote: >>> Has someone over the past few months added some sort of automatic hangup >>> feature for parked calls? >>> >>> Our application lets an operator log in, and parks their call until >>> there is a customer available for them to bridge to. ?Previously this >>> worked just fine -- operators could sit for minutes or hours (much to >>> the annoyance of managers), waiting for a call. >>> >>> However, something has changed. ?Freeswitch now seems to automatically >>> unpark the call and hang it up, even though it hasn't been told to do >>> _anything_ by the application. >>> >>> -- >>> Mark Sobkow >>> Senior Developer >>> MarkeTel Multi-Line Dialing Systems LTD. >>> 428 Victoria Ave >>> Regina, SK S4N-0P6 >>> Toll-Free: 800-289-8616-X533 >>> Local: 306-359-6893-X533 >>> Fax: 306-359-6879 >>> Email: m.sobkow at marketelsystems.com >>> Web: http://www.marketelsystems.com >>> >>> Visit our Blog for industry related information. >>> http://marketel-systems.blogspot.com/ >>> >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >> >> > > > -- > Mark Sobkow > Senior Developer > MarkeTel Multi-Line Dialing Systems LTD. > 428 Victoria Ave > Regina, SK S4N-0P6 > Toll-Free: 800-289-8616-X533 > Local: 306-359-6893-X533 > Fax: 306-359-6879 > Email: m.sobkow at marketelsystems.com > Web: http://www.marketelsystems.com > > Visit our Blog for industry related information. > http://marketel-systems.blogspot.com/ > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From m.sobkow at marketelsystems.com Thu Apr 7 00:43:14 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 06 Apr 2011 14:43:14 -0600 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> Message-ID: <4D9CD062.6010208@marketelsystems.com> Trace posted to pastebin under "Mark Sobkow". Wiki mentions a pastebin number, but I don't see one to include in this message. On 06/04/2011 12:28 PM, Anthony Minessale wrote: > issue these 2 commands from cli > console loglevel debug (/log debug if in fs_cli) > sofia global siptrace on > > reproduce > look at the logs or post them to pastebin and let us have a look. > > > > > > On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow > wrote: >> I've never even heard of the session timers, much less programmed any >> into the software. What should I look for? I'm suspecting there may be >> a default timer that needs to be reset to disabled/no-timeout to get the >> behaviour we used to have. >> >> On 06/04/2011 11:15 AM, Anthony Minessale wrote: >>> no you are probably getting a session timer timeout. >>> >>> >>> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >>> wrote: >>>> Has someone over the past few months added some sort of automatic hangup >>>> feature for parked calls? >>>> >>>> Our application lets an operator log in, and parks their call until >>>> there is a customer available for them to bridge to. Previously this >>>> worked just fine -- operators could sit for minutes or hours (much to >>>> the annoyance of managers), waiting for a call. >>>> >>>> However, something has changed. Freeswitch now seems to automatically >>>> unpark the call and hang it up, even though it hasn't been told to do >>>> _anything_ by the application. >>>> >>>> -- >>>> Mark Sobkow >>>> Senior Developer >>>> MarkeTel Multi-Line Dialing Systems LTD. >>>> 428 Victoria Ave >>>> Regina, SK S4N-0P6 >>>> Toll-Free: 800-289-8616-X533 >>>> Local: 306-359-6893-X533 >>>> Fax: 306-359-6879 >>>> Email: m.sobkow at marketelsystems.com >>>> Web: http://www.marketelsystems.com >>>> >>>> Visit our Blog for industry related information. >>>> http://marketel-systems.blogspot.com/ >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>> >> >> -- >> Mark Sobkow >> Senior Developer >> MarkeTel Multi-Line Dialing Systems LTD. >> 428 Victoria Ave >> Regina, SK S4N-0P6 >> Toll-Free: 800-289-8616-X533 >> Local: 306-359-6893-X533 >> Fax: 306-359-6879 >> Email: m.sobkow at marketelsystems.com >> Web: http://www.marketelsystems.com >> >> Visit our Blog for industry related information. >> http://marketel-systems.blogspot.com/ >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From anthony.minessale at gmail.com Thu Apr 7 01:01:34 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 6 Apr 2011 16:01:34 -0500 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: <4D9CD062.6010208@marketelsystems.com> References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> <4D9CD062.6010208@marketelsystems.com> Message-ID: Reading your paste: Line 284 getting an invite from user-agent: "SIPPER for phoner" Line 580 Sending 200 OK Numerous packets after that re-sending 200 OK and never getting an ACK. Line 958 Gives up and sends a BYE Either you have a network issue that impedes your client from receiving the 200OK or sending an ACK in return or your client placing the call is deciding not to send an ACK intentionally for some reason. On Wed, Apr 6, 2011 at 3:43 PM, Mark Sobkow wrote: > Trace posted to pastebin under "Mark Sobkow". ?Wiki mentions a pastebin > number, but I don't see one to include in this message. > > On 06/04/2011 12:28 PM, Anthony Minessale wrote: >> issue these 2 commands from cli >> console loglevel debug (/log debug if in fs_cli) >> sofia global siptrace on >> >> reproduce >> look at the logs or post them to pastebin and let us have a look. >> >> >> >> >> >> On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow >> ?wrote: >>> I've never even heard of the session timers, much less programmed any >>> into the software. ?What should I look for? ?I'm suspecting there may be >>> a default timer that needs to be reset to disabled/no-timeout to get the >>> behaviour we used to have. >>> >>> On 06/04/2011 11:15 AM, Anthony Minessale wrote: >>>> no you are probably getting a session timer timeout. >>>> >>>> >>>> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >>>> ? ?wrote: >>>>> Has someone over the past few months added some sort of automatic hangup >>>>> feature for parked calls? >>>>> >>>>> Our application lets an operator log in, and parks their call until >>>>> there is a customer available for them to bridge to. ?Previously this >>>>> worked just fine -- operators could sit for minutes or hours (much to >>>>> the annoyance of managers), waiting for a call. >>>>> >>>>> However, something has changed. ?Freeswitch now seems to automatically >>>>> unpark the call and hang it up, even though it hasn't been told to do >>>>> _anything_ by the application. >>>>> >>>>> -- >>>>> Mark Sobkow >>>>> Senior Developer >>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>> 428 Victoria Ave >>>>> Regina, SK S4N-0P6 >>>>> Toll-Free: 800-289-8616-X533 >>>>> Local: 306-359-6893-X533 >>>>> Fax: 306-359-6879 >>>>> Email: m.sobkow at marketelsystems.com >>>>> Web: http://www.marketelsystems.com >>>>> >>>>> Visit our Blog for industry related information. >>>>> http://marketel-systems.blogspot.com/ >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-dev mailing list >>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>> http://www.freeswitch.org >>>>> >>>> >>> >>> -- >>> Mark Sobkow >>> Senior Developer >>> MarkeTel Multi-Line Dialing Systems LTD. >>> 428 Victoria Ave >>> Regina, SK S4N-0P6 >>> Toll-Free: 800-289-8616-X533 >>> Local: 306-359-6893-X533 >>> Fax: 306-359-6879 >>> Email: m.sobkow at marketelsystems.com >>> Web: http://www.marketelsystems.com >>> >>> Visit our Blog for industry related information. >>> http://marketel-systems.blogspot.com/ >>> >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >> >> > > > -- > Mark Sobkow > Senior Developer > MarkeTel Multi-Line Dialing Systems LTD. > 428 Victoria Ave > Regina, SK S4N-0P6 > Toll-Free: 800-289-8616-X533 > Local: 306-359-6893-X533 > Fax: 306-359-6879 > Email: m.sobkow at marketelsystems.com > Web: http://www.marketelsystems.com > > Visit our Blog for industry related information. > http://marketel-systems.blogspot.com/ > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From m.sobkow at marketelsystems.com Thu Apr 7 01:15:58 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 06 Apr 2011 15:15:58 -0600 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> <4D9CD062.6010208@marketelsystems.com> Message-ID: <4D9CD80E.80100@marketelsystems.com> We've tried several different softphones: Phoner (SIPPER stack), X-Lite, and our internal M-Phone (I forget what SIP stack implementation that uses, as I didn't write it.) All exhibit the same problem. Our network admin insists it's not a network problem. As we hadn't had this problem previously, I can't argue with him over it. I'm a little unclear about what is supposed to be sending the ACK -- the softphone, FreeSwitch, or our application code. On 06/04/2011 3:01 PM, Anthony Minessale wrote: > Reading your paste: > > Line 284 getting an invite from user-agent: "SIPPER for phoner" > Line 580 Sending 200 OK > Numerous packets after that re-sending 200 OK and never getting an ACK. > Line 958 Gives up and sends a BYE > > Either you have a network issue that impedes your client from > receiving the 200OK or sending an ACK in return or your client placing > the call is deciding not to send an ACK intentionally for some reason. > > > > > On Wed, Apr 6, 2011 at 3:43 PM, Mark Sobkow > wrote: >> Trace posted to pastebin under "Mark Sobkow". Wiki mentions a pastebin >> number, but I don't see one to include in this message. >> >> On 06/04/2011 12:28 PM, Anthony Minessale wrote: >>> issue these 2 commands from cli >>> console loglevel debug (/log debug if in fs_cli) >>> sofia global siptrace on >>> >>> reproduce >>> look at the logs or post them to pastebin and let us have a look. >>> >>> >>> >>> >>> >>> On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow >>> wrote: >>>> I've never even heard of the session timers, much less programmed any >>>> into the software. What should I look for? I'm suspecting there may be >>>> a default timer that needs to be reset to disabled/no-timeout to get the >>>> behaviour we used to have. >>>> >>>> On 06/04/2011 11:15 AM, Anthony Minessale wrote: >>>>> no you are probably getting a session timer timeout. >>>>> >>>>> >>>>> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >>>>> wrote: >>>>>> Has someone over the past few months added some sort of automatic hangup >>>>>> feature for parked calls? >>>>>> >>>>>> Our application lets an operator log in, and parks their call until >>>>>> there is a customer available for them to bridge to. Previously this >>>>>> worked just fine -- operators could sit for minutes or hours (much to >>>>>> the annoyance of managers), waiting for a call. >>>>>> >>>>>> However, something has changed. Freeswitch now seems to automatically >>>>>> unpark the call and hang it up, even though it hasn't been told to do >>>>>> _anything_ by the application. >>>>>> >>>>>> -- >>>>>> Mark Sobkow >>>>>> Senior Developer >>>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>>> 428 Victoria Ave >>>>>> Regina, SK S4N-0P6 >>>>>> Toll-Free: 800-289-8616-X533 >>>>>> Local: 306-359-6893-X533 >>>>>> Fax: 306-359-6879 >>>>>> Email: m.sobkow at marketelsystems.com >>>>>> Web: http://www.marketelsystems.com >>>>>> >>>>>> Visit our Blog for industry related information. >>>>>> http://marketel-systems.blogspot.com/ >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-dev mailing list >>>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>> http://www.freeswitch.org >>>>>> >>>> -- >>>> Mark Sobkow >>>> Senior Developer >>>> MarkeTel Multi-Line Dialing Systems LTD. >>>> 428 Victoria Ave >>>> Regina, SK S4N-0P6 >>>> Toll-Free: 800-289-8616-X533 >>>> Local: 306-359-6893-X533 >>>> Fax: 306-359-6879 >>>> Email: m.sobkow at marketelsystems.com >>>> Web: http://www.marketelsystems.com >>>> >>>> Visit our Blog for industry related information. >>>> http://marketel-systems.blogspot.com/ >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>> >> >> -- >> Mark Sobkow >> Senior Developer >> MarkeTel Multi-Line Dialing Systems LTD. >> 428 Victoria Ave >> Regina, SK S4N-0P6 >> Toll-Free: 800-289-8616-X533 >> Local: 306-359-6893-X533 >> Fax: 306-359-6879 >> Email: m.sobkow at marketelsystems.com >> Web: http://www.marketelsystems.com >> >> Visit our Blog for industry related information. >> http://marketel-systems.blogspot.com/ >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From msc at freeswitch.org Thu Apr 7 01:28:43 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Apr 2011 14:28:43 -0700 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: <4D9CD80E.80100@marketelsystems.com> References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> <4D9CD062.6010208@marketelsystems.com> <4D9CD80E.80100@marketelsystems.com> Message-ID: On Wed, Apr 6, 2011 at 2:15 PM, Mark Sobkow wrote: > We've tried several different softphones: Phoner (SIPPER stack), X-Lite, > and our internal M-Phone (I forget what SIP stack implementation that > uses, as I didn't write it.) All exhibit the same problem. > > Our network admin insists it's not a network problem. As we hadn't had > this problem previously, I can't argue with him over it. > > I'm a little unclear about what is supposed to be sending the ACK -- the > softphone, FreeSwitch, or our application code. > This is typical SIP traffic, so the UA sending the INVITE should be ACKing the appropriate responses from FS. In your case, FreeSWITCH sends a 200 OK several times, indicating that the phone has answered the INVITE. However, FS is not receiving the required ACK from the UA so it gives up and sends a BYE. So, what Tony said is correct - for *some* reason the ACK is not making it back to FreeSWITCH. This is *not* a problem with FreeSWITCH, it's a problem on the network or with the originating UA. Are you able to confirm that the UA is seeing the 200 OK from FreeSWITCH? That sounds like a reasonable first place to troubleshoot. