From cal.leeming at simplicitymedialtd.co.uk Tue Oct 1 00:05:47 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 30 Sep 2013 21:05:47 +0100 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: Any idea why libsofia only handles 1 thread per profile? As mentioned before, you should run multiple profiles if this becomes a bottleneck, but running multiple profiles for inbound calls might not be suitable (i.e. if a provider gives you a number and will only send traffic to 1 IP/port at a time). Are there any other alternative workarounds? Cal On Mon, Sep 30, 2013 at 8:36 PM, Guillermo Ruiz Camauer wrote: > Steven, > > Ah, I think I get it now: profiles LISTEN on an IP:PORT combination, so I > can have many IP:PORT combinations to MAKE calls, as long as I send them to > my provider's IP on the 5060 port, but I would only receive calls from my > provider on my one profile bount to the IP:5060 port. Is this correct? > > Guillermo > > > On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: > >> If I only have one Voip provider, can I still have more than one SIP >>> profile? My provider authenticates via IP >> >> >> Profiles bind to a single ip:port combination. >> >> You could run multiple profiles each on a separate port. That way they'll >> all send to the provider using the same IP. >> >> >> >> >> >> On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: >> >>> In the Wiki, under "Performance testing and configurations", one of the >>> suggestions given under "Recommended SIP Settings" is: >>> >>> libsofia only handles 1 thread per profile, so if that is your bottle >>> neck use more profiles >>> >>> If I only have one Voip provider, can I still have more than one SIP >>> profile? My provider authenticates via IP. I currently run 240 concurrent >>> calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core >>> machine. This means that one core is maxing out. How can I get more >>> threads up to distribute the load? >>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/6b678d03/attachment-0001.html From mike at jerris.com Tue Oct 1 00:10:45 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Sep 2013 16:10:45 -0400 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: <6F61532D-CBCC-4357-9DBB-111FDD8BFC53@jerris.com> This is not going to be a bottleneck? Your going down the wrong road chasing this. Mike On Sep 30, 2013, at 4:05 PM, Cal Leeming [Simplicity Media Ltd] wrote: > Any idea why libsofia only handles 1 thread per profile? > > As mentioned before, you should run multiple profiles if this becomes a bottleneck, but running multiple profiles for inbound calls might not be suitable (i.e. if a provider gives you a number and will only send traffic to 1 IP/port at a time). > > Are there any other alternative workarounds? > > Cal > > > On Mon, Sep 30, 2013 at 8:36 PM, Guillermo Ruiz Camauer wrote: > Steven, > > Ah, I think I get it now: profiles LISTEN on an IP:PORT combination, so I can have many IP:PORT combinations to MAKE calls, as long as I send them to my provider's IP on the 5060 port, but I would only receive calls from my provider on my one profile bount to the IP:5060 port. Is this correct? > > Guillermo > > > On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: > If I only have one Voip provider, can I still have more than one SIP profile? My provider authenticates via IP > > Profiles bind to a single ip:port combination. > > You could run multiple profiles each on a separate port. That way they'll all send to the provider using the same IP. > > > > > > On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: > In the Wiki, under "Performance testing and configurations", one of the suggestions given under "Recommended SIP Settings" is: > > libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles > > If I only have one Voip provider, can I still have more than one SIP profile? My provider authenticates via IP. I currently run 240 concurrent calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core machine. This means that one core is maxing out. How can I get more threads up to distribute the load? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/8f120758/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Oct 1 00:13:20 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 30 Sep 2013 21:13:20 +0100 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: Ignore my question, this has just been answered by Mike in IRC; For sake of archives; foxx[cleeming]: yes.. I know why.. thats how it was written? * AnGrYfUrBy (~AnGrYfUrB at pdpc/supporter/active/angryfurby) has joined #freeswitch that being said.. your not hitting that bottleneck and its a waste of time to track down any reason it was written like that for? is it difficult/impossible to have multi threading in that part of fs? it is multi threaded in the freeswitch part its not an issue at all.. * foxx[cleeming] has very little understanding of the fundementals in C btw, so apologies if this is a stupid question no reason to spend another second thinking about it lol okay, got it * sekil (~Ognjen at 78.24.104.82) has joined #freeswitch its just the way the library was written so the bottleneck in this case, is the time spent in parsing the udp packets rihgt? there is no bottleneck if you have a bottleneck? your doing way more traffic on one box than you should ah, so we're talking like 100mbit+ before this becomes an issue right? i got 200 sessions per second before hitting any bottleneck in sofia talking 10's of thousands of calls and > 1000 cps? if your doing this.. it should be split up between multiple boxes using a proxy doing load bal crienzo: and its been improved sense then I think too.. thats why I was saying.. silly conversation for a non issue yeah makes more sense now i think 200 is enough Cal On Mon, Sep 30, 2013 at 9:05 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Any idea why libsofia only handles 1 thread per profile? > > As mentioned before, you should run multiple profiles if this becomes a > bottleneck, but running multiple profiles for inbound calls might not be > suitable (i.e. if a provider gives you a number and will only send traffic > to 1 IP/port at a time). > > Are there any other alternative workarounds? > > Cal > > > On Mon, Sep 30, 2013 at 8:36 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> Steven, >> >> Ah, I think I get it now: profiles LISTEN on an IP:PORT combination, so >> I can have many IP:PORT combinations to MAKE calls, as long as I send them >> to my provider's IP on the 5060 port, but I would only receive calls from >> my provider on my one profile bount to the IP:5060 port. Is this correct? >> >> Guillermo >> >> >> On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: >> >>> If I only have one Voip provider, can I still have more than one SIP >>>> profile? My provider authenticates via IP >>> >>> >>> Profiles bind to a single ip:port combination. >>> >>> You could run multiple profiles each on a separate port. That way >>> they'll all send to the provider using the same IP. >>> >>> >>> >>> >>> >>> On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: >>> >>>> In the Wiki, under "Performance testing and configurations", one of >>>> the suggestions given under "Recommended SIP Settings" is: >>>> >>>> libsofia only handles 1 thread per profile, so if that is your bottle >>>> neck use more profiles >>>> >>>> If I only have one Voip provider, can I still have more than one SIP >>>> profile? My provider authenticates via IP. I currently run 240 concurrent >>>> calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core >>>> machine. This means that one core is maxing out. How can I get more >>>> threads up to distribute the load? >>>> >>>> >>>> >>>> -- >>>> Guillermo Ruiz Camauer >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/2c007550/attachment.html From ssinyagin at yahoo.com Tue Oct 1 03:09:28 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Mon, 30 Sep 2013 16:09:28 -0700 (PDT) Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: <1380582568.138.YahooMailNeo@web126201.mail.ne1.yahoo.com> This command will give you a better presentation of how each of your cores are utilized: mpstat -P ALL 1 ________________________________ From: Guillermo Ruiz Camauer To: FreeSWITCH Users Help Sent: Monday, September 30, 2013 9:26 PM Subject: Re: [Freeswitch-users] Optimizing profile Yes, TOP is giving me the 12.5% reading.? I understand that this is an average reading of all cores.? The fact that it never seems to go over 12.5% tells me that I am never using more than 1 core at a time, and that is is just the one process jumping around the different cores.? Is this a valid assumption?? I want to test using more threads to see if I a hitting a bottleneck with this.. ? Guillermo On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: If I only have one Voip provider, can I still have more than one SIP profile?? My provider authenticates via IP > > >Profiles bind to a single ip:port combination. > > >You could run multiple profiles each on a separate port. That way they'll all send to the provider using the same IP. > > > > > > > > > >On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: > >In the Wiki, under "Performance testing and configurations", one of the suggestions given under "Recommended SIP Settings" is: >>? >>libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles >> >>? >>If I only have one Voip provider, can I still have more than one SIP profile?? My provider authenticates via IP.? I currently run 240 concurrent calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core machine.? This means that one core is maxing out.? How can I get more threads up to distribute the load?? >>? >> >>-- >>Guillermo Ruiz Camauer >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Guillermo Ruiz Camauer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/c4827159/attachment-0001.html From sirimmfs at gmail.com Tue Oct 1 03:33:42 2013 From: sirimmfs at gmail.com (Siri MM) Date: Tue, 1 Oct 2013 09:33:42 +1000 Subject: [Freeswitch-users] Differentiating internal/external calls In-Reply-To: <1380528957.87347.YahooMailNeo@web126202.mail.ne1.yahoo.com> References: <1380528957.87347.YahooMailNeo@web126202.mail.ne1.yahoo.com> Message-ID: Hi Stanislav, Thanks for the reply. I am working in an open system, where the local extensions/gateways don't authenticate with Freeswitch, and hence I am forced to use the public context alone. Again, since the gateway doesn't authenticate, I am unable to use the "variables" defined within gateway settings. On Mon, Sep 30, 2013 at 6:15 PM, Stanislav Sinyagin wrote: > the scenario that you described is just ideal for two different contexts. > Why do you want to have them in one? > > Of course you can set a variable in an "inline" action in the beginning of > a context, like this: > > > > and later in the dialplan, use this variable in conditions. > Also the variable can be assigned at the SIP gateway, then incoming calls > will have it automatically: > > > > > > > > > > > > > > > > > direction="inbound" means to apply this variable for inbound calls. > > > > > ------------------------------ > *From:* Siri MM > *To:* FreeSWITCH Users Help > *Sent:* Monday, September 30, 2013 9:27 AM > *Subject:* [Freeswitch-users] Differentiating internal/external calls > > Hi, > > Is it possible for me to have two sets of dialplans - one for incoming (to > FS) and other for outgoing (from FS), in the same context? On receiving an > incoming call, I want to run through only a subset of xmls, and give up on > not hitting, and the same for outbound - is this feasible? Or any channel > variable that would help me differentiate between a external call and > internal call? > > Thanks! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/7d67b0ae/attachment.html From grcamauer at gmail.com Tue Oct 1 03:42:25 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 30 Sep 2013 20:42:25 -0300 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: <1380582568.138.YahooMailNeo@web126201.mail.ne1.yahoo.com> References: <1380582568.138.