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110406/2af01256/attachment.html From anthony.minessale at gmail.com Thu Apr 7 01:30:25 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 6 Apr 2011 16:30:25 -0500 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: <4D9CD80E.80100@marketelsystems.com> References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> <4D9CD062.6010208@marketelsystems.com> <4D9CD80E.80100@marketelsystems.com> Message-ID: I recommend some research on your part on how SIP works but the short version is: CLIENT sends FS INVITE FS sends CLIENT 200 OK CLIENT sends FS ACK In your case FS sends CLIENT like 15 or so 200 OK and never gets any ACK so it gives up pursuant to the specification. Maybe you should get a pcap from both the FS box and the box where your client is running. Chances are you have iptables or some other firewall or invisible SIP ALG eating the packets. As for FreeSWITCH, I can't tell what version you are running because its not properly present in the User Agent. On Wed, Apr 6, 2011 at 4:15 PM, Mark Sobkow wrote: > We've tried several different softphones: Phoner (SIPPER stack), X-Lite, > and our internal M-Phone (I forget what SIP stack implementation that > uses, as I didn't write it.) ?All exhibit the same problem. > > Our network admin insists it's not a network problem. ?As we hadn't had > this problem previously, I can't argue with him over it. > > I'm a little unclear about what is supposed to be sending the ACK -- the > softphone, FreeSwitch, or our application code. > > On 06/04/2011 3:01 PM, Anthony Minessale wrote: >> Reading your paste: >> >> Line 284 getting an invite from user-agent: "SIPPER for phoner" >> Line 580 Sending 200 OK >> Numerous packets after that re-sending 200 OK and never getting an ACK. >> Line 958 Gives up and sends a BYE >> >> Either you have a network issue that impedes your client from >> receiving the 200OK or sending an ACK in return or your client placing >> the call is deciding not to send an ACK intentionally for some reason. >> >> >> >> >> On Wed, Apr 6, 2011 at 3:43 PM, Mark Sobkow >> ?wrote: >>> Trace posted to pastebin under "Mark Sobkow". ?Wiki mentions a pastebin >>> number, but I don't see one to include in this message. >>> >>> On 06/04/2011 12:28 PM, Anthony Minessale wrote: >>>> issue these 2 commands from cli >>>> console loglevel debug (/log debug if in fs_cli) >>>> sofia global siptrace on >>>> >>>> reproduce >>>> look at the logs or post them to pastebin and let us have a look. >>>> >>>> >>>> >>>> >>>> >>>> On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow >>>> ? ?wrote: >>>>> I've never even heard of the session timers, much less programmed any >>>>> into the software. ?What should I look for? ?I'm suspecting there may be >>>>> a default timer that needs to be reset to disabled/no-timeout to get the >>>>> behaviour we used to have. >>>>> >>>>> On 06/04/2011 11:15 AM, Anthony Minessale wrote: >>>>>> no you are probably getting a session timer timeout. >>>>>> >>>>>> >>>>>> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >>>>>> ? ? ?wrote: >>>>>>> Has someone over the past few months added some sort of automatic hangup >>>>>>> feature for parked calls? >>>>>>> >>>>>>> Our application lets an operator log in, and parks their call until >>>>>>> there is a customer available for them to bridge to. ?Previously this >>>>>>> worked just fine -- operators could sit for minutes or hours (much to >>>>>>> the annoyance of managers), waiting for a call. >>>>>>> >>>>>>> However, something has changed. ?Freeswitch now seems to automatically >>>>>>> unpark the call and hang it up, even though it hasn't been told to do >>>>>>> _anything_ by the application. >>>>>>> >>>>>>> -- >>>>>>> Mark Sobkow >>>>>>> Senior Developer >>>>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>>>> 428 Victoria Ave >>>>>>> Regina, SK S4N-0P6 >>>>>>> Toll-Free: 800-289-8616-X533 >>>>>>> Local: 306-359-6893-X533 >>>>>>> Fax: 306-359-6879 >>>>>>> Email: m.sobkow at marketelsystems.com >>>>>>> Web: http://www.marketelsystems.com >>>>>>> >>>>>>> Visit our Blog for industry related information. >>>>>>> http://marketel-systems.blogspot.com/ >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-dev mailing list >>>>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>>> http://www.freeswitch.org >>>>>>> >>>>> -- >>>>> Mark Sobkow >>>>> Senior Developer >>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>> 428 Victoria Ave >>>>> Regina, SK S4N-0P6 >>>>> Toll-Free: 800-289-8616-X533 >>>>> Local: 306-359-6893-X533 >>>>> Fax: 306-359-6879 >>>>> Email: m.sobkow at marketelsystems.com >>>>> Web: http://www.marketelsystems.com >>>>> >>>>> Visit our Blog for industry related information. >>>>> http://marketel-systems.blogspot.com/ >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-dev mailing list >>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>> http://www.freeswitch.org >>>>> >>>> >>> >>> -- >>> Mark Sobkow >>> Senior Developer >>> MarkeTel Multi-Line Dialing Systems LTD. >>> 428 Victoria Ave >>> Regina, SK S4N-0P6 >>> Toll-Free: 800-289-8616-X533 >>> Local: 306-359-6893-X533 >>> Fax: 306-359-6879 >>> Email: m.sobkow at marketelsystems.com >>> Web: http://www.marketelsystems.com >>> >>> Visit our Blog for industry related information. >>> http://marketel-systems.blogspot.com/ >>> >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >> >> > > > -- > Mark Sobkow > Senior Developer > MarkeTel Multi-Line Dialing Systems LTD. > 428 Victoria Ave > Regina, SK S4N-0P6 > Toll-Free: 800-289-8616-X533 > Local: 306-359-6893-X533 > Fax: 306-359-6879 > Email: m.sobkow at marketelsystems.com > Web: http://www.marketelsystems.com > > Visit our Blog for industry related information. > http://marketel-systems.blogspot.com/ > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From m.sobkow at marketelsystems.com Thu Apr 7 02:42:09 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 06 Apr 2011 16:42:09 -0600 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> <4D9CD062.6010208@marketelsystems.com> <4D9CD80E.80100@marketelsystems.com> Message-ID: <4D9CEC41.5060609@marketelsystems.com> Found it. The "Gateways" configuration was wrong. It was referencing the IP address of another server (TestSrv), so the ACK packets were getting sent to the wrong machine. Very subtle bug -- other than the hangups things have been working ok, so I didn't suspect FS configuration to be the issue. Found it by using WireShark to trace the SIP packets. Thanks for the help, everyone. On 06/04/2011 3:30 PM, Anthony Minessale wrote: > I recommend some research on your part on how SIP works but the short > version is: > > CLIENT sends FS INVITE > FS sends CLIENT 200 OK > CLIENT sends FS ACK > > In your case FS sends CLIENT like 15 or so 200 OK and never gets any > ACK so it gives up pursuant to the specification. > > Maybe you should get a pcap from both the FS box and the box where > your client is running. Chances are you have iptables or some other > firewall or invisible SIP ALG eating the packets. > > As for FreeSWITCH, I can't tell what version you are running because > its not properly present in the User Agent. > > > > > On Wed, Apr 6, 2011 at 4:15 PM, Mark Sobkow > wrote: >> We've tried several different softphones: Phoner (SIPPER stack), X-Lite, >> and our internal M-Phone (I forget what SIP stack implementation that >> uses, as I didn't write it.) All exhibit the same problem. >> >> Our network admin insists it's not a network problem. As we hadn't had >> this problem previously, I can't argue with him over it. >> >> I'm a little unclear about what is supposed to be sending the ACK -- the >> softphone, FreeSwitch, or our application code. >> >> On 06/04/2011 3:01 PM, Anthony Minessale wrote: >>> Reading your paste: >>> >>> Line 284 getting an invite from user-agent: "SIPPER for phoner" >>> Line 580 Sending 200 OK >>> Numerous packets after that re-sending 200 OK and never getting an ACK. >>> Line 958 Gives up and sends a BYE >>> >>> Either you have a network issue that impedes your client from >>> receiving the 200OK or sending an ACK in return or your client placing >>> the call is deciding not to send an ACK intentionally for some reason. >>> >>> >>> >>> >>> On Wed, Apr 6, 2011 at 3:43 PM, Mark Sobkow >>> wrote: >>>> Trace posted to pastebin under "Mark Sobkow". Wiki mentions a pastebin >>>> number, but I don't see one to include in this message. >>>> >>>> On 06/04/2011 12:28 PM, Anthony Minessale wrote: >>>>> issue these 2 commands from cli >>>>> console loglevel debug (/log debug if in fs_cli) >>>>> sofia global siptrace on >>>>> >>>>> reproduce >>>>> look at the logs or post them to pastebin and let us have a look. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow >>>>> wrote: >>>>>> I've never even heard of the session timers, much less programmed any >>>>>> into the software. What should I look for? I'm suspecting there may be >>>>>> a default timer that needs to be reset to disabled/no-timeout to get the >>>>>> behaviour we used to have. >>>>>> >>>>>> On 06/04/2011 11:15 AM, Anthony Minessale wrote: >>>>>>> no you are probably getting a session timer timeout. >>>>>>> >>>>>>> >>>>>>> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >>>>>>> wrote: >>>>>>>> Has someone over the past few months added some sort of automatic hangup >>>>>>>> feature for parked calls? >>>>>>>> >>>>>>>> Our application lets an operator log in, and parks their call until >>>>>>>> there is a customer available for them to bridge to. Previously this >>>>>>>> worked just fine -- operators could sit for minutes or hours (much to >>>>>>>> the annoyance of managers), waiting for a call. >>>>>>>> >>>>>>>> However, something has changed. Freeswitch now seems to automatically >>>>>>>> unpark the call and hang it up, even though it hasn't been told to do >>>>>>>> _anything_ by the application. >>>>>>>> >>>>>>>> -- >>>>>>>> Mark Sobkow >>>>>>>> Senior Developer >>>>>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>>>>> 428 Victoria Ave >>>>>>>> Regina, SK S4N-0P6 >>>>>>>> Toll-Free: 800-289-8616-X533 >>>>>>>> Local: 306-359-6893-X533 >>>>>>>> Fax: 306-359-6879 >>>>>>>> Email: m.sobkow at marketelsystems.com >>>>>>>> Web: http://www.marketelsystems.com >>>>>>>> >>>>>>>> Visit our Blog for industry related information. >>>>>>>> http://marketel-systems.blogspot.com/ >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-dev mailing list >>>>>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>> -- >>>>>> Mark Sobkow >>>>>> Senior Developer >>>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>>> 428 Victoria Ave >>>>>> Regina, SK S4N-0P6 >>>>>> Toll-Free: 800-289-8616-X533 >>>>>> Local: 306-359-6893-X533 >>>>>> Fax: 306-359-6879 >>>>>> Email: m.sobkow at marketelsystems.com >>>>>> Web: http://www.marketelsystems.com >>>>>> >>>>>> Visit our Blog for industry related information. >>>>>> http://marketel-systems.blogspot.com/ >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-dev mailing list >>>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>> http://www.freeswitch.org >>>>>> >>>> -- >>>> Mark Sobkow >>>> Senior Developer >>>> MarkeTel Multi-Line Dialing Systems LTD. >>>> 428 Victoria Ave >>>> Regina, SK S4N-0P6 >>>> Toll-Free: 800-289-8616-X533 >>>> Local: 306-359-6893-X533 >>>> Fax: 306-359-6879 >>>> Email: m.sobkow at marketelsystems.com >>>> Web: http://www.marketelsystems.com >>>> >>>> Visit our Blog for industry related information. >>>> http://marketel-systems.blogspot.com/ >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>> >> >> -- >> Mark Sobkow >> Senior Developer >> MarkeTel Multi-Line Dialing Systems LTD. >> 428 Victoria Ave >> Regina, SK S4N-0P6 >> Toll-Free: 800-289-8616-X533 >> Local: 306-359-6893-X533 >> Fax: 306-359-6879 >> Email: m.sobkow at marketelsystems.com >> Web: http://www.marketelsystems.com >> >> Visit our Blog for industry related information. >> http://marketel-systems.blogspot.com/ >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From peter.olsson at visionutveckling.se Thu Apr 7 10:42:38 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 7 Apr 2011 08:42:38 +0200 Subject: [Freeswitch-dev] Automatic hangups? In-Reply-To: <4D9CEC41.5060609@marketelsystems.com> References: <4D9C97A4.3060508@marketelsystems.com> <4D9CACEA.1020107@marketelsystems.com> <4D9CD062.6010208@marketelsystems.com> <4D9CD80E.80100@marketelsystems.com> <4D9CEC41.5060609@marketelsystems.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4D30608@cooper> Could you explain this a bit more? I don't understand how a bad gateway configuration in you FS server would cause your client to send ACK's to the wrong IP? Or am I missing something here? /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Mark Sobkow Skickat: den 7 april 2011 00:42 Till: freeswitch-dev at lists.freeswitch.org ?mne: [SPAM] - Re: [Freeswitch-dev] Automatic hangups? Found it. The "Gateways" configuration was wrong. It was referencing the IP address of another server (TestSrv), so the ACK packets were getting sent to the wrong machine. Very subtle bug -- other than the hangups things have been working ok, so I didn't suspect FS configuration to be the issue. Found it by using WireShark to trace the SIP packets. Thanks for the help, everyone. On 06/04/2011 3:30 PM, Anthony Minessale wrote: > I recommend some research on your part on how SIP works but the short > version is: > > CLIENT sends FS INVITE > FS sends CLIENT 200 OK > CLIENT sends FS ACK > > In your case FS sends CLIENT like 15 or so 200 OK and never gets any > ACK so it gives up pursuant to the specification. > > Maybe you should get a pcap from both the FS box and the box where > your client is running. Chances are you have iptables or some other > firewall or invisible SIP ALG eating the packets. > > As for FreeSWITCH, I can't tell what version you are running because > its not properly present in the User Agent. > > > > > On Wed, Apr 6, 2011 at 4:15 PM, Mark Sobkow > wrote: >> We've tried several different softphones: Phoner (SIPPER stack), X-Lite, >> and our internal M-Phone (I forget what SIP stack implementation that >> uses, as I didn't write it.) All exhibit the same problem. >> >> Our network admin insists it's not a network problem. As we hadn't had >> this problem previously, I can't argue with him over it. >> >> I'm a little unclear about what is supposed to be sending the ACK -- the >> softphone, FreeSwitch, or our application code. >> >> On 06/04/2011 3:01 PM, Anthony Minessale wrote: >>> Reading your paste: >>> >>> Line 284 getting an invite from user-agent: "SIPPER for phoner" >>> Line 580 Sending 200 OK >>> Numerous packets after that re-sending 200 OK and never getting an ACK. >>> Line 958 Gives up and sends a BYE >>> >>> Either you have a network issue that impedes your client from >>> receiving the 200OK or sending an ACK in return or your client placing >>> the call is deciding not to send an ACK intentionally for some reason. >>> >>> >>> >>> >>> On Wed, Apr 6, 2011 at 3:43 PM, Mark Sobkow >>> wrote: >>>> Trace posted to pastebin under "Mark Sobkow". Wiki mentions a pastebin >>>> number, but I don't see one to include in this message. >>>> >>>> On 06/04/2011 12:28 PM, Anthony Minessale wrote: >>>>> issue these 2 commands from cli >>>>> console loglevel debug (/log debug if in fs_cli) >>>>> sofia global siptrace on >>>>> >>>>> reproduce >>>>> look at the logs or post them to pastebin and let us have a look. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Wed, Apr 6, 2011 at 1:11 PM, Mark Sobkow >>>>> wrote: >>>>>> I've never even heard of the session timers, much less programmed any >>>>>> into the software. What should I look for? I'm suspecting there may be >>>>>> a default timer that needs to be reset to disabled/no-timeout to get the >>>>>> behaviour we used to have. >>>>>> >>>>>> On 06/04/2011 11:15 AM, Anthony Minessale wrote: >>>>>>> no you are probably getting a session timer timeout. >>>>>>> >>>>>>> >>>>>>> On Wed, Apr 6, 2011 at 11:41 AM, Mark Sobkow >>>>>>> wrote: >>>>>>>> Has someone over the past few months added some sort of automatic hangup >>>>>>>> feature for parked calls? >>>>>>>> >>>>>>>> Our application lets an operator log in, and parks their call until >>>>>>>> there is a customer available for them to bridge to. Previously this >>>>>>>> worked just fine -- operators could sit for minutes or hours (much to >>>>>>>> the annoyance of managers), waiting for a call. >>>>>>>> >>>>>>>> However, something has changed. Freeswitch now seems to automatically >>>>>>>> unpark the call and hang it up, even though it hasn't been told to do >>>>>>>> _anything_ by the application. >>>>>>>> >>>>>>>> -- >>>>>>>> Mark Sobkow >>>>>>>> Senior Developer >>>>>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>>>>> 428 Victoria Ave >>>>>>>> Regina, SK S4N-0P6 >>>>>>>> Toll-Free: 800-289-8616-X533 >>>>>>>> Local: 306-359-6893-X533 >>>>>>>> Fax: 306-359-6879 >>>>>>>> Email: m.sobkow at marketelsystems.com >>>>>>>> Web: http://www.marketelsystems.com >>>>>>>> >>>>>>>> Visit our Blog for industry related information. >>>>>>>> http://marketel-systems.blogspot.com/ >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-dev mailing list >>>>>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>> -- >>>>>> Mark Sobkow >>>>>> Senior Developer >>>>>> MarkeTel Multi-Line Dialing Systems LTD. >>>>>> 428 Victoria Ave >>>>>> Regina, SK S4N-0P6 >>>>>> Toll-Free: 800-289-8616-X533 >>>>>> Local: 306-359-6893-X533 >>>>>> Fax: 306-359-6879 >>>>>> Email: m.sobkow at marketelsystems.com >>>>>> Web: http://www.marketelsystems.com >>>>>> >>>>>> Visit our Blog for industry related information. >>>>>> http://marketel-systems.blogspot.com/ >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-dev mailing list >>>>>> FreeSWITCH-dev at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>>>> http://www.freeswitch.org >>>>>> >>>> -- >>>> Mark Sobkow >>>> Senior Developer >>>> MarkeTel Multi-Line Dialing Systems LTD. >>>> 428 Victoria Ave >>>> Regina, SK S4N-0P6 >>>> Toll-Free: 800-289-8616-X533 >>>> Local: 306-359-6893-X533 >>>> Fax: 306-359-6879 >>>> Email: m.sobkow at marketelsystems.com >>>> Web: http://www.marketelsystems.com >>>> >>>> Visit our Blog for industry related information. >>>> http://marketel-systems.blogspot.com/ >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-dev mailing list >>>> FreeSWITCH-dev at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>>> http://www.freeswitch.org >>>> >>> >> >> -- >> Mark Sobkow >> Senior Developer >> MarkeTel Multi-Line Dialing Systems LTD. >> 428 Victoria Ave >> Regina, SK S4N-0P6 >> Toll-Free: 800-289-8616-X533 >> Local: 306-359-6893-X533 >> Fax: 306-359-6879 >> Email: m.sobkow at marketelsystems.com >> Web: http://www.marketelsystems.com >> >> Visit our Blog for industry related information. >> http://marketel-systems.blogspot.com/ >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org !DSPAM:4d9ceccc32761663321281! From anthony.minessale at gmail.com Thu Apr 7 22:55:06 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 7 Apr 2011 13:55:06 -0500 Subject: [Freeswitch-dev] Job Opening at Barracuda Networks / CudaTel - Technical Escalation Engineer Message-ID: Looking for someone well-versed in debugging voice and data using common tools like wireshark etc to take escalation incidents from tech support at CudaTel. Should be well versed in unix systems and internet tools and above average diagnostic skills. Preferred location or relocation to Ann Arbor MI but will entertain applicants from the bay area in CA as well. Please reply with resume to jobs at 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 at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From Min.Xie at convergys.com Tue Apr 5 22:54:09 2011 From: Min.Xie at convergys.com (Min Xie) Date: Tue, 5 Apr 2011 14:54:09 -0400 Subject: [Freeswitch-dev] "play_and_get_digits" blocked on call attaching to the conference. Message-ID: Hi, All, I am trying to execute "play_and_get_digits" on a call which is attaching to a conference. The command was sent to the channel but did not get executed immediately. I got this from the console: switch_core_session.c:954 Send signal sofia/internal/1001 at 192.168.0.16 [BREAK] Once I expel the call from the conference, the "play_and_get_digits" starts to execute. Is there a way that you can do "play_and_get_digits" on a call which is part of an active conference? Thanks! Min ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110405/184a4b9a/attachment.html From neon at artkis.pl Fri Apr 1 15:41:00 2011 From: neon at artkis.pl (Marcin Szymankiewicz) Date: Fri, 1 Apr 2011 13:41:00 +0200 Subject: [Freeswitch-dev] Library mod_odbc_query Message-ID: Hi. How do the library mod_obdc_query compilation under windows? Best regards. Marcin. From peter.olsson at visionutveckling.se Tue Apr 12 14:16:58 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 12 Apr 2011 12:16:58 +0200 Subject: [Freeswitch-dev] module load problem In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C500F61C@cooper> This is not at all supported by FreeSWITCH. If you want g729 support, buy real licenses from the FS team. FreeSWITCH does not support the usage of illegal g729 licenses. /Peter Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r barisyanar Skickat: den 24 mars 2011 16:43 Till: freeswitch-dev at lists.freeswitch.org ?mne: [Freeswitch-dev] module load problem Hi, I compiled a module using intel ipp library and freeswitch 1.0.7 source. SWITCH_API_VERSION is 5 in switch_types.h However I cannot load the module with fs_cli. Below is the error I get: 2011-03-24 16:35:00.584545 [CRIT] switch_loadable_module.c:928 Error Loading module /opt/freeswitch/mod/mod_tri.so **Trying to load an out of date module, please rebuild the module.** How could this happen? I expect the .so file has the same API_VERSION with the freeswitch. Baris !DSPAM:4da4226b32761158220547! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110412/034123e1/attachment-0001.html From peter.olsson at visionutveckling.se Tue Apr 12 14:18:15 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 12 Apr 2011 12:18:15 +0200 Subject: [Freeswitch-dev] CoreDump - Possibly User Error In-Reply-To: <1ec301cbea49$19c4a120$4d4de360$@com> References: <1ec301cbea49$19c4a120$4d4de360$@com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C500F61F@cooper> Bugs go to Jira - not the dev mailing list. /Peter Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Robert Huddleston Skickat: den 24 mars 2011 18:30 Till: freeswitch-dev at lists.freeswitch.org ?mne: [Freeswitch-dev] CoreDump - Possibly User Error http://pastebin.freeswitch.org/15833 I was editing mod_xml_cdr config file while FreeSwitch was up and running. Tried to reconnect to fs_cli and could not connect. Found that FreeSwitch had coredumped. Tried to use GDB BT but it's complaining about some symbols missing - which is odd as I built the box from GIT and source is installed. I admit I'm a n00b when it comes to GDB and BT Any help? !DSPAM:4da4227732768078111652! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110412/da72e92c/attachment.html From peter.olsson at visionutveckling.se Tue Apr 12 14:22:29 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 12 Apr 2011 12:22:29 +0200 Subject: [Freeswitch-dev] "play_and_get_digits" blocked on call attaching to the conference. In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C500F623@cooper> When it's inside the conference the call is handled by the conference module only. There are other api's, like "conference play" to play something to a specific member in a specific conference. /Peter Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Min Xie Skickat: den 5 april 2011 20:54 Till: 'freeswitch-dev at lists.freeswitch.org' ?mne: [Freeswitch-dev] "play_and_get_digits" blocked on call attaching to the conference. Hi, All, I am trying to execute "play_and_get_digits" on a call which is attaching to a conference. The command was sent to the channel but did not get executed immediately. I got this from the console: switch_core_session.c:954 Send signal sofia/internal/1001 at 192.168.0.16 [BREAK] Once I expel the call from the conference, the "play_and_get_digits" starts to execute. Is there a way that you can do "play_and_get_digits" on a call which is part of an active conference? Thanks! Min ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. !DSPAM:4da422c632761547312415! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110412/5409b75a/attachment.html From peter.olsson at visionutveckling.se Tue Apr 12 14:23:17 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 12 Apr 2011 12:23:17 +0200 Subject: [Freeswitch-dev] Freeswitch latest source rebuild issue under Windows In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C500F624@cooper> Don't use 1.0.7, it's old. Use latest git HEAD and try again. /Peter Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Andrew Keil Skickat: den 29 mars 2011 04:11 Till: freeswitch-dev at lists.freeswitch.org ?mne: [Freeswitch-dev] Freeswitch latest source rebuild issue under Windows To Freeswitch developers.... OK. Here goes, what I have done so far... 1) Setup a Virtual Machine running 32-bit XP SP3 (with all Critical Updates applied). That way I can quickly go back in time and try an install again if necessary. 2) Since I noticed that you now support Visual C++ 2010 Express, I downloaded and installed that (without the option for SQL Server Express 2008). 3) I also then installed the Windows SDK (7.1 (the latest version)). The reason for this is when I download your latest build of Freeswitch 1.0.7 today it complained about 64bit settings throughout various project files when opening Freeswitch.2010.express.sln 4) Then I simply downloaded your latest build from http://latest.freeswitch.org/ and extracted it out to c:\FreeSWITCH (so that there is a sub-directory freeswitch-1.0.7 below it with all the files) - this looks the same layout as the Freeswitch book example, except 1.0.7 version instead. 5) Opened the Freeswitch.2010.express.sln 5.1) Stated inside the Visual Studio Output Window: "Some of the properties associated with the solution could not be read." 6) Built the solution via F7 See attached for the build errors from Visual Studio. I guess my questions are the following: Q1) Should I be running Visual C++ 2010 Express or Visual C++ 2008 Express - it does not matter to me, however I always like to move with the times? Q2) Was I right in installing the Windows SDK (http://msdn.microsoft.com/en-us/windows/bb980924.aspx) after the installation of Visual C++ 2010 Express? Perhaps this should be added to the Freeswitch windows installation instructions page: http://wiki.freeswitch.org/wiki/Installation_for_Windows Q3) Should I expect the http://latest.freeswitch.org/ version of freeswitch to Build first time? Has this been tested? Q4) Should I simply use the 1.0.6 version with Visual C++ 2008 Express and follow the Freeswitch book word for word? This seems a reliable path, but I am a 'C' developer myself and would like to start out with a more up-to-date dev. environment and a version of freeswitch that is closer to the current version. Q5) Can someone make comments on the Build errors that are inside the attached file. Obviously I could go through and debug each build error, however I feel it best to have my questions answered above to avoid wasting time. Please note. I can happily start again and go back in time to the point prior to the installation of Visual C++ 2010 Express, since I am running a ESXi Virtual Machine. All I aiming for is a clean setup of FreeSWITCH under XP for testing and developing purposes (this is not to run in a production environment). I would like to be as up-to-date as possible in order to progress quickly (obviously I will install GIT/TortoiseGIT etc.. to make my dev. environment more integrated once I have a working base version of Freeswitch). Thanks in advance, Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110412/7e2f801d/attachment-0001.html From anthony.minessale at gmail.com Tue Apr 12 19:10:42 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 12 Apr 2011 10:10:42 -0500 Subject: [Freeswitch-dev] Questions on implementing mod_nelly codec In-Reply-To: References: Message-ID: Almost forgot about this thread, there was no reply for some time. See latest GIT and just configure your codec with accurate numbers and it should be right. is 32 ptime a requirement? you will get more performance out of 40 but 32 should work as of this latest version. commit 82e3d49fd20ca91306a87d43bb88613938b2c6fd Author: Anthony Minessale Date: Tue Apr 12 09:47:11 2011 -0500 add L16 def for 32ms and allow timer matrix to drop to 1ms to support nelly On Tue, Mar 22, 2011 at 4:46 PM, Richard Alam wrote: > Yes, we plan on contributing the module once it's working. > > On Tue, Mar 22, 2011 at 5:19 PM, Anthony Minessale > wrote: >> Are you planning to contribute this module? >> >> On Mon, Mar 21, 2011 at 1:58 PM, Richard Alam wrote: >>> Hi, >>> >>> We've been trying to implement a mod_nelly codec but so far has been >>> unsuccessful. We use a Nellymoser compatible codec submitted to ffmpeg >>> (see June 16, 2008 entry of http://ffmpeg.org/). >>> >>> Our plan is to use Flash using Nellymoser to connect to a conference >>> in FreeSWITCH. >>> >>> The hurdle seems to be that Nellymoser decodes 64 byte nelly audio to >>> 512 byte L16. Nellymoser is 11-Khz 8-bit per sample with 32 ptime. >>> However, L16 implementations are all 20 ptime >>> which have uncompressed bytes in multiples of 320. This doesn't align >>> with what mod_nelly expects when encoding (convert 512 bytes to 64 >>> bytes) and decoding (64 bytes to 512 bytes). >>> >>> Here is the SWITCH_MODULE_LOAD_FUNCTION(mod_nelly_load) >>> [http://pastebin.freeswitch.org/15764] now. We've played around with >>> the samples per second (SPS), actual samples per second (ASPS), >>> bits per second (BPS), and the ptime (PTIME) but with no luck. It's >>> either we get bad very choppy audio, incompatible destination (when >>> codecs can't match), or no audio at all. We've managed to get the >>> "You will now be placed into the conference" audio played correctly >>> but the audio from the caller is bad. Just the paying of wav file from >>> FS is correct. >>> >>> My question is how does transcoding works in FS? Am I correct to >>> assume decoding as Nelly->L16 and encoding as L16->Nelly? >>> So basically the media flow for our use case is Nelly -> L16 --> >>> mod_conference -> L16 -> Nelly. >>> >>> Do I need to make the frames align? In this case, Nelly expect 256 >>> samples per frame while L16 has 160, 320, ... samples per frame. >>> >>> Hope I've articulated clearly what we are trying to accomplish. >>> >>> Suggestions and ideas on what to try out next is much appreciated. >>> >>> Thanks in advance. >>> >>> Richard >>> >>> >>> -- >>> --- >>> BigBlueButton >>> http://www.bigbluebutton.org >>> http://code.google.com/p/bigbluebutton >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > > -- > --- > BigBlueButton > http://www.bigbluebutton.org > http://code.google.com/p/bigbluebutton > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From steveu at coppice.org Tue Apr 12 20:40:10 2011 From: steveu at coppice.org (Steve Underwood) Date: Wed, 13 Apr 2011 00:40:10 +0800 Subject: [Freeswitch-dev] Questions on implementing mod_nelly codec In-Reply-To: References: Message-ID: <4DA4806A.1030704@coppice.org> Hi, The Nelly Moser codec seems to be patent encumbered, so accepting a module for need to be considered with caution. It doesn't seem to be that useful, now. It used to have value, because it was needed for flash. Flash now supports speex, which is a better option. Steve On 04/12/2011 11:10 PM, Anthony Minessale wrote: > Almost forgot about this thread, there was no reply for some time. > > See latest GIT and just configure your codec with accurate numbers and > it should be right. > is 32 ptime a requirement? you will get more performance out of 40 but > 32 should work as of this latest version. > > commit 82e3d49fd20ca91306a87d43bb88613938b2c6fd > Author: Anthony Minessale > Date: Tue Apr 12 09:47:11 2011 -0500 > > add L16 def for 32ms and allow timer matrix to drop to 1ms to support nelly > > > On Tue, Mar 22, 2011 at 4:46 PM, Richard Alam wrote: >> Yes, we plan on contributing the module once it's working. >> >> On Tue, Mar 22, 2011 at 5:19 PM, Anthony Minessale >> wrote: >>> Are you planning to contribute this module? >>> >>> On Mon, Mar 21, 2011 at 1:58 PM, Richard Alam wrote: >>>> Hi, >>>> >>>> We've been trying to implement a mod_nelly codec but so far has been >>>> unsuccessful. We use a Nellymoser compatible codec submitted to ffmpeg >>>> (see June 16, 2008 entry of http://ffmpeg.org/). >>>> >>>> Our plan is to use Flash using Nellymoser to connect to a conference >>>> in FreeSWITCH. >>>> >>>> The hurdle seems to be that Nellymoser decodes 64 byte nelly audio to >>>> 512 byte L16. Nellymoser is 11-Khz 8-bit per sample with 32 ptime. >>>> However, L16 implementations are all 20 ptime >>>> which have uncompressed bytes in multiples of 320. This doesn't align >>>> with what mod_nelly expects when encoding (convert 512 bytes to 64 >>>> bytes) and decoding (64 bytes to 512 bytes). >>>> >>>> Here is the SWITCH_MODULE_LOAD_FUNCTION(mod_nelly_load) >>>> [http://pastebin.freeswitch.org/15764] now. We've played around with >>>> the samples per second (SPS), actual samples per second (ASPS), >>>> bits per second (BPS), and the ptime (PTIME) but with no luck. It's >>>> either we get bad very choppy audio, incompatible destination (when >>>> codecs can't match), or no audio at all. We've managed to get the >>>> "You will now be placed into the conference" audio played correctly >>>> but the audio from the caller is bad. Just the paying of wav file from >>>> FS is correct. >>>> >>>> My question is how does transcoding works in FS? Am I correct to >>>> assume decoding as Nelly->L16 and encoding as L16->Nelly? >>>> So basically the media flow for our use case is Nelly -> L16 --> >>>> mod_conference -> L16 -> Nelly. >>>> >>>> Do I need to make the frames align? In this case, Nelly expect 256 >>>> samples per frame while L16 has 160, 320, ... samples per frame. >>>> >>>> Hope I've articulated clearly what we are trying to accomplish. >>>> >>>> Suggestions and ideas on what to try out next is much appreciated. >>>> >>>> Thanks in advance. >>>> >>>> Richard From msc at freeswitch.org Wed Apr 13 08:05:15 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 12 Apr 2011 21:05:15 -0700 Subject: [Freeswitch-dev] FreeSWITCH: No Longer The Best Kept Secret In OSS VoIP Software Message-ID: As you know, the FreeSWITCH project and community are both growing rapidly. We are always looking for people to step up and help out. This article talks about some of the things that you can do to assist in dealing with the growing pains that we all feel: http://www.freeswitch.org/node/320 Thanks for your support and please keep up the good work! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110412/e992183b/attachment.html From msc at freeswitch.org Wed Apr 13 19:07:21 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 13 Apr 2011 08:07:21 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today - SIPVicious (friendly-scanner) Author Message-ID: Hello! Please join us for the conference call today as we welcome Sandro Gauci, author of SIPVicious. You may know this better by the name "friend-scanner" that the script kiddies are using. Hear from the SIPVicious author to find out what lead to him creating this tool, some of the challenges he's faced, and how he's dealt with nefarious types who've been misusing it. Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_13 Talk to you all soon! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110413/64dac92b/attachment.html From math.parent at gmail.com Thu Apr 14 16:57:48 2011 From: math.parent at gmail.com (Mathieu Parent) Date: Thu, 14 Apr 2011 14:57:48 +0200 Subject: [Freeswitch-dev] Implementing Forward-{All, Busy, NoAnswer}, ring policies and DND in FS Message-ID: Hello, *Context*: --------- I want to implement the following in mod_skinny (FS-3048 and FS-3047) in a way shared with other endpoints modules: - call forwarding[cfw], - DND and ring policies *Call forwarding*: --------- There are three main kinds of forwarding: - forward all (immediately) - forward when line is busy - forward when no answer We should also store for each the state of the forwarding (enabled or not). I think a reasonable default here would be: - to forward "no answer" to voicemail (and enabled) - to forward busy to voicemail (but disabled) - to forward all to "" (and disabled) We also should define what a busy line is, especially on shared lines (perhaps define a busy threshold, defaulting to 1). Call forwarding should be implemented in dialplan, but endpoint modules should know when a call is forwarded (the bridge event is probably sufficient). *Ring policies and DND*: --------- A do-not-disturb flag can be set on an extension so that the phone doesn't ring (but the user still knows he/she is called). DND is a specific case of ring policies. I propose the following default ring policies: - on idle: ring, blink (toggled to blink with DND button) - on busy: blink only (also available: ring once) This should be implemented per endpoint module. *Proposal*: --------- I propose to use hashes to store the various data: Where ${realm} is one of: - ${domain_name}-forward-{all,busy,noanswer}-destination: a destination number - ${domain_name}-forward-{all,busy,noanswer}-status: true/false - ${domain_name}-busy-threshold: [0-9]+ - ${domain_name}-ring-on-idle: true/false/once - ${domain_name}-blink-on-idle: true/false - ${domain_name}-ring-on-busy: true/false/once - ${domain_name}-blink-on-busy: true/false The dialplan should be enhanced to manage forwarding. The endpoint modules should be enhanced to manage forwarding (notify that the call has been forwarded) and ring policies. Does this makes sense? How can we improve it? *Next steps*: --------- - proposal discussion - forward-* dialplan implementation (I need help here, I am not an dialplan expert) - mod_skinny forward-* implementation FS-3048 - mod_skinny ring policies implementation FS-3047 - other endpoints implementation Regards Mathieu [cfw]: http://en.wikipedia.org/wiki/Call_forwarding [dnd]: http://en.wikipedia.org/wiki/Do_Not_Disturb_%28telecommunications%29 From msc at freeswitch.org Wed Apr 20 20:42:38 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Apr 2011 09:42:38 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today - VoIP Abuse/PBX Honeypot Message-ID: Hello all! Our agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_20 At about 2PM EDT (11AM PDT) J. Oquendo will join us to talk about PBX Honeypot and VoIP Abuse projects. We will still start the conference at 1PM EDT/10AM PDT and do our agenda items until J. joins us. Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110420/290108bb/attachment.html From michal.bielicki at seventhsignal.de Tue Apr 12 22:44:10 2011 From: michal.bielicki at seventhsignal.de (Michal Bielicki) Date: Tue, 12 Apr 2011 20:44:10 +0200 Subject: [Freeswitch-dev] Valet Parking improvements In-Reply-To: <8C8A3D4965236A42BDFF1758727F049A263BAD@ITEX1.bc.local> References: <8C8A3D4965236A42BDFF1758727F049A263BAD@ITEX1.bc.local> Message-ID: The easiest is to create a ticket in Jira and add the patches to it and let everybody know on the list where they are and ask people to try it. Am 30.03.2011 um 16:08 schrieb Josh M. Patten: > Our development team is in the process of writing some improvements into the valet parking module that we hope to get pushed upstream to the community. Among these improvements are: > > ? Ability to set timeout value > > o Ability to send call back to extension that originally parked the call after timeout > > o Ability to send call to static extension (for example, an operator) after timeout > > o Ability to disconnect the call after timeout > > ? Ability to set escape button, such as *, #, 0-9 > > o Ability to send call back to extension that originally parked the call after caller presses escape button > > o Ability to send call to static extension (for example, an operator) after caller presses escape button > > o Ability to disconnect the call after caller presses escape button > > > > In order to make these changes we are going to have to add a few arguments to the module. My questions are: > > > > How should we get this code submitted for review and testing? > > What do we need to do to get this code merged after it is reviewed and tested? > > > > Thanks! > > > > > > Josh Patten > > Brazos County Network Engineer > > 979.361.4676 > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org Michal Bielicki Gesch?ftsf?hrer / CEO Seventh Signal Ltd. & Co. KG Weigandufer 45, B?ro 115, D-12059 Berlin Voice: +49 30 60988730 Amtsgericht Charlottenburg HRA 44413 B Ust.-ID: DE266981999 Gesch?ftsf?hrer: Michal Bielicki Pers?nlich Haftende Gesellschafterin: Seventh Signal Ltd, 69 Great Hampton St. Birmingham, B18 6EW, GB, Company Nr.: 06889439 WWW.: http://www.seventhsignal.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110412/3d9e4397/attachment-0001.html From daniel.bryars at aeriandi.com Wed Apr 13 13:42:25 2011 From: daniel.bryars at aeriandi.com (Daniel Bryars) Date: Wed, 13 Apr 2011 10:42:25 +0100 Subject: [Freeswitch-dev] Freeswitch latest source rebuild issue under Windows References: <549CFEF87AEDE841A38E9D15EAB4C04C58C500F624@cooper> Message-ID: <2CC6B62FEE3A634CAD6B5D44FC9315E24F9CFC1329@aeramsdc1.AERIANDI.LAN> In my experience I have found that the nightly snapshot (http://files.freeswitch.org/freeswitch-snapshot.tar.gz)builds reliably on Windows. I use FreeSwitch.2010.sln but last night I tried the FreeSwitch.2010.express.sln and this worked fine too. However, I have not personally been able to build FS successfully if I get the HEAD from GIT directly. I spent some time last night looking at the difference between the files in the "snapshot" and the files from the HEAD (via git) and there are several empty header files in the version from GIT which are causing the build to fail. Alongside these header files are some ".IN" files which look like they are templates for the header files. I'm guessing there is some sort of intermediate pre-processor step to parse these .IN files and create the headers. I don't know. Does this make any sense to anyone? @Andrew - Perhaps you have the same problem? I'd like to use the HEAD rather than the "snapshot"; I plan on spending some time on this tomorrow night, I'll let you know what I find out. Dan From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 12 April 2011 11:23 To: 'freeswitch-dev at lists.freeswitch.org' Subject: Re: [Freeswitch-dev] Freeswitch latest source rebuild issue under Windows Don't use 1.0.7, it's old. Use latest git HEAD and try again. /Peter Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Andrew Keil Skickat: den 29 mars 2011 04:11 Till: freeswitch-dev at lists.freeswitch.org ?mne: [Freeswitch-dev] Freeswitch latest source rebuild issue under Windows To Freeswitch developers.... OK. Here goes, what I have done so far... 1) Setup a Virtual Machine running 32-bit XP SP3 (with all Critical Updates applied). That way I can quickly go back in time and try an install again if necessary. 2) Since I noticed that you now support Visual C++ 2010 Express, I downloaded and installed that (without the option for SQL Server Express 2008). 3) I also then installed the Windows SDK (7.1 (the latest version)). The reason for this is when I download your latest build of Freeswitch 1.0.7 today it complained about 64bit settings throughout various project files when opening Freeswitch.2010.express.sln 4) Then I simply downloaded your latest build from http://latest.freeswitch.org/ and extracted it out to c:\FreeSWITCH (so that there is a sub-directory freeswitch-1.0.7 below it with all the files) - this looks the same layout as the Freeswitch book example, except 1.0.7 version instead. 5) Opened the Freeswitch.2010.express.sln 5.1) Stated inside the Visual Studio Output Window: "Some of the properties associated with the solution could not be read." 6) Built the solution via F7 See attached for the build errors from Visual Studio. I guess my questions are the following: Q1) Should I be running Visual C++ 2010 Express or Visual C++ 2008 Express - it does not matter to me, however I always like to move with the times? Q2) Was I right in installing the Windows SDK (http://msdn.microsoft.com/en-us/windows/bb980924.aspx) after the installation of Visual C++ 2010 Express? Perhaps this should be added to the Freeswitch windows installation instructions page: http://wiki.freeswitch.org/wiki/Installation_for_Windows Q3) Should I expect the http://latest.freeswitch.org/ version of freeswitch to Build first time? Has this been tested? Q4) Should I simply use the 1.0.6 version with Visual C++ 2008 Express and follow the Freeswitch book word for word? This seems a reliable path, but I am a 'C' developer myself and would like to start out with a more up-to-date dev. environment and a version of freeswitch that is closer to the current version. Q5) Can someone make comments on the Build errors that are inside the attached file. Obviously I could go through and debug each build error, however I feel it best to have my questions answered above to avoid wasting time. Please note. I can happily start again and go back in time to the point prior to the installation of Visual C++ 2010 Express, since I am running a ESXi Virtual Machine. All I aiming for is a clean setup of FreeSWITCH under XP for testing and developing purposes (this is not to run in a production environment). I would like to be as up-to-date as possible in order to progress quickly (obviously I will install GIT/TortoiseGIT etc.. to make my dev. environment more integrated once I have a working base version of Freeswitch). Thanks in advance, Andrew Keil ________________________________ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from your computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110413/23d67174/attachment-0001.html From Min.Xie at convergys.com Fri Apr 15 00:16:02 2011 From: Min.Xie at convergys.com (Min Xie) Date: Thu, 14 Apr 2011 16:16:02 -0400 Subject: [Freeswitch-dev] How to use "say" command for a conference. Message-ID: All, We tried to do a "say" command to a conference but always got the following. What's the right console command line parameters to perform "conference say" or "conference saymember"? freeswitch at 192.168.0.123@internal> conference myconference-name say "hello world" (say) Error! say Thanks! Min Xie ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110414/2960b11a/attachment-0001.html From singhujjwal at gmail.com Fri Apr 15 16:02:13 2011 From: singhujjwal at gmail.com (Ujjwal SIngh) Date: Fri, 15 Apr 2011 17:32:13 +0530 Subject: [Freeswitch-dev] Noise heard when a user holds a conference in SRTP mode Message-ID: Hi, I have FreeSwitch configured as a conference bridge, the sip_secure_media=true in dialplan and vars.xml. User A, B and C are in a conference, the media is established in SRTP mode, when any user presses hold, other members hear noise, while in RTP mode normal Music on Hold is heard. Can anybody please help what is going wrong in SRTP case. I have uploaded FreeSwitch log in pastebin below http://pastebin.freeswitch.org/16103 Thanks for the help. Regards, Ujjwal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110415/4a721e9c/attachment-0001.html From habib at alexcoder.com Mon Apr 18 13:52:34 2011 From: habib at alexcoder.com (Mohammed Habib) Date: Mon, 18 Apr 2011 11:52:34 +0200 Subject: [Freeswitch-dev] Originated session callback. Message-ID: I need help getting events from originated session. This is my lua script: function onInput_MainSession(s, type, obj) -- This one is working fine. freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); end function onInput_NewSession(s, type, obj) -- This one is never called. freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); end session:answer(); session:setInputCallback("onInput_MainSession"); session:sleep(200); session:execute("detect_speech", "unimrcp testgrammer trestgrammer"); newsession = freeswitch.Session("user/1002"); newsession:setInputCallback("onInput_NewSession"); newsession:sleep(200); newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer"); while ((session:ready() == true) ) and (newsession:ready() == true) do -- Loop sleep(200); end I am unable to capture any of the new session events or dtmf. Please help. Thank you, Mohammed Habib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110418/9974ade9/attachment.html From mochouinard at moctel.com Wed Apr 20 15:49:11 2011 From: mochouinard at moctel.com (Marc Olivier Chouinard) Date: Wed, 20 Apr 2011 07:49:11 -0400 Subject: [Freeswitch-dev] Proposal to add a Deprecated Log Level Message-ID: <4DAEC837.4080507@moctel.com> Hi, I would like to propose adding a Deprecated Log Level to Freeswitch. This would serve as a special warning to user that some feature shouldn't be used and might be removed in a more visible way. Also I propose we use a number for every log we do. like DEPRECATED00001 (or whatever), and create wiki entry related to it. In the wiki, we would have to put the information the user need to change to use the current function. Also we can specify a due date for removal of the feature. We could also create a special macro that we can specify the unix time for the removal of the feature and make it display as critical error if the date is less than 1 month (or something). This would help to cleanup the code in the long term without having the adverse affect of creating surprises to user. Moc From mochouinard at moctel.com Wed Apr 20 16:08:47 2011 From: mochouinard at moctel.com (Marc Olivier Chouinard) Date: Wed, 20 Apr 2011 08:08:47 -0400 Subject: [Freeswitch-dev] [Re] Implementing Forward-{All, Busy, NoAnswer}, ring policies and DND in FS Message-ID: <4DAECCCF.4070009@moctel.com> Hi Mathieu, I think it a great idea... it something I was thinking to go look for that server side forward feature of SIP supported by Polycom. Going to be hard to define something standard for all endpoint. The basic forward all,no answer(number of ring), busy are fine to do, but the ring-on-idle and blink-on-idle... seem specific to skinny. As far as I know, in sip you would send an alert with a silence ring tone. So not sure how we would want to implement that one. In case of SIP implementation, it probably a SIP XML msg we receive... So what do we do with it ? Generate a dummy call to the dialplan ? Or should the different endpoint generate an standard event, and we make a new module that monitor those event and does populate whatever is needed ? I'm trying to think long term also of other feature we might want to implement (SIP Conferencing monitoring support for example...) Moc From brian at freeswitch.org Thu Apr 21 01:43:34 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 20 Apr 2011 16:43:34 -0500 Subject: [Freeswitch-dev] Noise heard when a user holds a conference in SRTP mode In-Reply-To: References: Message-ID: I'm going to guess your device method of placing you on hold is wrong. And possibly doesn't encrypt the data or doesn't signal a key change... is it white noise? /b On Apr 15, 2011, at 7:02 AM, Ujjwal SIngh wrote: > Hi, > > I have FreeSwitch configured as a conference bridge, the > sip_secure_media=true in dialplan and vars.xml. > User A, B and C are in a conference, the media is established in SRTP mode, > when any user presses hold, other members hear noise, while in RTP mode > normal Music on Hold is heard. > > Can anybody please help what is going wrong in SRTP case. > I have uploaded FreeSwitch log in pastebin below > > http://pastebin.freeswitch.org/16103 > > > Thanks for the help. > Regards, > Ujjwal > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From singhujjwal at gmail.com Thu Apr 21 15:26:37 2011 From: singhujjwal at gmail.com (Ujjwal SIngh) Date: Thu, 21 Apr 2011 16:56:37 +0530 Subject: [Freeswitch-dev] Noise heard when a user holds a conference in SRTP mode In-Reply-To: References: Message-ID: Yes Brian its a white noise, but the hold method works perfectly fine when used with RTP, the hold is implemented according to the draft http://datatracker.ietf.org/doc/draft-worley-service-example/?include_text=1 Regards, Ujjwal On Thu, Apr 21, 2011 at 3:13 AM, Brian West wrote: > I'm going to guess your device method of placing you on hold is wrong. And > possibly doesn't encrypt the data or doesn't signal a key change... is it > white noise? > > /b > > On Apr 15, 2011, at 7:02 AM, Ujjwal SIngh wrote: > > > Hi, > > > > I have FreeSwitch configured as a conference bridge, the > > sip_secure_media=true in dialplan and vars.xml. > > User A, B and C are in a conference, the media is established in SRTP > mode, > > when any user presses hold, other members hear noise, while in RTP mode > > normal Music on Hold is heard. > > > > Can anybody please help what is going wrong in SRTP case. > > I have uploaded FreeSwitch log in pastebin below > > > > http://pastebin.freeswitch.org/16103 > > > > > > Thanks for the help. > > Regards, > > Ujjwal > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110421/77fd18c8/attachment.html From brian at freeswitch.org Thu Apr 21 18:08:21 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 21 Apr 2011 09:08:21 -0500 Subject: [Freeswitch-dev] Noise heard when a user holds a conference in SRTP mode In-Reply-To: References: Message-ID: Yah SRTP might NEVER work with that method of hold because now the MOH server and the endpoint have to exchange SRTP keys when you refer to them and they probably are NOT doing that right now which is why you get white noise. /b On Apr 21, 2011, at 6:26 AM, Ujjwal SIngh wrote: > Yes Brian its a white noise, but the hold method works perfectly fine when > used with RTP, the hold is > > implemented according to the draft > http://datatracker.ietf.org/doc/draft-worley-service-example/?include_text=1 > > > Regards, > Ujjwal > > > > > On Thu, Apr 21, 2011 at 3:13 AM, Brian West wrote: > >> I'm going to guess your device method of placing you on hold is wrong. And >> possibly doesn't encrypt the data or doesn't signal a key change... is it >> white noise? >> >> /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110421/77598f2f/attachment.html From msc at freeswitch.org Thu Apr 21 23:12:21 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 Apr 2011 12:12:21 -0700 Subject: [Freeswitch-dev] How to use "say" command for a conference. In-Reply-To: References: Message-ID: Which TTS module do you have installed? -MC On Thu, Apr 14, 2011 at 1:16 PM, Min Xie wrote: > All, > > > > We tried to do a ?say? command to a conference but always got the > following. > > What?s the right console command line parameters to perform ?conference > say? or ?conference saymember?? > > > > freeswitch at 192.168.0.123@internal> conference myconference-name say ?hello > world? > > (say) Error! > > say > > > > Thanks! > > Min Xie > > ------------------------------ > NOTICE: The information contained in this electronic mail transmission is > intended by Convergys Corporation for the use of the named individual or > entity to which it is directed and may contain information that is > privileged or otherwise confidential. If you have received this electronic > mail transmission in error, please delete it from your system without > copying or forwarding it, and notify the sender of the error by reply email > or by telephone (collect), so that the sender's address records can be > corrected. > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110421/fd4139bc/attachment-0001.html From jalsot at gmail.com Fri Apr 22 21:34:00 2011 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Fri, 22 Apr 2011 19:34:00 +0200 Subject: [Freeswitch-dev] [article] High-performance Timing on Linux / Windows Message-ID: Hello, I've found an article about the topic, maybe you can find something in it: http://tdistler.com/2010/06/27/high-performance-timing-on-linux-windows If not, please ignore :) Regards, T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110422/01df0b5c/attachment.html From anthony.minessale at gmail.com Sat Apr 23 01:25:21 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 22 Apr 2011 16:25:21 -0500 Subject: [Freeswitch-dev] [article] High-performance Timing on Linux / Windows In-Reply-To: References: Message-ID: We are well beyond the knowledge in that article. Try out latest GIT especially with a newer kernel. On Fri, Apr 22, 2011 at 12:34 PM, Tamas Jalsovszky wrote: > Hello, > > I've found an article about the topic, maybe you can find something in it: > http://tdistler.com/2010/06/27/high-performance-timing-on-linux-windows > > If not, please ignore :) > > Regards, > ?? T. > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From khovayko at gmail.com Sat Apr 23 02:31:45 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Fri, 22 Apr 2011 18:31:45 -0400 Subject: [Freeswitch-dev] Help: how to disable dynamic RTP mapping? Message-ID: <4DB201D1.8080604@gmail.com> Hi, My FreeSWITCH machine is located behind the NAT-device. When I receive incoming call, I see 10 secs delay between I picked up handset and voice connection establishing. During these 10s, caller hears long beeps, as same as "ringing" signal. I investigated problem, and found: after recipient pick up handset, FreeSWITCH tried dynamically map two UDP ports for RTP-connections. I use Fios router from Verizon, and I assume, it slowly adding dynamic port mapping. It adds one RTP port per 5s. So, when FS dynamically maps both ports, it loose 10 seconds (5+5). Log follows. I have added static rule into router -- map UDP ports 20000~25000 to FS box. And, in the FS_box, I added into file ./autoload_configs/switch.conf.xml But, anyway, FS tries dynamically assign ports for each income call, and lost 10s every time. Please suggest, how to disable dynamic port mapping, and use static port range mapping (20000-25000) only? I assume, this will increase speedup of my system. Thanks, Oleg 2011-04-22 18:14:45.590723 [DEBUG] switch_channel.c:2813 (sofia/internal/sip:1001 at 192.168.1.151:5060) Callstate Change RINGING -> ACTIVE . . . . 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1000 at 192.168.1.152:5060 [BREAK] 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1288 Session 57 (sofia/internal/sip:1000 at 192.168.1.152:5060) Locked, Waiting on external entities <5s delay> 2011-04-22 18:14:50.018621 [DEBUG] sofia_reg.c:1765 Changing expire time to 109 by request of proxy sip:callcentric.com 2011-04-22 18:14:50.209565 [DEBUG] switch_nat.c:502 mapped public port 20794 protocol UDP to localport 20794 2011-04-22 18:14:50.341416 [DEBUG] sofia_reg.c:1765 Changing expire time to 104 by request of proxy sip:callcentric.com <5s delay> 2011-04-22 18:14:55.213775 [DEBUG] switch_nat.c:502 mapped public port 20795 protocol UDP to localport 20795 2011-04-22 18:14:55.213775 [DEBUG] sofia_glue.c:3001 AUDIO RTP [sofia/external/2404760839 at 66.54.140.46] 192.168.1.5 port 20794 -> 204.11.192.22 port 60140 codec: 0 ms: 20 2011-04-22 18:14:55.213775 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms From anthony.minessale at gmail.com Sat Apr 23 03:36:57 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 22 Apr 2011 18:36:57 -0500 Subject: [Freeswitch-dev] Help: how to disable dynamic RTP mapping? In-Reply-To: <4DB201D1.8080604@gmail.com> References: <4DB201D1.8080604@gmail.com> Message-ID: run freeswitch -nonat On Fri, Apr 22, 2011 at 5:31 PM, Oleg Khovayko wrote: > Hi, > > My FreeSWITCH machine is located behind the NAT-device. > > When I receive incoming call, I see 10 secs delay between I picked up > handset and voice connection establishing. > During these 10s, caller hears long beeps, as same as "ringing" signal. > > I investigated problem, and found: after recipient pick up handset, > FreeSWITCH tried dynamically map two UDP ports for RTP-connections. > > I use Fios router from Verizon, and I assume, it slowly adding dynamic > port mapping. It adds one RTP port per 5s. > So, when FS dynamically maps both ports, it loose 10 seconds (5+5). Log > follows. > > I have added static rule into router -- map UDP ports 20000~25000 to FS > box. And, in the FS_box, I added into file > ./autoload_configs/switch.conf.xml > > > > > But, anyway, FS tries dynamically assign ports for each income call, and > lost 10s every time. > > Please suggest, how to disable dynamic port mapping, and use static port > range mapping (20000-25000) only? > I assume, this will increase speedup of my system. > > Thanks, > > Oleg > > > > 2011-04-22 18:14:45.590723 [DEBUG] switch_channel.c:2813 > (sofia/internal/sip:1001 at 192.168.1.151:5060) Callstate Change RINGING -> > ACTIVE > > . . . . > 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1116 Send > signal sofia/internal/sip:1000 at 192.168.1.152:5060 [BREAK] > 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1288 Session 57 > (sofia/internal/sip:1000 at 192.168.1.152:5060) Locked, Waiting on external > entities > <5s delay> > 2011-04-22 18:14:50.018621 [DEBUG] sofia_reg.c:1765 Changing expire time > to 109 by request of proxy sip:callcentric.com > 2011-04-22 18:14:50.209565 [DEBUG] switch_nat.c:502 mapped public port > 20794 protocol UDP to localport 20794 > 2011-04-22 18:14:50.341416 [DEBUG] sofia_reg.c:1765 Changing expire time > to 104 by request of proxy sip:callcentric.com > <5s delay> > 2011-04-22 18:14:55.213775 [DEBUG] switch_nat.c:502 mapped public port > 20795 protocol UDP to localport 20795 > 2011-04-22 18:14:55.213775 [DEBUG] sofia_glue.c:3001 AUDIO RTP > [sofia/external/2404760839 at 66.54.140.46] 192.168.1.5 port 20794 -> > 204.11.192.22 port 60140 codec: 0 ms: 20 > 2011-04-22 18:14:55.213775 [DEBUG] switch_rtp.c:1623 Starting timer > [soft] 160 bytes per 20ms > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From gabe at gundy.org Sat Apr 23 03:55:09 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 22 Apr 2011 17:55:09 -0600 Subject: [Freeswitch-dev] [article] High-performance Timing on Linux / Windows In-Reply-To: References: Message-ID: Doesn't your work in this area make the platform differences (including Windows) a non-issue? http://www.freeswitch.org/node/316 Best, Gabe Yes, I see the date. On Fri, Apr 22, 2011 at 3:25 PM, Anthony Minessale wrote: > We are well beyond the knowledge in that article. ?Try out latest GIT > especially with a newer kernel. > > > > On Fri, Apr 22, 2011 at 12:34 PM, Tamas Jalsovszky wrote: >> Hello, >> >> I've found an article about the topic, maybe you can find something in it: >> http://tdistler.com/2010/06/27/high-performance-timing-on-linux-windows >> >> If not, please ignore :) >> >> Regards, >> ?? T. >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > From jaybinks at gmail.com Sat Apr 23 06:04:55 2011 From: jaybinks at gmail.com (Jay Binks) Date: Sat, 23 Apr 2011 12:04:55 +1000 Subject: [Freeswitch-dev] [article] High-performance Timing on Linux / Windows In-Reply-To: References: Message-ID: <8D8E6E96-D886-4350-B650-61E580F3E7D5@gmail.com> Ummmm. ..... ??? ;) Jay On 23/04/2011, at 9:55 AM, Gabriel Gunderson wrote: > Doesn't your work in this area make the platform differences > (including Windows) a non-issue? > > http://www.freeswitch.org/node/316 > > Best, > Gabe > > > > > > > > > > Yes, I see the date. > > On Fri, Apr 22, 2011 at 3:25 PM, Anthony Minessale > wrote: >> We are well beyond the knowledge in that article. Try out latest GIT >> especially with a newer