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: <3037230892926276772@unknownmsgid> I'll try that, thanks. Guillermo Sent from my iPhone On 30/09/2013, at 20:12, Stanislav Sinyagin wrote: This command will give you a better presentation of how each of your cores are utilized: mpstat -P ALL 1 ------------------------------ *From:* Guillermo Ruiz Camauer *To:* FreeSWITCH Users Help *Sent:* Monday, September 30, 2013 9:26 PM *Subject:* Re: [Freeswitch-users] Optimizing profile Yes, TOP is giving me the 12.5% reading. I understand that this is an average reading of all cores. The fact that it never seems to go over 12.5% tells me that I am never using more than 1 core at a time, and that is is just the one process jumping around the different cores. Is this a valid assumption? I want to test using more threads to see if I a hitting a bottleneck with this.. Guillermo On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: If I only have one Voip provider, can I still have more than one SIP profile? My provider authenticates via IP Profiles bind to a single ip:port combination. You could run multiple profiles each on a separate port. That way they'll all send to the provider using the same IP. On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: In the Wiki, under "Performance testing and configurations", one of the suggestions given under "Recommended SIP Settings" is: libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles If I only have one Voip provider, can I still have more than one SIP profile? My provider authenticates via IP. I currently run 240 concurrent calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core machine. This means that one core is maxing out. How can I get more threads up to distribute the load? -- Guillermo Ruiz Camauer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Guillermo Ruiz Camauer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/e515d88f/attachment.html From sirimmfs at gmail.com Tue Oct 1 04:00:12 2013 From: sirimmfs at gmail.com (Siri MM) Date: Tue, 1 Oct 2013 10:00:12 +1000 Subject: [Freeswitch-users] No ringback for calls from cell phones Message-ID: Hi, I am using FreesWITCH with Sangoma 102 and E1 connection. When I make a call from a fixed number to one of the internal extensions, ringback can be heard on the fixed number. However, when I make a call from a cell phone, although the internal extension rings, I cannot hear any ringing on the cell phone. I have tried two different mobile phone operators, but the result is the same. I have tried specifying the channel variables such as ringback and transfer_ringback int he dialplan, but doesn't help. Any pointers on how to debug this? Could it be the telco, in between the cell phone operators and FS, causing this? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/72621413/attachment-0001.html From steveayre at gmail.com Tue Oct 1 04:10:28 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 1 Oct 2013 01:10:28 +0100 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: That'd probably be correct, though it'd really depend on the provider. Certainly sending to a single port is the simplest easiest setup, and therefore the most likely. Other possibilities would be round-robin or failover distribution among a number of ip:port addresses or SIP registrations, or controlling which calls are routed to based on prefix. But you probably won't have those options. On 30 September 2013 20:36, Guillermo Ruiz Camauer wrote: > Steven, > > Ah, I think I get it now: profiles LISTEN on an IP:PORT combination, so I > can have many IP:PORT combinations to MAKE calls, as long as I send them to > my provider's IP on the 5060 port, but I would only receive calls from my > provider on my one profile bount to the IP:5060 port. Is this correct? > > Guillermo > > > On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: > >> If I only have one Voip provider, can I still have more than one SIP >>> profile? My provider authenticates via IP >> >> >> Profiles bind to a single ip:port combination. >> >> You could run multiple profiles each on a separate port. That way they'll >> all send to the provider using the same IP. >> >> >> >> >> >> On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: >> >>> In the Wiki, under "Performance testing and configurations", one of the >>> suggestions given under "Recommended SIP Settings" is: >>> >>> libsofia only handles 1 thread per profile, so if that is your bottle >>> neck use more profiles >>> >>> If I only have one Voip provider, can I still have more than one SIP >>> profile? My provider authenticates via IP. I currently run 240 concurrent >>> calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core >>> machine. This means that one core is maxing out. How can I get more >>> threads up to distribute the load? >>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/9ddf51ad/attachment.html From steveayre at gmail.com Tue Oct 1 04:15:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 1 Oct 2013 01:15:17 +0100 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: Personally I prefer htop. You'll see CPU usage broken out for each core. top itself also has a similar option - run top and press '1' On 30 September 2013 20:26, Guillermo Ruiz Camauer wrote: > Yes, TOP is giving me the 12.5% reading. I understand that this is an > average reading of all cores. The fact that it never seems to go over > 12.5% tells me that I am never using more than 1 core at a time, and that > is is just the one process jumping around the different cores. Is this a > valid assumption? I want to test using more threads to see if I a hitting > a bottleneck with this.. > > Guillermo > > > On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: > >> If I only have one Voip provider, can I still have more than one SIP >>> profile? My provider authenticates via IP >> >> >> Profiles bind to a single ip:port combination. >> >> You could run multiple profiles each on a separate port. That way they'll >> all send to the provider using the same IP. >> >> >> >> >> >> On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: >> >>> In the Wiki, under "Performance testing and configurations", one of the >>> suggestions given under "Recommended SIP Settings" is: >>> >>> libsofia only handles 1 thread per profile, so if that is your bottle >>> neck use more profiles >>> >>> If I only have one Voip provider, can I still have more than one SIP >>> profile? My provider authenticates via IP. I currently run 240 concurrent >>> calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core >>> machine. This means that one core is maxing out. How can I get more >>> threads up to distribute the load? >>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/198a2781/attachment.html From anthony.minessale at gmail.com Tue Oct 1 07:44:32 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Sep 2013 22:44:32 -0500 Subject: [Freeswitch-users] Minimal configuration for new projects In-Reply-To: <8CB7A453-581C-4E02-959D-D4A05B2E863D@freeswitch.org> References: <1380408896.91752.YahooMailNeo@web126203.mail.ne1.yahoo.com> <0B307292-1A25-4D57-B236-6394CBB734D3@freeswitch.org> <000001416709fc5f-fc6ac4fc-ad63-4b72-b4b8-b638571eced5-000000@email.amazonses.com> <1380454843.78757.YahooMailNeo@web126203.mail.ne1.yahoo.com> <8CB7A453-581C-4E02-959D-D4A05B2E863D@freeswitch.org> Message-ID: We aim to support many config sets. Work on it and submit them. On Mon, Sep 30, 2013 at 7:46 AM, Brian West wrote: > You were NOT told it wasn't interesting at all. You were told it was very > similar to the soft phone example thats included which is the simplest > config sample in tree. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > On Sep 30, 2013, at 7:34 AM, Eugene Prokopiev wrote: > > > I have another configuration from scratch, almost independent of the > > vanilla. There are no comments, default values, and optional modules. > > > > Configuration includes three user accounts, switching between them and > > uplink. The idea is to have the minimal working and browsable > > configuration in a single file, which can be extended only when > > necessary. > > > > I have already talked about this but I was told that this example is > > not interesting - http://jira.freeswitch.org/browse/FS-4874 > > > > -- > > WBR, > > Eugene Prokopiev > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/398af105/attachment-0001.html From jleung at v10networks.ca Tue Oct 1 07:53:17 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 30 Sep 2013 20:53:17 -0700 Subject: [Freeswitch-users] Minimal configuration for new projects References: <1380408896.91752.YahooMailNeo@web126203.mail.ne1.yahoo.com><0B307292-1A25-4D57-B236-6394CBB734D3@freeswitch.org><000001416709fc5f-fc6ac4fc-ad63-4b72-b4b8-b638571eced5-000000@email.amazonses.com><1380454843.78757.YahooMailNeo@web126203.mail.ne1.yahoo.com><8CB7A453-581C-4E02-959D-D4A05B2E863D@freeswitch.org> Message-ID: I remember submitting a similar one here a while ago... I'll check back on my local git repo and see if I still have my minimal PBX config... -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Anthony Minessale Sent: Mon 9/30/2013 8:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Minimal configuration for new projects We aim to support many config sets. Work on it and submit them. On Mon, Sep 30, 2013 at 7:46 AM, Brian West wrote: > You were NOT told it wasn't interesting at all. You were told it was very > similar to the soft phone example thats included which is the simplest > config sample in tree. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > On Sep 30, 2013, at 7:34 AM, Eugene Prokopiev wrote: > > > I have another configuration from scratch, almost independent of the > > vanilla. There are no comments, default values, and optional modules. > > > > Configuration includes three user accounts, switching between them and > > uplink. The idea is to have the minimal working and browsable > > configuration in a single file, which can be extended only when > > necessary. > > > > I have already talked about this but I was told that this example is > > not interesting - http://jira.freeswitch.org/browse/FS-4874 > > > > -- > > WBR, > > Eugene Prokopiev > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4758 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/9175fc7c/attachment.bin From moises.silva at gmail.com Tue Oct 1 08:05:02 2013 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 1 Oct 2013 00:05:02 -0400 Subject: [Freeswitch-users] CALL-ID In-Reply-To: References: <524932B8.6090804@softnet.si> Message-ID: On Mon, Sep 30, 2013 at 8:49 AM, Brian West wrote: > You don't, we aren't a proxy, we are a B2BUA, so each leg will have its > own call-id! > I seem to remember using the "sip_invite_call_id" variable successfully, which is handy, even for a B2BUA. You still have the From/To tags to differentiate the inbound/outbound dialogs, so there isn't, strictly-speaking, a reason for a B2BUA to not allow to pass-thru the call-id I think ... Moy *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/d7513d0d/attachment.html From richard.mace at gmail.com Tue Oct 1 09:50:40 2013 From: richard.mace at gmail.com (Richard Mace) Date: Tue, 1 Oct 2013 06:50:40 +0100 Subject: [Freeswitch-users] vars.xml Message-ID: Hi All, I see that in vars.xml there is an entry for Default Country. Where can I find out what each of the available Default Country options are please? Thanks Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/a989c155/attachment.html From ssinyagin at yahoo.com Tue Oct 1 10:34:36 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Mon, 30 Sep 2013 23:34:36 -0700 (PDT) Subject: [Freeswitch-users] Differentiating internal/external calls In-Reply-To: References: <1380528957.87347.YahooMailNeo@web126202.mail.ne1.yahoo.com> Message-ID: <1380609276.85817.YahooMailNeo@web126206.mail.ne1.yahoo.com> Siri, that's right, and you can still split the logic between contexts. For example, in public context, do the condition matching to classify your call and set a variable, say, "target_context". Then at the bottom of public context, transfer to ${target_context}. Then that target context would do the actual call processing. ________________________________ From: Siri MM To: FreeSWITCH Users Help Sent: Tuesday, October 1, 2013 1:33 AM Subject: Re: [Freeswitch-users] Differentiating internal/external calls Hi Stanislav, Thanks for the reply. I am working in an open system, where the local extensions/gateways don't authenticate with Freeswitch, and hence I am forced to use the public context alone. Again, since the gateway doesn't authenticate, I am unable to use the "variables" defined within gateway settings. On Mon, Sep 30, 2013 at 6:15 PM, Stanislav Sinyagin wrote: the scenario that you described is just ideal for two different contexts. Why do you want to have them in one? > >Of course you can set a variable in an "inline" action in the beginning of a context, like this: > >and later in the dialplan, use this variable in conditions. >Also the variable can be assigned at the SIP gateway, then incoming calls will have it automatically: > > > direction="inbound" means to apply this variable for inbound calls. > > > > > > > > >________________________________ > From: Siri MM >To: FreeSWITCH Users Help >Sent: Monday, September 30, 2013 9:27 AM >Subject: [Freeswitch-users] Differentiating internal/external calls > > > >Hi, > > >Is it possible for me to have two sets of dialplans - one for incoming (to FS) and other for outgoing (from FS), in the same context? On receiving an incoming call, I want to run through only a subset of xmls, and give up on not hitting, and the same for outbound - is this feasible? Or any channel variable that would help me differentiate between a external call and internal call? > > >Thanks! > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130930/c2806506/attachment-0001.html From peter at olssononline.se Tue Oct 1 10:38:36 2013 From: peter at olssononline.se (Peter Olsson) Date: Tue, 1 Oct 2013 08:38:36 +0200 Subject: [Freeswitch-users] lua session() is blocking... In-Reply-To: <1380558118.28284.10.camel@marces.madrid.commsmundi.com> References: <1380544273.32043.73.camel@marces.madrid.commsmundi.com> <857FEC87-7EED-4912-BE6E-89E7CEAC2C34@freeswitch.org> <1380550488.32043.151.camel@marces.madrid.commsmundi.com> <1380558118.28284.10.camel@marces.madrid.commsmundi.com> Message-ID: Yes, this is expected, that's that way it works. /Peter 2013/9/30 Antonio Silva > ** > I found it.. > > it's always blocking if no media is received... in this case the > destination reply with a SIP/180 and the session is waiting for media to > continue in the script... > > I put a pastebin where you can see the output and the lua script, > http://pastebin.freeswitch.org/21474 > > (in my minimum configuration i had the action answer before doing the > bridge... my bad...) > > > Regards, > Ant?nio > > On Mon, 2013-09-30 at 16:14 +0200, Antonio Silva wrote: > > hum... > > just try without ignore_early_media=true... same behaviour... > > then i tried with a minimum FS configuration and it work with and without > ignore_early_media=true... so it's a different problem... > > I'm trying to figure out... but thanks for the tip. > > > Regards, > Ant?nio > > > > > On Mon, 2013-09-30 at 07:48 -0500, Brian West wrote: > > You told it to do exactly that when you said ignore_early_media=true > > -- > Brian Westbrian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwesthttp://www.freeswitchbook.comhttp://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > On Sep 30, 2013, at 7:31 AM, Antonio Silva wrote: > > > The problem is that the freswitch.session is blocking and it only returns when the call is already answer... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/9ec7f6a5/attachment.html From enp at itx.ru Tue Oct 1 08:15:11 2013 From: enp at itx.ru (Eugene Prokopiev) Date: Tue, 1 Oct 2013 08:15:11 +0400 Subject: [Freeswitch-users] Minimal configuration for new projects In-Reply-To: <1380553254.22913.YahooMailNeo@web126202.mail.ne1.yahoo.com> References: <1380408896.91752.YahooMailNeo@web126203.mail.ne1.yahoo.com> <0B307292-1A25-4D57-B236-6394CBB734D3@freeswitch.org> <000001416709fc5f-fc6ac4fc-ad63-4b72-b4b8-b638571eced5-000000@email.amazonses.com> <1380454843.78757.YahooMailNeo@web126203.mail.ne1.yahoo.com> <1380553254.22913.YahooMailNeo@web126202.mail.ne1.yahoo.com> Message-ID: 2013/9/30 Stanislav Sinyagin : > looks quite extreme :) > My purpose was to have a starting template for new projects, so I preserved > the file structure, just removed unnecessary stuff. > > In what situation would you need a single file? I can't figure out a > scenario where this would be more beneficial than multiple files. One file is not goal. My goal is minimal size without default values for easier reading and use git for tracking configuration changes. Some sections can be extracted into separate files while main configuration file growth. -- WBR, Eugene Prokopiev From enp at itx.ru Tue Oct 1 08:08:19 2013 From: enp at itx.ru (Eugene Prokopiev) Date: Tue, 1 Oct 2013 08:08:19 +0400 Subject: [Freeswitch-users] Minimal configuration for new projects In-Reply-To: References: <1380408896.91752.YahooMailNeo@web126203.mail.ne1.yahoo.com> <0B307292-1A25-4D57-B236-6394CBB734D3@freeswitch.org> <000001416709fc5f-fc6ac4fc-ad63-4b72-b4b8-b638571eced5-000000@email.amazonses.com> <1380454843.78757.YahooMailNeo@web126203.mail.ne1.yahoo.com> <8CB7A453-581C-4E02-959D-D4A05B2E863D@freeswitch.org> Message-ID: 2013/10/1 Anthony Minessale : > We aim to support many config sets. Work on it and submit them. > On Mon, Sep 30, 2013 at 7:46 AM, Brian West wrote: >> You were NOT told it wasn't interesting at all. You were told it was very >> similar to the soft phone example thats included which is the simplest >> config sample in tree. The main difference with soft phone example configuration is ready for simple sip server installation. I need only sofia and dialplan sections for this task. Soft phone example configuration contains portaudio, enum and cdr sections. So, I reopened my issue. -- WBR, Eugene Prokopiev From ssinyagin at yahoo.com Tue Oct 1 11:00:09 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 1 Oct 2013 00:00:09 -0700 (PDT) Subject: [Freeswitch-users] Differentiating internal/external calls In-Reply-To: <1380609276.85817.YahooMailNeo@web126206.mail.ne1.yahoo.com> References: <1380528957.87347.YahooMailNeo@web126202.mail.ne1.yahoo.com> <1380609276.85817.YahooMailNeo@web126206.mail.ne1.yahoo.com> Message-ID: <1380610809.19307.YahooMailNeo@web126203.mail.ne1.yahoo.com> you can also do immediate transfer inside the condition, and perform in-place digit manipulation. here's an example, it returns the call to the same context, but it can also be a different context if needed: http://txlab.wordpress.com/2013/07/06/handling-e164-numbers-in-freeswitch/ ________________________________ From: Stanislav Sinyagin To: FreeSWITCH Users Help Sent: Tuesday, October 1, 2013 8:34 AM Subject: Re: [Freeswitch-users] Differentiating internal/external calls Siri, that's right, and you can still split the logic between contexts. For example, in public context, do the condition matching to classify your call and set a variable, say, "target_context". Then at the bottom of public context, transfer to ${target_context}. Then that target context would do the actual call processing. ________________________________ From: Siri MM To: FreeSWITCH Users Help Sent: Tuesday, October 1, 2013 1:33 AM Subject: Re: [Freeswitch-users] Differentiating internal/external calls Hi Stanislav, Thanks for the reply. I am working in an open system, where the local extensions/gateways don't authenticate with Freeswitch, and hence I am forced to use the public context alone. Again, since the gateway doesn't authenticate, I am unable to use the "variables" defined within gateway settings. On Mon, Sep 30, 2013 at 6:15 PM, Stanislav Sinyagin wrote: the scenario that you described is just ideal for two different contexts. Why do you want to have them in one? > >Of course you can set a variable in an "inline" action in the beginning of a context, like this: > >and later in the dialplan, use this variable in conditions. >Also the variable can be assigned at the SIP gateway, then incoming calls will have it automatically: > > > direction="inbound" means to apply this variable for inbound calls. > > > > > > > > >________________________________ > From: Siri MM >To: FreeSWITCH Users Help >Sent: Monday, September 30, 2013 9:27 AM >Subject: [Freeswitch-users] Differentiating internal/external calls > > > >Hi, > > >Is it possible for me to have two sets of dialplans - one for incoming (to FS) and other for outgoing (from FS), in the same context? On receiving an incoming call, I want to run through only a subset of xmls, and give up on not hitting, and the same for outbound - is this feasible? Or any channel variable that would help me differentiate between a external call and internal call? > > >Thanks! > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/450b9696/attachment-0001.html From sirimmfs at gmail.com Tue Oct 1 12:00:44 2013 From: sirimmfs at gmail.com (Siri MM) Date: Tue, 1 Oct 2013 18:00:44 +1000 Subject: [Freeswitch-users] Differentiating internal/external calls In-Reply-To: <1380610809.19307.YahooMailNeo@web126203.mail.ne1.yahoo.com> References: <1380528957.87347.YahooMailNeo@web126202.mail.ne1.yahoo.com> <1380609276.85817.YahooMailNeo@web126206.mail.ne1.yahoo.com> <1380610809.19307.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: Thanks Stanislav, noted. On Tue, Oct 1, 2013 at 5:00 PM, Stanislav Sinyagin wrote: > you can also do immediate transfer inside the condition, and perform > in-place digit manipulation. > here's an example, it returns the call to the same context, but it can > also be a different context if needed: > http://txlab.wordpress.com/2013/07/06/handling-e164-numbers-in-freeswitch/ > > > > > ------------------------------ > *From:* Stanislav Sinyagin > > *To:* FreeSWITCH Users Help > *Sent:* Tuesday, October 1, 2013 8:34 AM > > *Subject:* Re: [Freeswitch-users] Differentiating internal/external calls > > Siri, that's right, and you can still split the logic between contexts. > > For example, in public context, do the condition matching to classify your > call and set a variable, say, "target_context". > > Then at the bottom of public context, transfer to ${target_context}. Then > that target context would do the actual call processing. > > > > > ------------------------------ > *From:* Siri MM > *To:* FreeSWITCH Users Help > *Sent:* Tuesday, October 1, 2013 1:33 AM > *Subject:* Re: [Freeswitch-users] Differentiating internal/external calls > > Hi Stanislav, > > Thanks for the reply. > > I am working in an open system, where the local extensions/gateways don't > authenticate with Freeswitch, and hence I am forced to use the public > context alone. > > Again, since the gateway doesn't authenticate, I am unable to use the > "variables" defined within gateway settings. > > > > On Mon, Sep 30, 2013 at 6:15 PM, Stanislav Sinyagin wrote: > > the scenario that you described is just ideal for two different contexts. > Why do you want to have them in one? > > Of course you can set a variable in an "inline" action in the beginning of > a context, like this: > > > > and later in the dialplan, use this variable in conditions. > Also the variable can be assigned at the SIP gateway, then incoming calls > will have it automatically: > > > > > > > > > > > > > > > > > direction="inbound" means to apply this variable for inbound calls. > > > > > ------------------------------ > *From:* Siri MM > *To:* FreeSWITCH Users Help > *Sent:* Monday, September 30, 2013 9:27 AM > *Subject:* [Freeswitch-users] Differentiating internal/external calls > > Hi, > > Is it possible for me to have two sets of dialplans - one for incoming (to > FS) and other for outgoing (from FS), in the same context? On receiving an > incoming call, I want to run through only a subset of xmls, and give up on > not hitting, and the same for outbound - is this feasible? Or any channel > variable that would help me differentiate between a external call and > internal call? > > Thanks! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/0c8d85e1/attachment.html From andrew at cassidywebservices.co.uk Tue Oct 1 12:42:56 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 1 Oct 2013 09:42:56 +0100 Subject: [Freeswitch-users] No ringback for calls from cell phones In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Variable_ringback On 1 October 2013 01:00, Siri MM wrote: > Hi, > > I am using FreesWITCH with Sangoma 102 and E1 connection. When I make a > call from a fixed number to one of the internal extensions, ringback can be > heard on the fixed number. However, when I make a call from a cell phone, > although the internal extension rings, I cannot hear any ringing on the > cell phone. I have tried two different mobile phone operators, but the > result is the same. I have tried specifying the channel variables such as > ringback and transfer_ringback int he dialplan, but doesn't help. > > Any pointers on how to debug this? Could it be the telco, in between the > cell phone operators and FS, causing this? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/572345cb/attachment-0001.html From asilva at wirelessmundi.com Tue Oct 1 14:30:00 2013 From: asilva at wirelessmundi.com (Antonio Silva) Date: Tue, 01 Oct 2013 12:30:00 +0200 Subject: [Freeswitch-users] lua session() is blocking... In-Reply-To: References: <1380544273.32043.73.camel@marces.madrid.commsmundi.com> <857FEC87-7EED-4912-BE6E-89E7CEAC2C34@freeswitch.org> <1380550488.32043.151.camel@marces.madrid.commsmundi.com> <1380558118.28284.10.camel@marces.madrid.commsmundi.com> Message-ID: <1380623400.28284.109.camel@marces.madrid.commsmundi.com> Ok... so the only way to trigger a second call from the ringing state is doing it with events... or is there another way? On Tue, 2013-10-01 at 08:38 +0200, Peter Olsson wrote: > Yes, this is expected, that's that way it works. > > > /Peter > > > 2013/9/30 Antonio Silva > > I found it.. > > it's always blocking if no media is received... in this case > the destination reply with a SIP/180 and the session is > waiting for media to continue in the script... > > I put a pastebin where you can see the output and the lua > script, http://pastebin.freeswitch.org/21474 > > (in my minimum configuration i had the action answer before > doing the bridge... my bad...) > > > Regards, > Ant?nio > > On Mon, 2013-09-30 at 16:14 +0200, Antonio Silva wrote: > > > hum... > > > > just try without ignore_early_media=true... same > > behaviour... > > > > then i tried with a minimum FS configuration and it work > > with and without ignore_early_media=true... so it's a > > different problem... > > > > I'm trying to figure out... but thanks for the tip. > > > > > > Regards, > > Ant?nio > > > > > > > > > > On Mon, 2013-09-30 at 07:48 -0500, Brian West wrote: > > > > > You told it to do exactly that when you said ignore_early_media=true > > > > > > -- > > > Brian West > > > brian at freeswitch.org > > > FreeSWITCH Solutions, LLC > > > PO BOX PO BOX 2531 > > > Brookfield, WI 53008-2531 > > > Twitter: @FreeSWITCH_Wire , @briankwest > > > http://www.freeswitchbook.com > > > http://www.freeswitchcookbook.com > > > > > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > > > iNUM: +883 5100 1420 9001 > > > ISN: 410*543 > > > Skype:briankwest > > > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Sep 30, 2013, at 7:31 AM, Antonio Silva wrote: > > > > > > > The problem is that the freswitch.session is blocking and it only returns when the call is already answer... > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/a175a147/attachment.html From fdelawarde at wirelessmundi.com Tue Oct 1 14:53:49 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=) Date: Tue, 01 Oct 2013 12:53:49 +0200 Subject: [Freeswitch-users] ANI missing in cli originate Message-ID: <1380624829.9197.75.camel@luna.madrid.commsmundi.com> Hello, I've seen some FS users using ANI for billing purposes, and I'm not sure if it is normal that the ANI variable is missing on calls originated from CLI (or other originate with no session associated). Would a small patch copying the specified Caller ID Number to ANI in those cases be an acceptable solution? Thanks, Fran?ois. From callum.guy at x-on.co.uk Tue Oct 1 13:51:31 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 1 Oct 2013 10:51:31 +0100 Subject: [Freeswitch-users] RTP Audio Stream Initialise Message-ID: I am trying to understand how audio streams are initialised in FreeSWITCH. I'm working with ESL (inbound and outbound) and trying to record some audio. Through testing i have proven that audio is working correctly for bridged calls, conferences and file playback however not for single legged audio recording - that is, until I play a file first. In my example the playback application is executed from an outbound socket, the call is then parked and the inbound socket captures the park event and starts recording. With the playback in place i get audio, and without it I simply get a silent file of the correct duration presumably meaning that the the RTP stream is not active. My question therefore is: How can I initialise the RTP stream for recording, without having played a file first? Is there a command for this, or an argument I can pass to the record application? My objective is to understand how RTP streams are initialised in FreeSWITCH and any input would be greatly appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/fcdc79a6/attachment.html From elton.machado at gmail.com Tue Oct 1 15:50:14 2013 From: elton.machado at gmail.com (Elton Machado) Date: Tue, 1 Oct 2013 13:50:14 +0200 Subject: [Freeswitch-users] Freeswitch+Radius+IAS/NPS Message-ID: Does anyone have or know how to setup freeswitch radius module to work against NPS or IAS? I'm try to use the documentation for the freeradius but I'm not succeeding in doing it well. TIA, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/447737a5/attachment.html From mike at jerris.com Tue Oct 1 16:21:23 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Oct 2013 08:21:23 -0400 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: References: Message-ID: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> On Oct 1, 2013, at 5:51 AM, Callum Guy wrote: > I am trying to understand how audio streams are initialised in FreeSWITCH. I'm working with ESL (inbound and outbound) and trying to record some audio. Through testing i have proven that audio is working correctly for bridged calls, conferences and file playback however not for single legged audio recording - that is, until I play a file first. > > In my example the playback application is executed from an outbound socket, the call is then parked and the inbound socket captures the park event and starts recording. With the playback in place i get audio, and without it I simply get a silent file of the correct duration presumably meaning that the the RTP stream is not active. > You can try a brief silence_stream at the start of the call to address this. > My question therefore is: How can I initialise the RTP stream for recording, without having played a file first? Is there a command for this, or an argument I can pass to the record application? > FreeSWITCH already does this, its likely whatever your talking to does not. What is on the other end of the call. > My objective is to understand how RTP streams are initialised in FreeSWITCH and any input would be greatly appreciated. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/a9db0186/attachment-0001.html From callum.guy at x-on.co.uk Tue Oct 1 17:00:53 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 1 Oct 2013 14:00:53 +0100 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> Message-ID: Hi Michael, Thanks you for your input. I can confirm that inserting a silence stream prior to the recording works in the same way that playing some actual audio does, resulting in a successful recording. Worth noting is that there appears to be a duration threshold where if the duration is too short (in my case <100ms) then the recording come out blank - presumably this is because the stream has not had enough time to prepare the channel before recording starts. In response to your question the call is a remotely registered VOIP handset (OpenSIPS) dialling directly into FreeSWITCH with all call control handled by ESL. When I stated "initialise the RTP stream for recording" that is my way of trying to communicate that its my understanding that this is what playing the audio file (or silence stream following your comment) was performing in FreeSWITCH. I feel that you have answered my question however i wanted to post some fs_cli output to confirm that this is expected behaviour (or something I should learn to expect!). The lines in red are what i have called "RTP stream initialise". *Call with a silence stream (This records successfully)* 2013-10-01 15:10:35.830044 [DEBUG] switch_ivr.c:612 sofia/internal/ 3333320116 at x-onsip.com Command Execute playback(silence_stream://1000) EXECUTE sofia/internal/3333320116 at x-onsip.complayback(silence_stream://1000) 2013-10-01 15:10:35.830044 [DEBUG] switch_ivr_play_say.c:1308 Codec Activated L16 at 8000hz 1 channels 20ms 2013-10-01 15:10:35.860032 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:10:35.860032 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:10:35.860032 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:10:35.880047 [DEBUG] switch_rtp.c:3704 Correct ip/port confirmed. 2013-10-01 15:10:35.880047 [DEBUG] sofia.c:5685 Channel sofia/internal/ 3333320116 at x-onsip.com entering state [ready][200] 2013-10-01 15:10:36.820081 [DEBUG] switch_ivr_play_say.c:1693 done playing file silence_stream://1000 2013-10-01 15:10:36.820081 [DEBUG] switch_ivr.c:612 sofia/internal/ 3333320116 at x-onsip.com Command Execute record(/usr/share/sounds/recordings/test.wav 10 200) EXECUTE sofia/internal/3333320116 at x-onsip.comrecord(/usr/share/sounds/recordings/test.wav 10 200) 2013-10-01 15:10:36.820081 [DEBUG] switch_ivr_play_say.c:599 Raw Codec Activated 2013-10-01 15:10:36.820081 [DEBUG] switch_core_codec.c:219 sofia/internal/ 3333320116 at x-onsip.com Push codec L16:70 2013-10-01 15:10:39.920081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:10:39.940081 [NOTICE] sofia.c:716 Hangup sofia/internal/ 3333320116 at x-onsip.com [CS_EXECUTE] [NORMAL_CLEARING] *Call without a silence stream (This records silence)* 2013-10-01 15:13:06.660045 [DEBUG] switch_ivr.c:612 sofia/internal/ 3333320116 at x-onsip.com Command Execute playback(silence_stream://10) EXECUTE sofia/internal/3333320116 at x-onsip.com playback(silence_stream://10) 2013-10-01 15:13:06.660045 [DEBUG] switch_ivr_play_say.c:1308 Codec Activated L16 at 8000hz 1 channels 20ms 2013-10-01 15:13:06.680011 [DEBUG] switch_ivr_play_say.c:1693 done playing file silence_stream://10 2013-10-01 15:13:06.680011 [DEBUG] switch_ivr.c:612 sofia/internal/ 3333320116 at x-onsip.com Command Execute record(/usr/share/sounds/recordings/test.wav 10 200) EXECUTE sofia/internal/3333320116 at x-onsip.comrecord(/usr/share/sounds/recordings/test.wav 10 200) 2013-10-01 15:13:06.680011 [DEBUG] switch_ivr_play_say.c:599 Raw Codec Activated 2013-10-01 15:13:06.680011 [DEBUG] switch_core_codec.c:219 sofia/internal/ 3333320116 at x-onsip.com Push codec L16:70 2013-10-01 15:13:06.760081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:13:06.760081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:13:06.760081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] 2013-10-01 15:13:06.780071 [DEBUG] switch_rtp.c:3704 Correct ip/port confirmed. 2013-10-01 15:13:06.780071 [DEBUG] sofia.c:5685 Channel sofia/internal/ 3333320116 at x-onsip.com entering state [ready][200] 2013-10-01 15:13:09.680081 [DEBUG] switch_core_codec.c:244 sofia/internal/ 3333320116 at x-onsip.com Restore previous codec PCMA:8. 2013-10-01 15:13:09.680081 [DEBUG] switch_ivr.c:612 sofia/internal/ 3333320116 at x-onsip.com Command Execute park() Please do let me know if there is anything interesting about the output i am seeing above or that i should apply caution to! Thanks, Callum ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 1 October 2013 13:21, Michael Jerris wrote: > > On Oct 1, 2013, at 5:51 AM, Callum Guy wrote: > > I am trying to understand how audio streams are initialised in > FreeSWITCH. I'm working with ESL (inbound and outbound) and trying to > record some audio. Through testing i have proven that audio is working > correctly for bridged calls, conferences and file playback however not for > single legged audio recording - that is, until I play a file first. > > In my example the playback application is executed from an outbound > socket, the call is then parked and the inbound socket captures the park > event and starts recording. With the playback in place i get audio, and > without it I simply get a silent file of the correct duration presumably > meaning that the the RTP stream is not active. > > You can try a brief silence_stream at the start of the call to address > this. > > My question therefore is: How can I initialise the RTP stream for > recording, without having played a file first? Is there a command for this, > or an argument I can pass to the record application? > > FreeSWITCH already does this, its likely whatever your talking to does > not. What is on the other end of the call. > > My objective is to understand how RTP streams are initialised in > FreeSWITCH and any input would be greatly appreciated. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/371aa375/attachment.html From mike at jerris.com Tue Oct 1 17:38:46 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Oct 2013 09:38:46 -0400 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> Message-ID: <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> You an confirm 100% with a packet trace, but I suspect the difference is, in the case that doesn't work, you are not actually getting any rtp data from the remote side. What type of voip handset is it talking to? Mike On Oct 1, 2013, at 9:00 AM, Callum Guy wrote: > Hi Michael, > > Thanks you for your input. > > I can confirm that inserting a silence stream prior to the recording works in the same way that playing some actual audio does, resulting in a successful recording. Worth noting is that there appears to be a duration threshold where if the duration is too short (in my case <100ms) then the recording come out blank - presumably this is because the stream has not had enough time to prepare the channel before recording starts. > > In response to your question the call is a remotely registered VOIP handset (OpenSIPS) dialling directly into FreeSWITCH with all call control handled by ESL. When I stated "initialise the RTP stream for recording" that is my way of trying to communicate that its my understanding that this is what playing the audio file (or silence stream following your comment) was performing in FreeSWITCH. > > I feel that you have answered my question however i wanted to post some fs_cli output to confirm that this is expected behaviour (or something I should learn to expect!). The lines in red are what i have called "RTP stream initialise". > > Call with a silence stream (This records successfully) > > 2013-10-01 15:10:35.830044 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at x-onsip.com Command Execute playback(silence_stream://1000) > EXECUTE sofia/internal/3333320116 at x-onsip.com playback(silence_stream://1000) > 2013-10-01 15:10:35.830044 [DEBUG] switch_ivr_play_say.c:1308 Codec Activated L16 at 8000hz 1 channels 20ms > 2013-10-01 15:10:35.860032 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:10:35.860032 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:10:35.860032 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:10:35.880047 [DEBUG] switch_rtp.c:3704 Correct ip/port confirmed. > 2013-10-01 15:10:35.880047 [DEBUG] sofia.c:5685 Channel sofia/internal/3333320116 at x-onsip.com entering state [ready][200] > 2013-10-01 15:10:36.820081 [DEBUG] switch_ivr_play_say.c:1693 done playing file silence_stream://1000 > 2013-10-01 15:10:36.820081 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at x-onsip.com Command Execute record(/usr/share/sounds/recordings/test.wav 10 200) > EXECUTE sofia/internal/3333320116 at x-onsip.com record(/usr/share/sounds/recordings/test.wav 10 200) > 2013-10-01 15:10:36.820081 [DEBUG] switch_ivr_play_say.c:599 Raw Codec Activated > 2013-10-01 15:10:36.820081 [DEBUG] switch_core_codec.c:219 sofia/internal/3333320116 at x-onsip.com Push codec L16:70 > 2013-10-01 15:10:39.920081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:10:39.940081 [NOTICE] sofia.c:716 Hangup sofia/internal/3333320116 at x-onsip.com [CS_EXECUTE] [NORMAL_CLEARING] > > Call without a silence stream (This records