kernel. >> >> >> >> On Fri, Apr 22, 2011 at 12:34 PM, Tamas Jalsovszky wrote: >>> Hello, >>> >>> I've found an article about the topic, maybe you can find something in it: >>> http://tdistler.com/2010/06/27/high-performance-timing-on-linux-windows >>> >>> If not, please ignore :) >>> >>> Regards, >>> T. >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From khovayko at gmail.com Mon Apr 25 04:13:25 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Sun, 24 Apr 2011 20:13:25 -0400 Subject: [Freeswitch-dev] Help: how to disable dynamic RTP mapping? In-Reply-To: References: <4DB201D1.8080604@gmail.com> Message-ID: <4DB4BCA5.6060809@gmail.com> Anthony Minessale wrote: > run freeswitch -nonat > > Antony, Thank you very much, I run FreeSWITCH with this option, and really, 10s delay is gone. And, works almost everything, but one thing: I have IpKall forwarding number, which forwards calls to number at mysite.cx:5080. With the old NAT mode, it was work OK. In the -nonat mode, call comes in, I picked up handset - but heard only one-side voice: from fs_client to IPKALL PSTN caller. Voice from the caller to client does not traversing. When I set in the IpKall account forward to in.callcentric.com, and FS is registered as client on that CallCentric account - everything works OK, voice goes to both directions. This time, I keep this workaround. I noticed, when call establishing correctly, following debug-message presents in the log-file twice per call: switch_rtp.c:3082 Correct ip/port confirmed. With this problem call, I see this message only once. I assume, something wrong with RTP connection establishing. Please, suggest me, how can I fix this trouble, or idea, what to see myself. Log of one-side-call in there: http://olegh.ath.cx/fs_log-2010-04-24.txt Thanks, Oleg From khovayko at gmail.com Mon Apr 25 04:45:32 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Sun, 24 Apr 2011 20:45:32 -0400 Subject: [Freeswitch-dev] Help: how to disable dynamic RTP mapping? In-Reply-To: References: <4DB201D1.8080604@gmail.com> Message-ID: <4DB4C42C.7030204@gmail.com> Anthony Minessale wrote: > run freeswitch -nonat > > Antony, I solved the problem, I removed default "auto" from internal/external profiles, and set up: Everything according Wiki page: http://wiki.freeswitch.org/wiki/NAT_Traversal Just strange behavior: When I press F9 in the fs_cli, I see strange value for "Ext-RTP-IP" RTP-IP 192.168.1.5 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.1.5 Ext-SIP-IP 173.79.237.112 Line from vars.xml looks OK: I assume, Ext-RTP-IP must be 173.79.237.112 - as same as 173.79.237.112. > On Fri, Apr 22, 2011 at 5:31 PM, Oleg Khovayko wrote: > >> Hi, >> >> My FreeSWITCH machine is located behind the NAT-device. >> >> When I receive incoming call, I see 10 secs delay between I picked up >> handset and voice connection establishing. >> During these 10s, caller hears long beeps, as same as "ringing" signal. >> >> I investigated problem, and found: after recipient pick up handset, >> FreeSWITCH tried dynamically map two UDP ports for RTP-connections. >> >> I use Fios router from Verizon, and I assume, it slowly adding dynamic >> port mapping. It adds one RTP port per 5s. >> So, when FS dynamically maps both ports, it loose 10 seconds (5+5). Log >> follows. >> >> I have added static rule into router -- map UDP ports 20000~25000 to FS >> box. And, in the FS_box, I added into file >> ./autoload_configs/switch.conf.xml >> >> >> >> >> But, anyway, FS tries dynamically assign ports for each income call, and >> lost 10s every time. >> >> Please suggest, how to disable dynamic port mapping, and use static port >> range mapping (20000-25000) only? >> I assume, this will increase speedup of my system. >> >> Thanks, >> >> Oleg >> >> >> >> 2011-04-22 18:14:45.590723 [DEBUG] switch_channel.c:2813 >> (sofia/internal/sip:1001 at 192.168.1.151:5060) Callstate Change RINGING -> >> ACTIVE >> >> . . . . >> 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1116 Send >> signal sofia/internal/sip:1000 at 192.168.1.152:5060 [BREAK] >> 2011-04-22 18:14:45.750984 [DEBUG] switch_core_session.c:1288 Session 57 >> (sofia/internal/sip:1000 at 192.168.1.152:5060) Locked, Waiting on external >> entities >> <5s delay> >> 2011-04-22 18:14:50.018621 [DEBUG] sofia_reg.c:1765 Changing expire time >> to 109 by request of proxy sip:callcentric.com >> 2011-04-22 18:14:50.209565 [DEBUG] switch_nat.c:502 mapped public port >> 20794 protocol UDP to localport 20794 >> 2011-04-22 18:14:50.341416 [DEBUG] sofia_reg.c:1765 Changing expire time >> to 104 by request of proxy sip:callcentric.com >> <5s delay> >> 2011-04-22 18:14:55.213775 [DEBUG] switch_nat.c:502 mapped public port >> 20795 protocol UDP to localport 20795 >> 2011-04-22 18:14:55.213775 [DEBUG] sofia_glue.c:3001 AUDIO RTP >> [sofia/external/2404760839 at 66.54.140.46] 192.168.1.5 port 20794 -> >> 204.11.192.22 port 60140 codec: 0 ms: 20 >> 2011-04-22 18:14:55.213775 [DEBUG] switch_rtp.c:1623 Starting timer >> [soft] 160 bytes per 20ms >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > From jcherukuri_necc at yahoo.com Tue Apr 26 18:16:17 2011 From: jcherukuri_necc at yahoo.com (Jyotshna Cherukuri) Date: Tue, 26 Apr 2011 07:16:17 -0700 (PDT) Subject: [Freeswitch-dev] One way audio problem from B-leg of the call on Session Refresh and HOLD if its a dynamic payload type Message-ID: <1905.13225.qm@web110306.mail.gq1.yahoo.com> Hi , I am working with the latest revision of Freeswitch and I am having one way audio problem using SILK/8000 codec on B-leg of the call after Session Refresh or on HOLD. This is due to the fact that when FS sends an INVITE out to B-leg it sends 98 in its SDP offer and B-leg responds back with same code but with payload type "120" [ Offer ] v=0 o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 s=FreeSWITCH c=IN IP4 192.168.2.10 t=0 0 m=audio 33766 RTP/AVP 98 9 101 a=rtpmap:98 SILK/8000 a=fmtp:98 useinbandfec=1;usedtx=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 [ Answer ] v=0 o=- 3512815772 3512815773 IN IP4 192.168.4.121 s=pjmedia c=IN IP4 192.168.4.121 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 120 96 a=rtcp:4003 IN IP4 192.168.4.121 a=rtpmap:120 silk/8000 a=fmtp:120 useinbandfec=1;usedtx=0 a=sendrecv a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 Freeswitch handles this properly on the initial offer/answer as its using this patch (tell rtp stack about what remote payload type to expect when the receiving end follows the stupid SHOULD as WONT and sends a different dynamic payload number than the one in the offer) After one min into the call because Session Timers are enabled Freeswitch sends a Session Refresh with payload type now setting to "120" [Refresh Offer] v=0 o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 s=FreeSWITCH c=IN IP4 192.168.2.10 t=0 0 m=audio 33766 RTP/AVP 120 96 9 a=rtpmap:120 SILK/8000 a=fmtp:120 useinbandfec=1;usedtx=0 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 The remote end then starts sending RTP packets with payload number "120" in its RTP header and FS stops processing these packets and as a result is resulting in one-way audio issue. Any help is appreciated. Thanks in advance Regards Jyotshna P.S : The issue starts even when the remote end presses" HOLD" as it sends INVITE on hold with "120" and FS responds back with "120" in its answer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110426/ad296d5c/attachment.html From steveayre at gmail.com Tue Apr 26 19:02:09 2011 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 26 Apr 2011 16:02:09 +0100 Subject: [Freeswitch-dev] One way audio problem from B-leg of the call on Session Refresh and HOLD if its a dynamic payload type In-Reply-To: <1905.13225.qm@web110306.mail.gq1.yahoo.com> References: <1905.13225.qm@web110306.mail.gq1.yahoo.com> Message-ID: Open a jira for it so that the bug can be registered and tracked. http://jira.freeswitch.org/ -Steve On 26 April 2011 15:16, Jyotshna Cherukuri wrote: > Hi , > > I am working with the latest revision of Freeswitch and I am having one way > audio problem using SILK/8000 codec on B-leg of the call after Session > Refresh or on HOLD. This is due to the fact that when FS sends an INVITE out > to B-leg it sends 98 in its SDP offer and B-leg responds back with same code > but with payload type "120" > > [ Offer ] > v=0 > o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 > s=FreeSWITCH > c=IN IP4 192.168.2.10 > t=0 0 > m=audio 33766 RTP/AVP 98 9 101 > a=rtpmap:98 SILK/8000 > a=fmtp:98 useinbandfec=1;usedtx=0 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > [ Answer ] > > v=0 > o=- 3512815772 3512815773 IN IP4 192.168.4.121 > s=pjmedia > c=IN IP4 192.168.4.121 > t=0 0 > a=X-nat:0 > m=audio 4002 RTP/AVP 120 96 > a=rtcp:4003 IN IP4 192.168.4.121 > a=rtpmap:120 silk/8000 > a=fmtp:120 useinbandfec=1;usedtx=0 > a=sendrecv > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-15 > > Freeswitch handles this properly on the initial offer/answer as its using > this patch (tell rtp stack about what remote payload type to expect when > the receiving end follows the stupid SHOULD as WONT and sends a different > dynamic payload number than the one in the offer) > > After one min into the call because Session Timers are enabled Freeswitch > sends a Session Refresh with payload type now setting to "120" > > [Refresh Offer] > v=0 > o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 > s=FreeSWITCH > c=IN IP4 192.168.2.10 > t=0 0 > m=audio 33766 RTP/AVP 120 96 9 > a=rtpmap:120 SILK/8000 > a=fmtp:120 useinbandfec=1;usedtx=0 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > The remote end then starts sending RTP packets with payload number "120" in > its RTP header and FS stops processing these packets and as a result is > resulting in one-way audio issue. > > Any help is appreciated. > > Thanks in advance > Regards > Jyotshna > > > P.S : The issue starts even when the remote end presses" HOLD" as it sends > INVITE on hold with "120" and FS responds back with "120" in its answer. > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110426/3ac7bbb7/attachment.html From anthony.minessale at gmail.com Tue Apr 26 19:13:06 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 26 Apr 2011 10:13:06 -0500 Subject: [Freeswitch-dev] One way audio problem from B-leg of the call on Session Refresh and HOLD if its a dynamic payload type In-Reply-To: References: <1905.13225.qm@web110306.mail.gq1.yahoo.com> Message-ID: and when you do be sure to attach a full console log with sofia global siptrace on enabled. On Tue, Apr 26, 2011 at 10:02 AM, Steven Ayre wrote: > Open a jira for it so that the bug can be registered and tracked. > > http://jira.freeswitch.org/ > > -Steve > > > > On 26 April 2011 15:16, Jyotshna Cherukuri > wrote: >> >> Hi , >> I am working with the latest revision of Freeswitch and I am having one >> way audio problem ?using SILK/8000 codec??on B-leg of the call?after Session >> Refresh or on HOLD. This is due to the fact that when FS sends an INVITE out >> to B-leg it sends 98 in its SDP offer and B-leg responds back with same code >> but with payload type "120" >> [ Offer ] >> v=0 >> ?? o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 >> ?? s=FreeSWITCH >> ?? c=IN IP4 192.168.2.10 >> ?? t=0 0 >> ?? m=audio 33766 RTP/AVP 98 9 101 >> ?? a=rtpmap:98 SILK/8000 >> ?? a=fmtp:98 useinbandfec=1;usedtx=0 >> ?? a=rtpmap:101 telephone-event/8000 >> ?? a=fmtp:101 0-16 >> ?? a=silenceSupp:off - - - - >> ?? a=ptime:20 >> >> [ Answer ] >> ?v=0 >> ?? o=- 3512815772 3512815773 IN IP4 192.168.4.121 >> ?? s=pjmedia >> ?? c=IN IP4 192.168.4.121 >> ?? t=0 0 >> ?? a=X-nat:0 >> ?? m=audio 4002 RTP/AVP 120 96 >> ?? a=rtcp:4003 IN IP4 192.168.4.121 >> ?? a=rtpmap:120 silk/8000 >> ?? a=fmtp:120 useinbandfec=1;usedtx=0 >> ?? a=sendrecv >> ?? a=rtpmap:96 telephone-event/8000 >> ?? a=fmtp:96 0-15 >> Freeswitch handles this properly on the initial offer/answer as its using >> this patch (tell rtp stack about what remote payload type to expect when the >> receiving end follows the stupid SHOULD as WONT and sends a different >> dynamic payload number than the one in the offer) >> After one min into the call because Session Timers are enabled Freeswitch >> sends a Session Refresh with payload type now setting to "120" >> [Refresh Offer] >> v=0 >> ?? o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10 >> ?? s=FreeSWITCH >> ?? c=IN IP4 192.168.2.10 >> ?? t=0 0 >> ?? m=audio 33766 RTP/AVP 120 96 9 >> ?? a=rtpmap:120 SILK/8000 >> ?? a=fmtp:120 useinbandfec=1;usedtx=0 >> ?? a=rtpmap:96 telephone-event/8000 >> ?? a=fmtp:96 0-16 >> ?? a=silenceSupp:off - - - - >> ?? a=ptime:20 >> The remote end then starts sending RTP packets with payload number "120" >> in its RTP header and FS stops processing these packets and as a result is >> resulting in one-way audio issue. >> Any help is appreciated. >> Thanks in advance >> Regards >> Jyotshna >> >> P.S : ?The issue starts even when the remote end presses" HOLD" as it >> sends INVITE on hold with "120" and FS responds back with "120" in its >> answer. >> >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed Apr 27 20:29:30 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Apr 2011 09:29:30 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today - Safi Systems IVR Builder Message-ID: Hello all! We are happy to have Zac Wolfe from Safi Systems joining us today to talk about the IVR building system they have created. There is a gotomeeting.comlink on the agenda page here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_27 We look forward to this discussion! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110427/311a46ec/attachment.html From jmesquita at freeswitch.org Fri Apr 29 19:50:51 2011 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 29 Apr 2011 12:50:51 -0300 Subject: [Freeswitch-dev] mod_voicemail new API functions Message-ID: Guys, I see that Moc has added new API functions to set preferences to the mailbox. Those work great and are awesome, but I think there is one small feature missing. Shouldn't we be able to remove the greetings instead of just setting new ones so that mod_voicemail restores it's original configuration? I don't have any problem hacking in the feature, I am just making sure I am not going to do something that is already in there and that I have overlooked before. Thanks! Jo?o Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110429/941db304/attachment.html From msc at freeswitch.org Fri Apr 29 21:42:01 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 29 Apr 2011 10:42:01 -0700 Subject: [Freeswitch-dev] Reviewers needed: FreeSWITCH Cookbook Message-ID: Hello, As you know we are working on a FreeSWITCH Cookbook for Packt Publishing. We are in need of 3 or 4 capable reviewers. Please contact me off list if you want to help and meet these qualifications: * Read and write English fluently * Have at least one instance of FreeSWITCH (or a sandbox) where you can test each recipe * Have time to read and try all the recipes in the book * Can offer useful feedback and proofreading * Preferably have a later version of MS Word for editing If you meet the above qualifications and would like to help out then please contact me off list for more information. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110429/86cd143d/attachment-0001.html From gmaruzz at gmail.com Sat Apr 30 12:31:39 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 30 Apr 2011 10:31:39 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute Message-ID: Dear FreeSWITCHers, after a fair amount of effort, I ended up with a new way to install and use mod_skypopen on Linux. No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. First, we can use the readily available Skype client for OSS. (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS driver, that's very easy to compile and install, and do not need to mess with the operating system installation.) Second, I wrote an installer that automatically do all the tedious work for you: download and install the skype client, create the config directory for Skype clients, create the config file for mod_skypopen, create the script that launches the Skype clients. I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. Actually is ludicrously simple now, and after you compile FreeSWITCH, mod_skypopen and the skypopen.ko OSS driver it will take like less than one minute to have a complete installation of mod_skypopen ready to make and receive calls. All automatic, no more need to fiddle around with the Skype client download, configurations, authorization, etc. Is all well tested, but maybe there are still some bugs, and maybe the docs are not clear/easy enough. Please have a look at the new and improved wiki page and let me know what do you think about (and maybe test the procedures). http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR You must update to the latest git to have all the goodies. Thank you all for your support, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From daniel.neubert at solomo.de Sat Apr 30 16:28:21 2011 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Sat, 30 Apr 2011 14:28:21 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute In-Reply-To: References: Message-ID: <4DBC0065.1060206@solomo.de> Thanks for your great work! I've been playing around with skypopen in my spare time for a few days now. Every time I send a call to the skypopen endpoint, it fails with 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364 (skypopen/skype101/MySkypeUser) State ROUTING going to sleep 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325 (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747 [32b8f10|9350fb9] [DEBUG_SKYPE 747 ][skype101 ][IDLE,IDLE] skype101 CHANNEL CONSUME_MEDIA 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173 [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][IDLE,IDLE] READING: |||CALL 39 STATUS UNPLACED||| 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714 [32b8f10|9350fb9] [DEBUG_SKYPE 714 ][skype101 ][DIALING,UNPLACD] skype_call: 39 is now UNPLACED 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,UNPLACD] READING: |||CALL 39 STATUS ROUTING||| 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709 [32b8f10|9350fb9] [DEBUG_SKYPE 709 ][skype101 ][DIALING,ROUTING] skype_call: 39 is now ROUTING 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] READING: |||CALL 39 FAILUREREASON 7||| 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540 [32b8f10|9350fb9] [DEBUG_SKYPE 540 ][skype101 ][DIALING,ROUTING] Skype FAILED on skype_call 39. Let's wait for the FAILED message. 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] READING: |||CALL 39 VAA_INPUT_STATUS FALSE||| 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] READING: |||CALL 39 STATUS FAILED||| 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683 [32b8f10|9350fb9] [DEBUG_SKYPE 683 ][skype101 ][DIALING,FAILED] we tried to call Skype on skype_call 39 and Skype has now FAILED 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413 [32b8f10|9350fb9] [DEBUG_SKYPE 1413 ][skype101 ][DOWN,FAILED] skype call ended 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572 (skypopen/skype101/MySkypeUser) Callstate Change RINGING -> HANGUP I assume that FAILUREREASON 7 indicates an issue regarding the audio interface, correct? I've tried to modify these values (tried 0,1 and 2 (which was default) ) - but did not change anything. 2 2 2 Could you give me a hint? Best regards / Mit freundlichen Gr??en, Daniel Neubert On 30.04.2011 10:31, Giovanni Maruzzelli wrote: > Dear FreeSWITCHers, > > after a fair amount of effort, I ended up with a new way to install > and use mod_skypopen on Linux. > > No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. > > First, we can use the readily available Skype client for OSS. > > (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS > driver, that's very easy to compile and install, and do not need to > mess with the operating system installation.) > > Second, I wrote an installer that automatically do all the tedious > work for you: download and install the skype client, create the config > directory for Skype clients, create the config file for mod_skypopen, > create the script that launches the Skype clients. > > I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. > > Actually is ludicrously simple now, and after you compile FreeSWITCH, > mod_skypopen and the skypopen.ko OSS driver it will take like less > than one minute to have a complete installation of mod_skypopen ready > to make and receive calls. > > All automatic, no more need to fiddle around with the Skype client > download, configurations, authorization, etc. > > Is all well tested, but maybe there are still some bugs, and maybe the > docs are not clear/easy enough. > > Please have a look at the new and improved wiki page and let me know > what do you think about (and maybe test the procedures). > > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR > > You must update to the latest git to have all the goodies. > > Thank you all for your support, > > -giovanni > From daniel.neubert at solomo.de Sat Apr 30 16:43:22 2011 From: daniel.neubert at solomo.de (Daniel Neubert) Date: Sat, 30 Apr 2011 14:43:22 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute In-Reply-To: <4DBC0065.1060206@solomo.de> References: <4DBC0065.1060206@solomo.de> Message-ID: <4DBC03EA.6050902@solomo.de> Looks like the issue is located in building the skypopen.ko module: make -C /lib/modules/2.6.38-8-generic/build M=/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss LDDINC=/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/../include modules make[1]: Betrete Verzeichnis '/usr/src/linux-headers-2.6.38-8-generic' CC [M] /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.o /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.c:315:2: error: unknown field ?ioctl? specified in initializer /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.c:315:2: warning: initialization from incompatible pointer type make[2]: *** [/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss/main.o] Fehler 1 make[1]: *** [_module_/usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss] Fehler 2 make[1]: Verlasse Verzeichnis '/usr/src/linux-headers-2.6.38-8-generic' make: *** [modules] Fehler 2 I guess using kernel 2.6.38-8 is not supported? 2.6.38-8-generic #42-Ubuntu SMP Mon Apr 11 03:31:50 UTC 2011 i686 i686 i386 GNU/Linux Best regards / Mit freundlichen Gr??en, Daniel Neubert On 30.04.2011 14:28, Daniel Neubert wrote: > Thanks for your great work! I've been playing around with skypopen in my > spare time for a few days now. > > Every time I send a call to the skypopen endpoint, it fails with > > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364 > (skypopen/skype101/MySkypeUser) State ROUTING going to sleep > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325 > (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 > (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA > 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747 > [32b8f10|9350fb9] [DEBUG_SKYPE 747 ][skype101 ][IDLE,IDLE] > skype101 CHANNEL CONSUME_MEDIA > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 > (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep > 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][IDLE,IDLE] > READING: |||CALL 39 STATUS UNPLACED||| > 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714 > [32b8f10|9350fb9] [DEBUG_SKYPE 714 ][skype101 ][DIALING,UNPLACD] > skype_call: 39 is now UNPLACED > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,UNPLACD] > READING: |||CALL 39 STATUS ROUTING||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709 > [32b8f10|9350fb9] [DEBUG_SKYPE 709 ][skype101 ][DIALING,ROUTING] > skype_call: 39 is now ROUTING > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] > READING: |||CALL 39 FAILUREREASON 7||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540 > [32b8f10|9350fb9] [DEBUG_SKYPE 540 ][skype101 ][DIALING,ROUTING] > Skype FAILED on skype_call 39. Let's wait for the FAILED message. > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] > READING: |||CALL 39 VAA_INPUT_STATUS FALSE||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] > READING: |||CALL 39 STATUS FAILED||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683 > [32b8f10|9350fb9] [DEBUG_SKYPE 683 ][skype101 ][DIALING,FAILED] > we tried to call Skype on skype_call 39 and Skype has now FAILED > 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413 > [32b8f10|9350fb9] [DEBUG_SKYPE 1413 ][skype101 ][DOWN,FAILED] > skype call ended > 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572 > (skypopen/skype101/MySkypeUser) Callstate Change RINGING -> HANGUP > > I assume that FAILUREREASON 7 indicates an issue regarding the audio > interface, correct? > > I've tried to modify these values (tried 0,1 and 2 (which was default) ) > - but did not change anything. > > 2 > 2 > 2 > > Could you give me a hint? > > Best regards / Mit freundlichen Gr??en, > Daniel Neubert > > On 30.04.2011 10:31, Giovanni Maruzzelli wrote: >> Dear FreeSWITCHers, >> >> after a fair amount of effort, I ended up with a new way to install >> and use mod_skypopen on Linux. >> >> No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. >> >> First, we can use the readily available Skype client for OSS. >> >> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS >> driver, that's very easy to compile and install, and do not need to >> mess with the operating system installation.) >> >> Second, I wrote an installer that automatically do all the tedious >> work for you: download and install the skype client, create the config >> directory for Skype clients, create the config file for mod_skypopen, >> create the script that launches the Skype clients. >> >> I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. >> >> Actually is ludicrously simple now, and after you compile FreeSWITCH, >> mod_skypopen and the skypopen.ko OSS driver it will take like less >> than one minute to have a complete installation of mod_skypopen ready >> to make and receive calls. >> >> All automatic, no more need to fiddle around with the Skype client >> download, configurations, authorization, etc. >> >> Is all well tested, but maybe there are still some bugs, and maybe the >> docs are not clear/easy enough. >> >> Please have a look at the new and improved wiki page and let me know >> what do you think about (and maybe test the procedures). >> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR >> >> You must update to the latest git to have all the goodies. >> >> Thank you all for your support, >> >> -giovanni >> From gmaruzz at celliax.org Sat Apr 30 16:50:02 2011 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 30 Apr 2011 14:50:02 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute In-Reply-To: <4DBC0065.1060206@solomo.de> References: <4DBC0065.1060206@solomo.de> Message-ID: 1) is very very bad to answer a mailing list post with something unrelated 2) use the new installer and the latest git to avoid any such problem 3) if you want or need to use the old way, refer to the old wikipage for the perfect install (first row in wiki page tell you where the old page is) 4) anyway, for bug, issues, etc, open a jira issue on http://jira.freeswitch.org Have a nice weekend and don't hiijack the threads ;) -giovanni On 4/30/11, Daniel Neubert wrote: > Thanks for your great work! I've been playing around with skypopen in my > spare time for a few days now. > > Every time I send a call to the skypopen endpoint, it fails with > > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364 > (skypopen/skype101/MySkypeUser) State ROUTING going to sleep > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325 > (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 > (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA > 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747 > [32b8f10|9350fb9] [DEBUG_SKYPE 747 ][skype101 ][IDLE,IDLE] > skype101 CHANNEL CONSUME_MEDIA > 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 > (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep > 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][IDLE,IDLE] > READING: |||CALL 39 STATUS UNPLACED||| > 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714 > [32b8f10|9350fb9] [DEBUG_SKYPE 714 ][skype101 ][DIALING,UNPLACD] > skype_call: 39 is now UNPLACED > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,UNPLACD] > READING: |||CALL 39 STATUS ROUTING||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709 > [32b8f10|9350fb9] [DEBUG_SKYPE 709 ][skype101 ][DIALING,ROUTING] > skype_call: 39 is now ROUTING > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] > READING: |||CALL 39 FAILUREREASON 7||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540 > [32b8f10|9350fb9] [DEBUG_SKYPE 540 ][skype101 ][DIALING,ROUTING] > Skype FAILED on skype_call 39. Let's wait for the FAILED message. > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] > READING: |||CALL 39 VAA_INPUT_STATUS FALSE||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 > [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] > READING: |||CALL 39 STATUS FAILED||| > 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683 > [32b8f10|9350fb9] [DEBUG_SKYPE 683 ][skype101 ][DIALING,FAILED] > we tried to call Skype on skype_call 39 and Skype has now FAILED > 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413 > [32b8f10|9350fb9] [DEBUG_SKYPE 1413 ][skype101 ][DOWN,FAILED] > skype call ended > 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572 > (skypopen/skype101/MySkypeUser) Callstate Change RINGING -> HANGUP > > I assume that FAILUREREASON 7 indicates an issue regarding the audio > interface, correct? > > I've tried to modify these values (tried 0,1 and 2 (which was default) ) > - but did not change anything. > > 2 > 2 > 2 > > Could you give me a hint? > > Best regards / Mit freundlichen Gr??en, > Daniel Neubert > > On 30.04.2011 10:31, Giovanni Maruzzelli wrote: >> Dear FreeSWITCHers, >> >> after a fair amount of effort, I ended up with a new way to install >> and use mod_skypopen on Linux. >> >> No more looking around the internet for the lost 2.0.0.72 Skype client for >> ALSA. >> >> First, we can use the readily available Skype client for OSS. >> >> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS >> driver, that's very easy to compile and install, and do not need to >> mess with the operating system installation.) >> >> Second, I wrote an installer that automatically do all the tedious >> work for you: download and install the skype client, create the config >> directory for Skype clients, create the config file for mod_skypopen, >> create the script that launches the Skype clients. >> >> I hope those improvements will lower the barriers for Skype calls on >> FreeSWITCH. >> >> Actually is ludicrously simple now, and after you compile FreeSWITCH, >> mod_skypopen and the skypopen.ko OSS driver it will take like