silence) > > 2013-10-01 15:13:06.660045 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at x-onsip.com Command Execute playback(silence_stream://10) > EXECUTE sofia/internal/3333320116 at x-onsip.com playback(silence_stream://10) > 2013-10-01 15:13:06.660045 [DEBUG] switch_ivr_play_say.c:1308 Codec Activated L16 at 8000hz 1 channels 20ms > 2013-10-01 15:13:06.680011 [DEBUG] switch_ivr_play_say.c:1693 done playing file silence_stream://10 > 2013-10-01 15:13:06.680011 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at x-onsip.com Command Execute record(/usr/share/sounds/recordings/test.wav 10 200) > EXECUTE sofia/internal/3333320116 at x-onsip.com record(/usr/share/sounds/recordings/test.wav 10 200) > 2013-10-01 15:13:06.680011 [DEBUG] switch_ivr_play_say.c:599 Raw Codec Activated > 2013-10-01 15:13:06.680011 [DEBUG] switch_core_codec.c:219 sofia/internal/3333320116 at x-onsip.com Push codec L16:70 > 2013-10-01 15:13:06.760081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:13:06.760081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:13:06.760081 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at x-onsip.com [BREAK] > 2013-10-01 15:13:06.780071 [DEBUG] switch_rtp.c:3704 Correct ip/port confirmed. > 2013-10-01 15:13:06.780071 [DEBUG] sofia.c:5685 Channel sofia/internal/3333320116 at x-onsip.com entering state [ready][200] > 2013-10-01 15:13:09.680081 [DEBUG] switch_core_codec.c:244 sofia/internal/3333320116 at x-onsip.com Restore previous codec PCMA:8. > 2013-10-01 15:13:09.680081 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at x-onsip.com Command Execute park() > > Please do let me know if there is anything interesting about the output i am seeing above or that i should apply caution to! > > Thanks, > > Callum > > ______________________________ > > Callum Guy > Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > > Please consider the environment before printing this email. > > > On 1 October 2013 13:21, Michael Jerris wrote: > > On Oct 1, 2013, at 5:51 AM, Callum Guy wrote: > >> I am trying to understand how audio streams are initialised in FreeSWITCH. I'm working with ESL (inbound and outbound) and trying to record some audio. Through testing i have proven that audio is working correctly for bridged calls, conferences and file playback however not for single legged audio recording - that is, until I play a file first. >> >> In my example the playback application is executed from an outbound socket, the call is then parked and the inbound socket captures the park event and starts recording. With the playback in place i get audio, and without it I simply get a silent file of the correct duration presumably meaning that the the RTP stream is not active. >> > > You can try a brief silence_stream at the start of the call to address this. > >> My question therefore is: How can I initialise the RTP stream for recording, without having played a file first? Is there a command for this, or an argument I can pass to the record application? >> > > FreeSWITCH already does this, its likely whatever your talking to does not. What is on the other end of the call. > >> My objective is to understand how RTP streams are initialised in FreeSWITCH and any input would be greatly appreciated. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/d0254050/attachment-0001.html From richard.mace at gmail.com Tue Oct 1 17:42:47 2013 From: richard.mace at gmail.com (Richard Mace) Date: Tue, 1 Oct 2013 14:42:47 +0100 Subject: [Freeswitch-users] vars.xml In-Reply-To: References: Message-ID: And, could I ask what it is used for ? Richard On 1 Oct 2013 06:50, "Richard Mace" wrote: > Hi All, > I see that in vars.xml there is an entry for Default Country. > Where can I find out what each of the available Default Country options > are please? > > Thanks > > Richard > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/3b7e08a1/attachment.html From mike at jerris.com Tue Oct 1 17:50:59 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Oct 2013 09:50:59 -0400 Subject: [Freeswitch-users] vars.xml In-Reply-To: References: Message-ID: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> conf/vanilla/directory/default/default.xml:21: conf/vanilla/vars.xml:222: Its used to set the numbering_plan variable which as far as I can tell is unused. On Oct 1, 2013, at 9:42 AM, Richard Mace wrote: > And, could I ask what it is used for ? > > Richard > > On 1 Oct 2013 06:50, "Richard Mace" wrote: > Hi All, > I see that in vars.xml there is an entry for Default Country. > Where can I find out what each of the available Default Country options are please? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/0e8bb8c4/attachment.html From callum.guy at x-on.co.uk Tue Oct 1 18:23:09 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 1 Oct 2013 15:23:09 +0100 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> Message-ID: I'm calling in from a Linksys SPA941 and I will be the only user on the FreeSWITCH server. I have been using tcpdump to capture my RTP traffic using the following command: tcpdump -i any -n dst portrange 10000-50000 I configured two test scripts as below: *Script 1: Answer call, generate 1000ms silence (as above) and then record. * This showed a steady stream of UDP data throughout the call and resulted in a recording. *Script 2: Answer call, sleep for 1 second and then record. * This showed a steady stream of UDP data throughout the call but resulted in a blank recording. This is interesting and indicates to me that there is probably something that the FreeSWITCH file play operation is doing (even when its a silence stream) that is not carried out by the record application. I would expect that its unusual for a telephony application to want to answer and record without any play operations first. Thanks again, Callum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/fc29917d/attachment.html From mike at jerris.com Tue Oct 1 18:33:36 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Oct 2013 10:33:36 -0400 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> Message-ID: <29221517-027C-4673-A3F3-09B686AB7317@jerris.com> Your saying you see media flowing in both directions even when the record doesn't work, and when you play the stream in wireshark, there is media in that stream, and its not silent? On Oct 1, 2013, at 10:23 AM, Callum Guy wrote: > I'm calling in from a Linksys SPA941 and I will be the only user on the FreeSWITCH server. I have been using tcpdump to capture my RTP traffic using the following command: > > tcpdump -i any -n dst portrange 10000-50000 > > I configured two test scripts as below: > > Script 1: Answer call, generate 1000ms silence (as above) and then record. > This showed a steady stream of UDP data throughout the call and resulted in a recording. > > Script 2: Answer call, sleep for 1 second and then record. > This showed a steady stream of UDP data throughout the call but resulted in a blank recording. > > This is interesting and indicates to me that there is probably something that the FreeSWITCH file play operation is doing (even when its a silence stream) that is not carried out by the record application. I would expect that its unusual for a telephony application to want to answer and record without any play operations first. > > Thanks again, > > Callum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/c5d9374e/attachment.html From richard.mace at gmail.com Tue Oct 1 18:38:15 2013 From: richard.mace at gmail.com (Richard Mace) Date: Tue, 1 Oct 2013 15:38:15 +0100 Subject: [Freeswitch-users] vars.xml In-Reply-To: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> Message-ID: Oh, OK thanks for that, I won't expose it in my GUI then :) Richard On 1 Oct 2013 14:52, "Michael Jerris" wrote: > conf/vanilla/directory/default/default.xml:21: name="numbering_plan" value="$${default_country}"/> > conf/vanilla/vars.xml:222: data="default_country=US"/> > > Its used to set the numbering_plan variable which as far as I can tell is > unused. > > On Oct 1, 2013, at 9:42 AM, Richard Mace wrote: > > And, could I ask what it is used for ? > > Richard > On 1 Oct 2013 06:50, "Richard Mace" wrote: > >> Hi All, >> I see that in vars.xml there is an entry for Default Country. >> Where can I find out what each of the available Default Country options >> are please? >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/1f119010/attachment-0001.html From callum.guy at x-on.co.uk Tue Oct 1 19:05:24 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 1 Oct 2013 16:05:24 +0100 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: <29221517-027C-4673-A3F3-09B686AB7317@jerris.com> References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> <29221517-027C-4673-A3F3-09B686AB7317@jerris.com> Message-ID: I have just checked and although my test configuration recorded an empty file the packets captured with tcpdump can be successfully played back in Wireshark. The audio in the capture is exactly as I would have expected the recording to be. Are we suggesting that there is a bug of some description here? Is there anything else I can test to confirm? To provide some clarity on my setup here is a list of what i have: 1. Registered Linksys SPA941 on OpenSIPS 2. Direct call from OpenSIPS to FreeSWITCH 3. XML dialplan with ESL outbound connection to socket server 4. Socket server issues SendMsg answer 5. Socket server issues SendMsg park 6. Socket client intercepts CHANNEL_PARK 7. Socket client issues SendMsg record (filepath/test.wav 15 200) ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 1 October 2013 15:33, Michael Jerris wrote: > Your saying you see media flowing in both directions even when the record > doesn't work, and when you play the stream in wireshark, there is media in > that stream, and its not silent? > > On Oct 1, 2013, at 10:23 AM, Callum Guy wrote: > > I'm calling in from a Linksys SPA941 and I will be the only user on the > FreeSWITCH server. I have been using tcpdump to capture my RTP traffic > using the following command: > > tcpdump -i any -n dst portrange 10000-50000 > > I configured two test scripts as below: > > *Script 1: Answer call, generate 1000ms silence (as above) and then > record. * > This showed a steady stream of UDP data throughout the call and resulted > in a recording. > > *Script 2: Answer call, sleep for 1 second and then record. * > This showed a steady stream of UDP data throughout the call but resulted > in a blank recording. > > This is interesting and indicates to me that there is probably something > that the FreeSWITCH file play operation is doing (even when its a silence > stream) that is not carried out by the record application. I would expect > that its unusual for a telephony application to want to answer and record > without any play operations first. > > Thanks again, > > Callum > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/52364e13/attachment.html From anthony.minessale at gmail.com Tue Oct 1 19:15:47 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Oct 2013 10:15:47 -0500 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> <29221517-027C-4673-A3F3-09B686AB7317@jerris.com> Message-ID: Are you running the answer application before you call the record app? It appears as you are not as the codec negotiation seems to take place after you call record. Also you are behind nat and you need to wait for the nat auto-correction to trigger before the media stream is correct so the 10ms is not enough. You need to exchange audio for about 1 seconds for it to fix the problem for you. If you fixed the other end to not need server side nat it would also help your issue. On Tue, Oct 1, 2013 at 10:05 AM, Callum Guy wrote: > I have just checked and although my test configuration recorded an empty > file the packets captured with tcpdump can be successfully played back in > Wireshark. The audio in the capture is exactly as I would have expected the > recording to be. > > Are we suggesting that there is a bug of some description here? Is there > anything else I can test to confirm? > > To provide some clarity on my setup here is a list of what i have: > > 1. Registered Linksys SPA941 on OpenSIPS > 2. Direct call from OpenSIPS to FreeSWITCH > 3. XML dialplan with ESL outbound connection to socket server > 4. Socket server issues SendMsg answer > 5. Socket server issues SendMsg park > 6. Socket client intercepts CHANNEL_PARK > 7. Socket client issues SendMsg record (filepath/test.wav 15 200) > > > > ______________________________ > > Callum Guy > Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 1 October 2013 15:33, Michael Jerris wrote: > >> Your saying you see media flowing in both directions even when the record >> doesn't work, and when you play the stream in wireshark, there is media in >> that stream, and its not silent? >> >> On Oct 1, 2013, at 10:23 AM, Callum Guy wrote: >> >> I'm calling in from a Linksys SPA941 and I will be the only user on the >> FreeSWITCH server. I have been using tcpdump to capture my RTP traffic >> using the following command: >> >> tcpdump -i any -n dst portrange 10000-50000 >> >> I configured two test scripts as below: >> >> *Script 1: Answer call, generate 1000ms silence (as above) and then >> record. * >> This showed a steady stream of UDP data throughout the call and resulted >> in a recording. >> >> *Script 2: Answer call, sleep for 1 second and then record. * >> This showed a steady stream of UDP data throughout the call but resulted >> in a blank recording. >> >> This is interesting and indicates to me that there is probably something >> that the FreeSWITCH file play operation is doing (even when its a silence >> stream) that is not carried out by the record application. I would expect >> that its unusual for a telephony application to want to answer and record >> without any play operations first. >> >> Thanks again, >> >> Callum >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/e6dad9ee/attachment-0001.html From phil at cappgen.com Tue Oct 1 18:56:35 2013 From: phil at cappgen.com (Phil Mickelson) Date: Tue, 1 Oct 2013 10:56:35 -0400 Subject: [Freeswitch-users] Documentation about available options Message-ID: Hello, Is there anywhere in the documentation where I can go and get a list of options available for any particular section of FreeSWITCH. To be more specific; I was looking for the options that I could use in the Directory XML files. For example, I was looking to find if I could specify the VM box for an extension so that multiple extensions would feed the same mailbox. After much searching I did find the "mailbox" setting. However, this took a long time and I just happened on it in a sample. I've now done this for several months and just want to find out if I'm missing something (easy, hard, or I'm just stupid) or it just doesn't exist and I need to continue searching like I do. Thank you. Phil Mickelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/3c504565/attachment.html From callum.guy at x-on.co.uk Tue Oct 1 19:31:53 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 1 Oct 2013 16:31:53 +0100 Subject: [Freeswitch-users] RTP Audio Stream Initialise In-Reply-To: References: <69C90E28-BF94-4115-8483-EB78305A8763@jerris.com> <436B3786-8A91-498C-9308-944A12E796AA@jerris.com> <29221517-027C-4673-A3F3-09B686AB7317@jerris.com> Message-ID: Hi Michael, Thanks for the response. I am sending the application executions in the sequence in my last mail and have just enabled event lock for the answer and confirmed that this has not made a difference to my call. It sounds like your suggestion of nat auto-correction explains the problem and is not something i had considered although i had realised that a 1 second pause solved the issue. I presume that the record application will only work if the RTP stream is ready when the application starts, and this is the reason why my packet capture contains the audio and the recorded file does not? I have attached a complete debug trace from fs_cli in case it helps to further explain my configuration. Many thanks, Callum ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 1 October 2013 16:15, Anthony Minessale wrote: > Are you running the answer application before you call the record app? > It appears as you are not as the codec negotiation seems to take place > after you call record. > Also you are behind nat and you need to wait for the nat auto-correction > to trigger before the media stream is correct so the 10ms is not enough. > You need to exchange audio for about 1 seconds for it to fix the problem > for you. If you fixed the other end to not need server side nat it would > also help your issue. > > > On Tue, Oct 1, 2013 at 10:05 AM, Callum Guy wrote: > >> I have just checked and although my test configuration recorded an empty >> file the packets captured with tcpdump can be successfully played back in >> Wireshark. The audio in the capture is exactly as I would have expected the >> recording to be. >> >> Are we suggesting that there is a bug of some description here? Is there >> anything else I can test to confirm? >> >> To provide some clarity on my setup here is a list of what i have: >> >> 1. Registered Linksys SPA941 on OpenSIPS >> 2. Direct call from OpenSIPS to FreeSWITCH >> 3. XML dialplan with ESL outbound connection to socket server >> 4. Socket server issues SendMsg answer >> 5. Socket server issues SendMsg park >> 6. Socket client intercepts CHANNEL_PARK >> 7. Socket client issues SendMsg record (filepath/test.wav 15 200) >> >> >> >> ______________________________ >> >> Callum Guy >> Developer >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> >> On 1 October 2013 15:33, Michael Jerris wrote: >> >>> Your saying you see media flowing in both directions even when the >>> record doesn't work, and when you play the stream in wireshark, there is >>> media in that stream, and its not silent? >>> >>> On Oct 1, 2013, at 10:23 AM, Callum Guy wrote: >>> >>> I'm calling in from a Linksys SPA941 and I will be the only user on the >>> FreeSWITCH server. I have been using tcpdump to capture my RTP traffic >>> using the following command: >>> >>> tcpdump -i any -n dst portrange 10000-50000 >>> >>> I configured two test scripts as below: >>> >>> *Script 1: Answer call, generate 1000ms silence (as above) and then >>> record. * >>> This showed a steady stream of UDP data throughout the call and resulted >>> in a recording. >>> >>> *Script 2: Answer call, sleep for 1 second and then record. * >>> This showed a steady stream of UDP data throughout the call but resulted >>> in a blank recording. >>> >>> This is interesting and indicates to me that there is probably something >>> that the FreeSWITCH file play operation is doing (even when its a silence >>> stream) that is not carried out by the record application. I would expect >>> that its unusual for a telephony application to want to answer and record >>> without any play operations first. >>> >>> Thanks again, >>> >>> Callum >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/97b11a4c/attachment-0001.html -------------- next part -------------- 2013-10-01 17:50:45.290072 [NOTICE] switch_channel.c:978 New Channel sofia/internal/3333320116 at sipserver.net [674ea0b0-2aad-11e3-ac3d-434c845291ea] 2013-10-01 17:50:45.290072 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.290072 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.290072 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_NEW 2013-10-01 17:50:45.290072 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/3333320116 at sipserver.net) State NEW 2013-10-01 17:50:45.310083 [DEBUG] sofia.c:7793 IP 193.104.89.45 Approved by acl "localnet.auto[]". Access Granted. 2013-10-01 17:50:45.310083 [DEBUG] sofia.c:5685 Channel sofia/internal/3333320116 at sipserver.net entering state [received][100] 2013-10-01 17:50:45.310083 [DEBUG] sofia.c:5696 Remote SDP: v=0 o=- 243381907 243381907 IN IP4 193.104.89.10 s=- c=IN IP4 193.104.89.46 t=0 0 m=audio 60794 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2013-10-01 17:50:45.310083 [DEBUG] sofia.c:5909 (sofia/internal/3333320116 at sipserver.net) State Change CS_NEW -> CS_INIT 2013-10-01 17:50:45.310083 [DEBUG] switch_core_session.c:1333 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_INIT 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/3333320116 at sipserver.net) State INIT 2013-10-01 17:50:45.310083 [DEBUG] mod_sofia.c:87 sofia/internal/3333320116 at sipserver.net SOFIA INIT 2013-10-01 17:50:45.310083 [DEBUG] mod_sofia.c:127 (sofia/internal/3333320116 at sipserver.net) State Change CS_INIT -> CS_ROUTING 2013-10-01 17:50:45.310083 [DEBUG] switch_core_session.c:1333 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/3333320116 at sipserver.net) State INIT going to sleep 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_ROUTING 2013-10-01 17:50:45.310083 [DEBUG] switch_channel.c:2034 (sofia/internal/3333320116 at sipserver.net) Callstate Change DOWN -> RINGING 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/3333320116 at sipserver.net) State ROUTING 2013-10-01 17:50:45.310083 [DEBUG] mod_sofia.c:150 sofia/internal/3333320116 at sipserver.net SOFIA ROUTING 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:117 sofia/internal/3333320116 at sipserver.net Standard ROUTING 2013-10-01 17:50:45.310083 [INFO] mod_dialplan_xml.c:557 Processing Callum Guy <3333320116>->7502 in context public Dialplan: sofia/internal/3333320116 at sipserver.net parsing [public->unloop] continue=false Dialplan: sofia/internal/3333320116 at sipserver.net Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/3333320116 at sipserver.net Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/3333320116 at sipserver.net parsing [public->outside_call] continue=true Dialplan: sofia/internal/3333320116 at sipserver.net Absolute Condition [outside_call] Dialplan: sofia/internal/3333320116 at sipserver.net Action set(outside_call=true) Dialplan: sofia/internal/3333320116 at sipserver.net Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/3333320116 at sipserver.net parsing [public->call_debug] continue=true Dialplan: sofia/internal/3333320116 at sipserver.net Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/3333320116 at sipserver.net parsing [public->public_extensions] continue=false Dialplan: sofia/internal/3333320116 at sipserver.net Regex (FAIL) [public_extensions] destination_number(7502) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/3333320116 at sipserver.net parsing [public->public_did] continue=false Dialplan: sofia/internal/3333320116 at sipserver.net Regex (FAIL) [public_did] destination_number(7502) =~ /^(5551212)$/ break=on-false Dialplan: sofia/internal/3333320116 at sipserver.net parsing [public->sendtosocket] continue=false Dialplan: sofia/internal/3333320116 at sipserver.net Regex (PASS) [sendtosocket] destination_number(7502) =~ /^[0-9]{1,15}$/ break=on-false Dialplan: sofia/internal/3333320116 at sipserver.net Action socket(192.168.5.83:8084) 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/3333320116 at sipserver.net) State Change CS_ROUTING -> CS_EXECUTE 2013-10-01 17:50:45.310083 [DEBUG] switch_core_session.c:1333 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/3333320116 at sipserver.net) State ROUTING going to sleep 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_EXECUTE 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/3333320116 at sipserver.net) State EXECUTE 2013-10-01 17:50:45.310083 [DEBUG] mod_sofia.c:243 sofia/internal/3333320116 at sipserver.net SOFIA EXECUTE 2013-10-01 17:50:45.310083 [DEBUG] switch_core_state_machine.c:209 sofia/internal/3333320116 at sipserver.net Standard EXECUTE EXECUTE sofia/internal/3333320116 at sipserver.net set(outside_call=true) 2013-10-01 17:50:45.310083 [DEBUG] mod_dptools.c:1373 sofia/internal/3333320116 at sipserver.net SET [outside_call]=[true] EXECUTE sofia/internal/3333320116 at sipserver.net export(RFC2822_DATE=Tue, 01 Oct 2013 17:50:45 +0100) 2013-10-01 17:50:45.310083 [DEBUG] switch_channel.c:1145 EXPORT (export_vars) [RFC2822_DATE]=[Tue, 01 Oct 2013 17:50:45 +0100] EXECUTE sofia/internal/3333320116 at sipserver.net socket(192.168.5.83:8084) 2013-10-01 17:50:45.340072 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at sipserver.net Command Execute answer() EXECUTE sofia/internal/3333320116 at sipserver.net answer() 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:5176 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:5176 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:5176 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:3119 Set Codec sofia/internal/3333320116 at sipserver.net PCMA/8000 20 ms 160 samples 64000 bits 2013-10-01 17:50:45.340072 [DEBUG] switch_core_codec.c:111 sofia/internal/3333320116 at sipserver.net Original read codec set to PCMA:8 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:5305 Set 2833 dtmf send/recv payload to 101 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:3378 AUDIO RTP [sofia/internal/3333320116 at sipserver.net] 193.104.89.37 port 17122 -> 193.104.89.46 port 60794 codec: 8 ms: 20 2013-10-01 17:50:45.340072 [DEBUG] switch_rtp.c:1985 Starting timer [soft] 160 bytes per 20ms 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:3642 Set 2833 dtmf send payload to 101 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:3648 Set 2833 dtmf receive payload to 101 2013-10-01 17:50:45.340072 [DEBUG] sofia_glue.c:3675 sofia/internal/3333320116 at sipserver.net Set rtp dtmf delay to 40 2013-10-01 17:50:45.340072 [NOTICE] sofia_glue.c:4286 Pre-Answer sofia/internal/3333320116 at sipserver.net! 2013-10-01 17:50:45.340072 [DEBUG] switch_channel.c:3265 (sofia/internal/3333320116 at sipserver.net) Callstate Change RINGING -> EARLY 2013-10-01 17:50:45.340072 [DEBUG] mod_sofia.c:864 Local SDP sofia/internal/3333320116 at sipserver.net: v=0 o=FreeSWITCH 1380629123 1380629124 IN IP4 193.104.89.37 s=FreeSWITCH c=IN IP4 193.104.89.37 t=0 0 m=audio 17122 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-10-01 17:50:45.340072 [DEBUG] switch_core_session.c:853 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.340072 [NOTICE] mod_dptools.c:1205 Channel [sofia/internal/3333320116 at sipserver.net] has been answered 2013-10-01 17:50:45.340072 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.340072 [DEBUG] switch_channel.c:3542 (sofia/internal/3333320116 at sipserver.net) Callstate Change EARLY -> ACTIVE 2013-10-01 17:50:45.340072 [DEBUG] sofia.c:5685 Channel sofia/internal/3333320116 at sipserver.net entering state [completed][200] 2013-10-01 17:50:45.340072 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at sipserver.net Command Execute park() EXECUTE sofia/internal/3333320116 at sipserver.net park() 2013-10-01 17:50:45.370072 [DEBUG] switch_core_session.c:1133 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.380023 [DEBUG] switch_ivr.c:612 sofia/internal/3333320116 at sipserver.net Command Execute record(/usr/share/sounds/recordings/test.wav 15 200) EXECUTE sofia/internal/3333320116 at sipserver.net record(/usr/share/sounds/recordings/test.wav 15 200) 2013-10-01 17:50:45.380023 [DEBUG] switch_ivr_play_say.c:599 Raw Codec Activated 2013-10-01 17:50:45.380023 [DEBUG] switch_core_codec.c:219 sofia/internal/3333320116 at sipserver.net Push codec L16:70 2013-10-01 17:50:45.390071 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.390071 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.390071 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:45.400072 [DEBUG] sofia.c:5685 Channel sofia/internal/3333320116 at sipserver.net entering state [ready][200] 2013-10-01 17:50:48.380081 [DEBUG] switch_core_codec.c:244 sofia/internal/3333320116 at sipserver.net Restore previous codec PCMA:8. 