less >> than one minute to have a complete installation of mod_skypopen ready >> to make and receive calls. >> >> All automatic, no more need to fiddle around with the Skype client >> download, configurations, authorization, etc. >> >> Is all well tested, but maybe there are still some bugs, and maybe the >> docs are not clear/easy enough. >> >> Please have a look at the new and improved wiki page and let me know >> what do you think about (and maybe test the procedures). >> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR >> >> You must update to the latest git to have all the goodies. >> >> Thank you all for your support, >> >> -giovanni >> > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Sat Apr 30 16:54:40 2011 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 30 Apr 2011 14:54:40 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute In-Reply-To: References: <4DBC0065.1060206@solomo.de> Message-ID: The supported linux distro are listed in the wiki page: centos 5.x and 6.x, Ubuntu 10.04. Maybe for newer or custom kernels you have to slightly modify the OSS driver code. If you do modify it successfully, please send a patch via http://jira.freeswitch.org Thanks in advance, -giovanni On 4/30/11, Giovanni Maruzzelli wrote: > 1) is very very bad to answer a mailing list post with something unrelated > 2) use the new installer and the latest git to avoid any such problem > 3) if you want or need to use the old way, refer to the old wikipage > for the perfect install (first row in wiki page tell you where the old > page is) > 4) anyway, for bug, issues, etc, open a jira issue on > http://jira.freeswitch.org > > Have a nice weekend and don't hiijack the threads ;) > > -giovanni > > > > On 4/30/11, Daniel Neubert wrote: >> Thanks for your great work! I've been playing around with skypopen in my >> spare time for a few days now. >> >> Every time I send a call to the skypopen endpoint, it fails with >> >> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364 >> (skypopen/skype101/MySkypeUser) State ROUTING going to sleep >> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325 >> (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA >> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 >> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA >> 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747 >> [32b8f10|9350fb9] [DEBUG_SKYPE 747 ][skype101 ][IDLE,IDLE] >> skype101 CHANNEL CONSUME_MEDIA >> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 >> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep >> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173 >> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][IDLE,IDLE] >> READING: |||CALL 39 STATUS UNPLACED||| >> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714 >> [32b8f10|9350fb9] [DEBUG_SKYPE 714 ][skype101 ][DIALING,UNPLACD] >> skype_call: 39 is now UNPLACED >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,UNPLACD] >> READING: |||CALL 39 STATUS ROUTING||| >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709 >> [32b8f10|9350fb9] [DEBUG_SKYPE 709 ][skype101 ][DIALING,ROUTING] >> skype_call: 39 is now ROUTING >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] >> READING: |||CALL 39 FAILUREREASON 7||| >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540 >> [32b8f10|9350fb9] [DEBUG_SKYPE 540 ][skype101 ][DIALING,ROUTING] >> Skype FAILED on skype_call 39. Let's wait for the FAILED message. >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] >> READING: |||CALL 39 VAA_INPUT_STATUS FALSE||| >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] >> READING: |||CALL 39 STATUS FAILED||| >> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683 >> [32b8f10|9350fb9] [DEBUG_SKYPE 683 ][skype101 ][DIALING,FAILED] >> we tried to call Skype on skype_call 39 and Skype has now FAILED >> 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413 >> [32b8f10|9350fb9] [DEBUG_SKYPE 1413 ][skype101 ][DOWN,FAILED] >> skype call ended >> 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572 >> (skypopen/skype101/MySkypeUser) Callstate Change RINGING -> HANGUP >> >> I assume that FAILUREREASON 7 indicates an issue regarding the audio >> interface, correct? >> >> I've tried to modify these values (tried 0,1 and 2 (which was default) ) >> - but did not change anything. >> >> 2 >> 2 >> 2 >> >> Could you give me a hint? >> >> Best regards / Mit freundlichen Gr??en, >> Daniel Neubert >> >> On 30.04.2011 10:31, Giovanni Maruzzelli wrote: >>> Dear FreeSWITCHers, >>> >>> after a fair amount of effort, I ended up with a new way to install >>> and use mod_skypopen on Linux. >>> >>> No more looking around the internet for the lost 2.0.0.72 Skype client >>> for >>> ALSA. >>> >>> First, we can use the readily available Skype client for OSS. >>> >>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS >>> driver, that's very easy to compile and install, and do not need to >>> mess with the operating system installation.) >>> >>> Second, I wrote an installer that automatically do all the tedious >>> work for you: download and install the skype client, create the config >>> directory for Skype clients, create the config file for mod_skypopen, >>> create the script that launches the Skype clients. >>> >>> I hope those improvements will lower the barriers for Skype calls on >>> FreeSWITCH. >>> >>> Actually is ludicrously simple now, and after you compile FreeSWITCH, >>> mod_skypopen and the skypopen.ko OSS driver it will take like less >>> than one minute to have a complete installation of mod_skypopen ready >>> to make and receive calls. >>> >>> All automatic, no more need to fiddle around with the Skype client >>> download, configurations, authorization, etc. >>> >>> Is all well tested, but maybe there are still some bugs, and maybe the >>> docs are not clear/easy enough. >>> >>> Please have a look at the new and improved wiki page and let me know >>> what do you think about (and maybe test the procedures). >>> >>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk >>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux >>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR >>> >>> You must update to the latest git to have all the goodies. >>> >>> Thank you all for your support, >>> >>> -giovanni >>> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -- > Sent from my mobile device > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Sat Apr 30 17:02:41 2011 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 30 Apr 2011 15:02:41 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute In-Reply-To: References: <4DBC0065.1060206@solomo.de> Message-ID: Also, don't forget to install all the required packages listed in the wikipage (eg: kernel headers, it seems not finding include files). Btw, I see you're on 32 bit. Never tested on 32 bit. Maybe there is a problem with 32 bit? Please open a Jira if you can't find a solution, or file a jira with the solution you found. Thanks again, -giovanni On 4/30/11, Giovanni Maruzzelli wrote: > The supported linux distro are listed in the wiki page: centos 5.x and > 6.x, Ubuntu 10.04. > Maybe for newer or custom kernels you have to slightly modify the OSS > driver code. > If you do modify it successfully, please send a patch via > http://jira.freeswitch.org > > Thanks in advance, > -giovanni > > On 4/30/11, Giovanni Maruzzelli wrote: >> 1) is very very bad to answer a mailing list post with something >> unrelated >> 2) use the new installer and the latest git to avoid any such problem >> 3) if you want or need to use the old way, refer to the old wikipage >> for the perfect install (first row in wiki page tell you where the old >> page is) >> 4) anyway, for bug, issues, etc, open a jira issue on >> http://jira.freeswitch.org >> >> Have a nice weekend and don't hiijack the threads ;) >> >> -giovanni >> >> >> >> On 4/30/11, Daniel Neubert wrote: >>> Thanks for your great work! I've been playing around with skypopen in my >>> spare time for a few days now. >>> >>> Every time I send a call to the skypopen endpoint, it fails with >>> >>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:364 >>> (skypopen/skype101/MySkypeUser) State ROUTING going to sleep >>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:325 >>> (skypopen/skype101/MySkypeUser) Running State Change CS_CONSUME_MEDIA >>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 >>> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA >>> 2011-04-30 14:15:53.532923 [DEBUG] mod_skypopen.c:747 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 747 ][skype101 ][IDLE,IDLE] >>> skype101 CHANNEL CONSUME_MEDIA >>> 2011-04-30 14:15:53.532923 [DEBUG] switch_core_state_machine.c:383 >>> (skypopen/skype101/MySkypeUser) State CONSUME_MEDIA going to sleep >>> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:173 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][IDLE,IDLE] >>> READING: |||CALL 39 STATUS UNPLACED||| >>> 2011-04-30 14:15:53.532923 [DEBUG] skypopen_protocol.c:714 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 714 ][skype101 ][DIALING,UNPLACD] >>> skype_call: 39 is now UNPLACED >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,UNPLACD] >>> READING: |||CALL 39 STATUS ROUTING||| >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:709 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 709 ][skype101 ][DIALING,ROUTING] >>> skype_call: 39 is now ROUTING >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] >>> READING: |||CALL 39 FAILUREREASON 7||| >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:540 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 540 ][skype101 ][DIALING,ROUTING] >>> Skype FAILED on skype_call 39. Let's wait for the FAILED message. >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] >>> READING: |||CALL 39 VAA_INPUT_STATUS FALSE||| >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:173 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] >>> READING: |||CALL 39 STATUS FAILED||| >>> 2011-04-30 14:15:53.612936 [DEBUG] skypopen_protocol.c:683 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 683 ][skype101 ][DIALING,FAILED] >>> we tried to call Skype on skype_call 39 and Skype has now FAILED >>> 2011-04-30 14:15:53.612936 [DEBUG] mod_skypopen.c:1413 >>> [32b8f10|9350fb9] [DEBUG_SKYPE 1413 ][skype101 ][DOWN,FAILED] >>> skype call ended >>> 2011-04-30 14:15:53.612936 [DEBUG] switch_channel.c:2572 >>> (skypopen/skype101/MySkypeUser) Callstate Change RINGING -> HANGUP >>> >>> I assume that FAILUREREASON 7 indicates an issue regarding the audio >>> interface, correct? >>> >>> I've tried to modify these values (tried 0,1 and 2 (which was default) ) >>> - but did not change anything. >>> >>> 2 >>> 2 >>> 2 >>> >>> Could you give me a hint? >>> >>> Best regards / Mit freundlichen Gr??en, >>> Daniel Neubert >>> >>> On 30.04.2011 10:31, Giovanni Maruzzelli wrote: >>>> Dear FreeSWITCHers, >>>> >>>> after a fair amount of effort, I ended up with a new way to install >>>> and use mod_skypopen on Linux. >>>> >>>> No more looking around the internet for the lost 2.0.0.72 Skype client >>>> for >>>> ALSA. >>>> >>>> First, we can use the readily available Skype client for OSS. >>>> >>>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS >>>> driver, that's very easy to compile and install, and do not need to >>>> mess with the operating system installation.) >>>> >>>> Second, I wrote an installer that automatically do all the tedious >>>> work for you: download and install the skype client, create the config >>>> directory for Skype clients, create the config file for mod_skypopen, >>>> create the script that launches the Skype clients. >>>> >>>> I hope those improvements will lower the barriers for Skype calls on >>>> FreeSWITCH. >>>> >>>> Actually is ludicrously simple now, and after you compile FreeSWITCH, >>>> mod_skypopen and the skypopen.ko OSS driver it will take like less >>>> than one minute to have a complete installation of mod_skypopen ready >>>> to make and receive calls. >>>> >>>> All automatic, no more need to fiddle around with the Skype client >>>> download, configurations, authorization, etc. >>>> >>>> Is all well tested, but maybe there are still some bugs, and maybe the >>>> docs are not clear/easy enough. >>>> >>>> Please have a look at the new and improved wiki page and let me know >>>> what do you think about (and maybe test the procedures). >>>> >>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk >>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux >>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR >>>> >>>> You must update to the latest git to have all the goodies. >>>> >>>> Thank you all for your support, >>>> >>>> -giovanni >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >> >> -- >> Sent from my mobile device >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> > > -- > Sent from my mobile device > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tayeb.meftah at gmail.com Sat Apr 30 23:01:34 2011 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 30 Apr 2011 21:01:34 +0200 Subject: [Freeswitch-dev] NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute In-Reply-To: References: Message-ID: <4DBC5C8E.6080506@gmail.com> thank you Giovanni for your facilitys you did a very great Job On 30/04/2011 10:31, Giovanni Maruzzelli wrote: > Dear FreeSWITCHers, > > after a fair amount of effort, I ended up with a new way to install > and use mod_skypopen on Linux. > > No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. > > First, we can use the readily available Skype client for OSS. > > (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS > driver, that's very easy to compile and install, and do not need to > mess with the operating system installation.) > > Second, I wrote an installer that automatically do all the tedious > work for you: download and install the skype client, create the config > directory for Skype clients, create the config file for mod_skypopen, > create the script that launches the Skype clients. > > I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. > > Actually is ludicrously simple now, and after you compile FreeSWITCH, > mod_skypopen and the skypopen.ko OSS driver it will take like less > than one minute to have a complete installation of mod_skypopen ready > to make and receive calls. > > All automatic, no more need to fiddle around with the Skype client > download, configurations, authorization, etc. > > Is all well tested, but maybe there are still some bugs, and maybe the > docs are not clear/easy enough. > > Please have a look at the new and improved wiki page and let me know > what do you think about (and maybe test the procedures). > > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux > http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR > > You must update to the latest git to have all the goodies. > > Thank you all for your support, > > -giovanni > > -- Meftah Tayeb inum: +883510001288000 phone: +13477595883 From nsirugudi at gmail.com Wed Apr 27 08:35:08 2011 From: nsirugudi at gmail.com (Narendra Sirugudi) Date: Wed, 27 Apr 2011 10:05:08 +0530 Subject: [Freeswitch-dev] using freeswich as for SIP signalling and programming an external HW. Message-ID: Hi All, I want to use freeswich for SIP signalling and program an external HW for RTP processing. Does freeswitch provide any mechanism/hooks to program an external HW for RTP processing ? thanks, --kumar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110427/21eea12b/attachment.html