2013-10-01 17:50:53.910084 [DEBUG] switch_core_session.c:998 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:53.920072 [NOTICE] sofia.c:716 Hangup sofia/internal/3333320116 at sipserver.net [CS_EXECUTE] [NORMAL_CLEARING] 2013-10-01 17:50:53.920072 [DEBUG] switch_channel.c:3096 Send signal sofia/internal/3333320116 at sipserver.net [KILL] 2013-10-01 17:50:53.920072 [DEBUG] switch_core_session.c:1333 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:53.920072 [DEBUG] switch_core_session.c:2731 sofia/internal/3333320116 at sipserver.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-10-01 17:50:53.920072 [DEBUG] switch_ivr.c:650 sofia/internal/3333320116 at sipserver.net skip receive message [AUDIO_SYNC] (channel is hungup already) 2013-10-01 17:50:53.920072 [DEBUG] switch_core_session.c:2731 sofia/internal/3333320116 at sipserver.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/3333320116 at sipserver.net) State EXECUTE going to sleep 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_HANGUP 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:676 (sofia/internal/3333320116 at sipserver.net) State HANGUP 2013-10-01 17:50:53.920072 [DEBUG] mod_sofia.c:504 Channel sofia/internal/3333320116 at sipserver.net hanging up, cause: NORMAL_CLEARING 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:48 sofia/internal/3333320116 at sipserver.net Standard HANGUP, cause: NORMAL_CLEARING 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:676 (sofia/internal/3333320116 at sipserver.net) State HANGUP going to sleep 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:687 (sofia/internal/3333320116 at sipserver.net) Callstate Change ACTIVE -> HANGUP 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/3333320116 at sipserver.net) State Change CS_HANGUP -> CS_REPORTING 2013-10-01 17:50:53.920072 [DEBUG] switch_core_session.c:1333 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_REPORTING 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:759 (sofia/internal/3333320116 at sipserver.net) State REPORTING 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:92 sofia/internal/3333320116 at sipserver.net Standard REPORTING, cause: NORMAL_CLEARING 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:759 (sofia/internal/3333320116 at sipserver.net) State REPORTING going to sleep 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/3333320116 at sipserver.net) State Change CS_REPORTING -> CS_DESTROY 2013-10-01 17:50:53.920072 [DEBUG] switch_core_session.c:1333 Send signal sofia/internal/3333320116 at sipserver.net [BREAK] 2013-10-01 17:50:53.920072 [DEBUG] switch_core_session.c:1541 Session 301 (sofia/internal/3333320116 at sipserver.net) Locked, Waiting on external entities 2013-10-01 17:50:53.920072 [NOTICE] switch_core_session.c:1559 Session 301 (sofia/internal/3333320116 at sipserver.net) Ended 2013-10-01 17:50:53.920072 [NOTICE] switch_core_session.c:1563 Close Channel sofia/internal/3333320116 at sipserver.net [CS_DESTROY] 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/3333320116 at sipserver.net) Callstate Change HANGUP -> DOWN 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:568 (sofia/internal/3333320116 at sipserver.net) Running State Change CS_DESTROY 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:578 (sofia/internal/3333320116 at sipserver.net) State DESTROY 2013-10-01 17:50:53.920072 [DEBUG] mod_sofia.c:397 sofia/internal/3333320116 at sipserver.net SOFIA DESTROY 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:99 sofia/internal/3333320116 at sipserver.net Standard DESTROY 2013-10-01 17:50:53.920072 [DEBUG] switch_core_state_machine.c:578 (sofia/internal/3333320116 at sipserver.net) State DESTROY going to sleep From avi at avimarcus.net Tue Oct 1 19:36:25 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 1 Oct 2013 15:36:25 +0000 Subject: [Freeswitch-users] Documentation about available options In-Reply-To: References: Message-ID: <0000014174a9c34c-3b7a7dd2-b34c-4f80-9c4c-4c5db9f1bb69-000000@email.amazonses.com> You can send calls to the VM of a specific account, e.g. 1234: You have to make sure they can check appropriately, too. -Avi On Tue, Oct 1, 2013 at 5:56 PM, Phil Mickelson wrote: > Hello, > > Is there anywhere in the documentation where I can go and get a list of > options available for any particular section of FreeSWITCH. To be more > specific; I was looking for the options that I could use in the Directory > XML files. For example, I was looking to find if I could specify the VM > box for an extension so that multiple extensions would feed the same > mailbox. After much searching I did find the "mailbox" setting. However, > this took a long time and I just happened on it in a sample. > > I've now done this for several months and just want to find out if I'm > missing something (easy, hard, or I'm just stupid) or it just doesn't exist > and I need to continue searching like I do. > > Thank you. > > Phil Mickelson > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/be5b76f8/attachment.html From intralanman at freeswitch.org Tue Oct 1 21:27:55 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 01 Oct 2013 13:27:55 -0400 Subject: [Freeswitch-users] vars.xml In-Reply-To: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> Message-ID: <524B061B.9050501@freeswitch.org> The numbering_plan variable was originally set a few years back for the concept of adding numbering plans to the default config set (i.e. for doing local dialing rules for the US, UK, HK, etc). The idea was that we'd include a US.xml dialplan that you could transfer the call into so that numbers could be formatted into e.164 according to predefined rules. Since then, mod_translate was added, which will translate the numbers for you in it's dialplan, using a dialplan app for the XML dialplan, or api function call. Using a module instead of multiple dialplans is arguably simpler, and more efficient. The sample config in mod_translate's conf/ directory in the source, will show you an example usage which happens to include the numbering_plan variable. There's also a very minimal wiki page at https://wiki.freeswitch.org/wiki/Mod_translate that gives basic usage examples. -Ray On 10/01/2013 09:50 AM, Michael Jerris wrote: > conf/vanilla/directory/default/default.xml:21: name="numbering_plan" value="$${default_country}"/> > conf/vanilla/vars.xml:222: data="default_country=US"/> > > Its used to set the numbering_plan variable which as far as I can tell > is unused. > > On Oct 1, 2013, at 9:42 AM, Richard Mace > wrote: > >> And, could I ask what it is used for ? >> >> Richard >> >> On 1 Oct 2013 06:50, "Richard Mace" > > wrote: >> >> Hi All, >> I see that in vars.xml there is an entry for Default Country. >> Where can I find out what each of the available Default Country >> options are please? >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/a38b62e5/attachment.html From phil at cappgen.com Tue Oct 1 21:43:19 2013 From: phil at cappgen.com (Phil Mickelson) Date: Tue, 1 Oct 2013 13:43:19 -0400 Subject: [Freeswitch-users] Documentation about available options Message-ID: Avi, I appreciate the response. I have found that before but unfortunately, it doesn't answer my question. I'm still searching for complete documentation. Let me add a couple of things also: 1. I have bought all the books: 1.06, 1.2 and the Cookbook. I read the 1.06 book cover to cover. Never did that before and I've been in this business for 35 years. 2. I paid for and attended ClueCon in August. Had a great time. One of the best conferences I've ever attended. I understand this is open source. No, I haven't paid anything directly for the software. I don't expect anything for free. But, if I have no way of getting answers on my own I'm stuck with two options: Either keep searching in the dark, or go back to Asterisk. Asterisk isn't as fast or capable (at least from what I've seen) as FreeSWITCH. One of my big hot-buttons is multi-tenant. After much struggling I've finally be able to get that to work, again, through no help of the docs. I plan to add my own chapter with a full explanation of how to make this work (if I stick around that long!). Sometimes it doesn't matter how good the product is. Thank you again for taking the time to respond. Regards, Phil Mickelson You can send calls to the VM of a specific account, e.g. 1234: You have to make sure they can check appropriately, too. -Avi On Tue, Oct 1, 2013 at 5:56 PM, Phil Mickelson > wrote: >* Hello,*>**>* Is there anywhere in the documentation where I can go and get a list of*>* options available for any particular section of FreeSWITCH. To be more*>* specific; I was looking for the options that I could use in the Directory*>* XML files. For example, I was looking to find if I could specify the VM*>* box for an extension so that multiple extensions would feed the same*>* mailbox. After much searching I did find the "mailbox" setting. However,*>* this took a long time and I just happened on it in a sample.*>**>* I've now done this for several months and just want to find out if I'm*>* missing something (easy, hard, or I'm just stupid) or it just doesn't exist*>* and I need to continue searching like I do.*>**>* Thank you.*>**>* Phil Mickelson* * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/7413440d/attachment.html From mickstevens at yahoo.com Tue Oct 1 21:58:44 2013 From: mickstevens at yahoo.com (Mick Stevens) Date: Tue, 1 Oct 2013 10:58:44 -0700 (PDT) Subject: [Freeswitch-users] Documentation about available options In-Reply-To: References: Message-ID: <1380650324.52937.YahooMailNeo@web160801.mail.bf1.yahoo.com> Hi Phil, In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble!??). API Commands https://wiki.freeswitch.org/wiki/Mod_commands Dialplan Applications https://wiki.freeswitch.org/wiki/Mod_dptools Channel Variables https://wiki.freeswitch.org/wiki/Channel_Variables XML Dialplan https://wiki.freeswitch.org/wiki/Dialplan_XML Documentation https://wiki.freeswitch.org/wiki/Documentation ? Rgds, Mick Tel/SMS. +44(0)7967 594432 Email/IM/MSN/Skype.?mickstevens at yahoo.com www.facebook.com/mickstevens @mickstevens ________________________________ From: Phil Mickelson To: freeswitch-users Sent: Tuesday, 1 October 2013, 18:43 Subject: Re: [Freeswitch-users] Documentation about available options Avi, I appreciate the response. ?I have found that before but unfortunately, it doesn't answer my question. ?I'm still searching for complete documentation. Let me add a couple of things also: 1. ?I have bought all the books: ?1.06, 1.2 and the Cookbook. ?I read the 1.06 book cover to cover. ?Never did that before and I've been in this business for 35 years. 2. ?I paid for and attended ClueCon in August. ?Had a great time. ?One of the best conferences I've ever attended. I understand this is open source. ?No, I haven't paid anything directly for the software. ?I don't expect anything for free. ?But, if I have no way of getting answers on my own I'm stuck with two options: ?Either keep searching in the dark, or go back to Asterisk. ?Asterisk isn't as fast or capable (at least from what I've seen) as FreeSWITCH. ?One of my big hot-buttons is multi-tenant. ? After much struggling I've finally be able to get that to work, again, through no help of the docs. ?I plan to add my own chapter with a full explanation of how to make this work (if I stick around that long!). Sometimes it doesn't matter how good the product is. Thank you again for taking the time to respond. Regards, Phil Mickelson You can send calls to the VM of a specific account, e.g. 1234: You have to make sure they can check appropriately, too. -Avi On Tue, Oct 1, 2013 at 5:56 PM, Phil Mickelson wrote: >Hello, >>Is there anywhere in the documentation where I can go and get a list of >options available for any particular section of FreeSWITCH. To be more >specific; I was looking for the options that I could use in the Directory >XML files. For example, I was looking to find if I could specify the VM >box for an extension so that multiple extensions would feed the same >mailbox. After much searching I did find the "mailbox" setting. However, >this took a long time and I just happened on it in a sample. >>I've now done this for several months and just want to find out if I'm >missing something (easy, hard, or I'm just stupid) or it just doesn't exist >and I need to continue searching like I do. >>Thank you. >>Phil Mickelson _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/681c0b4a/attachment-0001.html From alipey at gmail.com Tue Oct 1 22:12:26 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 1 Oct 2013 14:12:26 -0400 Subject: [Freeswitch-users] Sending and Receiving faxes - freeswitch In-Reply-To: References: <1378662827.61937.YahooMailNeo@web160502.mail.bf1.yahoo.com> <1378917517.32387.YahooMailNeo@web160504.mail.bf1.yahoo.com> <5241D3AD.1040002@yahoo.com> Message-ID: Thank you Ben. I see it now and was able to see the logs. It does detect the fax but it seems that the fax signalling doesn't go through and the call is dropped: Fax processing not successful - result (49) The call dropped prematurely. I would try to test with a different fax machine or fax software. Or maybe the line is bad and there is some packet loss or jitter. Fax is very sensitive to media. Regards, Ali Pey On Mon, Sep 30, 2013 at 2:03 PM, Ben Langfeld wrote: > Ali, that's the standard FS pastebin, the username and password are in the > challenge dialog. > > Would be nice if the log colouring stuff was simply submitted to Github > for inclusion in Gist... > > > On 30 September 2013 13:48, Ali Pey wrote: > >> Hi Ravi, >> >> You have provided very little info. It's not clear if it is a dialplan >> problem or a fax problem. I tried to access your logs on paste bin and it >> requires a username and password and I don't know how to get one. >> >> I do faxes with freeswitch and it is working pretty well for me. >> >> Regards, >> Ali Pey >> >> >> >> On Tue, Sep 24, 2013 at 2:02 PM, Ravi wrote: >> >>> Hello Everyone ! >>> >>> I hope, this time some one could help me resolve setting up faxes >>> through Freeswitch. I need some help with setting up faxes. Can anyone in >>> this mailing list offer some help to fix this ? >>> >>> Thanks. >>> Ravi >>> >>> >>> >>> On 11/09/13 10:08 PM, Ravi wrote: >>> >>> Brian, >>> >>> Firstly, many congratulations on your wedding ! My best wishes !! >>> >>> Secondly, I am trying to learn Freeswitch and I have made considerable >>> progress, with help from the mailing list :) >>> >>> The configuration that I have is a PRI line, and a Sangoma A101DE >>> card, with Cent OS. I am based out of India and the service provider is >>> called Bharti Airtel. I have setup the system so it works and I can make >>> and receive calls through Freeswitch. >>> >>> The firewall in the machine that runs the freeswitch server is >>> disabled. As far as NAT settings, this is the log that I have when >>> Freeswitch server is started. >>> >>> 2013-09-11 21:55:30.737518 [INFO] switch_nat.c:420 Scanning for NAT >>> 2013-09-11 21:55:30.737699 [DEBUG] switch_nat.c:170 Checking for PMP 1/5 >>> 2013-09-11 21:55:30.738500 [ERR] switch_nat.c:201 Error checking for PMP >>> [general error] >>> 2013-09-11 21:55:30.738513 [DEBUG] switch_nat.c:425 Checking for UPnP >>> 2013-09-11 21:55:34.339478 [INFO] switch_nat.c:434 NAT detected type: >>> upnp, ExtIP: '123.236.215.39' >>> 2013-09-11 21:55:34.340174 [DEBUG] switch_nat.c:264 NAT thread configured >>> 2013-09-11 21:55:34.340501 [DEBUG] switch_nat.c:275 NAT thread started >>> >>> >>> The log when I call the fax number (4302050) from a landline is >>> included in pastebin with link given below: >>> >>> http://pastebin.freeswitch.org/21396 >>> >>> I hope I have answered your questions. I am not sure if you mean that >>> the number 4302050 receives the call, by the term INVITE. Please let me >>> know what I am missing in this, or if any additional information is >>> required. >>> >>> Many Thanks ! >>> Ravi >>> >>> ------------------------------ >>> *From:* Brian West >>> *To:* FreeSWITCH Users Help >>> *Sent:* Sunday, September 8, 2013 11:30 PM >>> *Subject:* Re: [Freeswitch-users] Sending and Receiving faxes - >>> freeswitch >>> >>> How exactly is this number routed to you and via what provider? Have >>> you verified firewall and nat setting to make sure you actually receiving >>> the INVITE? >>> >>> On Sep 8, 2013, at 12:53 PM, Ravi wrote: >>> >>> > When this fax number is called from another telephone, I hear the fax >>> dialtone. But when the fax number is called from any fax machine and try to >>> send a fax, it gives a number not available message. I do not recieve this >>> call in the freeswitch server as well. >>> >>> >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> FreeSWITCH Solutions, LLC >>> PO BOX PO BOX 2531 >>> Brookfield, WI 53008-2531 >>> Twitter: @FreeSWITCH_Wire , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/50240da9/attachment-0001.html From phil at cappgen.com Tue Oct 1 22:55:44 2013 From: phil at cappgen.com (Phil Mickelson) Date: Tue, 1 Oct 2013 14:55:44 -0400 Subject: [Freeswitch-users] (no subject) Message-ID: Mick, Thank you for your list. I've search those also. Let me give you a quick example where I actually found the option but the docs don't supply any additional information. I believe I need to create new VM profiles since I'm running multi-tenant. I figured that one out by looking at other commands that expect a voicemail_profile value. In searching through the default profile I found the parameter "callback-context." Since "context" is generally used in FS in connection with multi-tenant I assume (perhaps incorrectly) that this is important for me. So, I go in search of the parameter in the docs. And, after much searching (I have gotten better at this over time) I find: callback-contextdefault That's it. The exact same line in the XML file. Doesn't explain anything. And, if you look at most of those parameters you'll see the exact same thing. Regards, Phil Mickelson Hi Phil, In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble! ). API Commands https://wiki.freeswitch.org/wiki/Mod_commands Dialplan Applications https://wiki.freeswitch.org/wiki/Mod_dptools Channel Variables https://wiki.freeswitch.org/wiki/Channel_Variables XML Dialplan https://wiki.freeswitch.org/wiki/Dialplan_XML Documentation https://wiki.freeswitch.org/wiki/Documentation Rgds, Mick Tel/SMS. +44(0)7967 594432 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/3f9d9694/attachment.html From dgarcia at anew.com.ve Tue Oct 1 22:57:13 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 01 Oct 2013 14:27:13 -0430 Subject: [Freeswitch-users] Documentation about available options In-Reply-To: References: Message-ID: <524B1B09.3090901@anew.com.ve> Hi Phil About FS docs, there are a lot info for some features, sometimes it seems to be reduce to the minimun. However, as FS is an opensource each module, interface and even the wiki depend on their community. For example, I just search in the wiki about your voicemail question, and I got it in less than 5 minutes: https://wiki.freeswitch.org/wiki/Voicemail On the other hand, there are others features that we have to search more to get it done or to understand how it works ... like valet parking, parking, hold, fifo or callcenter, etc, etc because there are a lot of scenaries that could be quite hard to document all of them. As FS is very powerful, and sometimes a feature can be archived from several ways, If every one who got feature working and take the time to document and post it in the wiki, could be nice. I have access to other solutions (comercial and opensource), and I have to said their docs sometimes does not cover all at all, soon or later you get in the dark trying to get something working. About FS, my opinion is: FS rocks! And the FS community is fantastic, their user list is awesome, you make a question, and you can get an answer asap as you can search in archives because some question has been already answered. Certainly, the wiki need more work to get all info well structured, coherent and explained but I think it can not rely on FS team only, and the community can help (and it help) in keep the wiki updated on the messeaure of their possibilities Regards On 10/01/2013 01:13 PM, Phil Mickelson wrote: > Avi, > > I appreciate the response. I have found that before but > unfortunately, it doesn't answer my question. I'm still searching for > complete documentation. > > Let me add a couple of things also: > > 1. I have bought all the books: 1.06, 1.2 and the Cookbook. I read > the 1.06 book cover to cover. Never did that before and I've been in > this business for 35 years. > 2. I paid for and attended ClueCon in August. Had a great time. One > of the best conferences I've ever attended. > > I understand this is open source. No, I haven't paid anything > directly for the software. I don't expect anything for free. But, if > I have no way of getting answers on my own I'm stuck with two options: > Either keep searching in the dark, or go back to Asterisk. Asterisk > isn't as fast or capable (at least from what I've seen) as FreeSWITCH. > One of my big hot-buttons is multi-tenant. After much struggling > I've finally be able to get that to work, again, through no help of > the docs. I plan to add my own chapter with a full explanation of how > to make this work (if I stick around that long!). > > Sometimes it doesn't matter how good the product is. > > Thank you again for taking the time to respond. > > Regards, > Phil Mickelson > > You can send calls to the VM of a specific account, e.g. 1234: > > > > You have to make sure they can check appropriately, too. > -Avi > > > On Tue, Oct 1, 2013 at 5:56 PM, Phil Mickelson > wrote: > > >/ Hello, > />/ > />/ Is there anywhere in the documentation where I can go and get a list of > />/ options available for any particular section of FreeSWITCH. To be more > />/ specific; I was looking for the options that I could use in the Directory > />/ XML files. For example, I was looking to find if I could specify the VM > />/ box for an extension so that multiple extensions would feed the same > />/ mailbox. After much searching I did find the "mailbox" setting. However, > />/ this took a long time and I just happened on it in a sample. > />/ > />/ I've now done this for several months and just want to find out if I'm > />/ missing something (easy, hard, or I'm just stupid) or it just doesn't exist > />/ and I need to continue searching like I do. > />/ > />/ Thank you. > />/ > />/ Phil Mickelson > / > / > / > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/04f02a9b/attachment.html From steveayre at gmail.com Tue Oct 1 23:08:41 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 1 Oct 2013 20:08:41 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: When you see context mentioned it'll mean a dialplan context - how to separate different types of extension. Multitenant systems may share a single context or have a context for each domain, or anywhere between the two. That's up to you On Tuesday, October 1, 2013, Phil Mickelson wrote: > Mick, > > Thank you for your list. I've search those also. Let me give you a quick > example where I actually found the option but the docs don't supply any > additional information. > > I believe I need to create new VM profiles since I'm running multi-tenant. > I figured that one out by looking at other commands that expect a > voicemail_profile value. In searching through the default profile I found > the parameter "callback-context." Since "context" is generally used in FS > in connection with multi-tenant I assume (perhaps incorrectly) that this is > important for me. So, I go in search of the parameter in the docs. And, > after much searching (I have gotten better at this over time) I find: > > callback-context default > > > > > That's it. The exact same line in the XML file. Doesn't explain > anything. And, if you look at most of those parameters you'll see the > exact same thing. > > Regards, > > Phil Mickelson > > > > Hi Phil, > > In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble! ). > > API Commands > https://wiki.freeswitch.org/wiki/Mod_commands > > > Dialplan Applications > https://wiki.freeswitch.org/wiki/Mod_dptools > > > Channel Variables > https://wiki.freeswitch.org/wiki/Channel_Variables > > > XML Dialplan > https://wiki.freeswitch.org/wiki/Dialplan_XML > > > Documentation > https://wiki.freeswitch.org/wiki/Documentation > > > > Rgds, Mick > Tel/SMS. +44(0)7967 594432 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/2174a698/attachment.html From alipey at gmail.com Tue Oct 1 23:43:34 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 1 Oct 2013 15:43:34 -0400 Subject: [Freeswitch-users] Documentation - fax_ident and fax_header Message-ID: Hello, I noticed that the wiki is not correct when it comes to fax_ident and fax_header: http://wiki.freeswitch.org/wiki/Mod_spandsp#By_mod_spandsp_fax.c - fax_ident - fax identity (numeric only) shown on remote machine during communication. Also may be printed on each page. - fax_header - fax identity (alpha-numeric) show on remote machine during communication. Also may be printed on each page. Fax_ident is the Caller's Station ID (CSID) that can be alpha numeric as well. Fax_header would be used to add a header on top of each page on send. I would be more than happy to fix the wiki page, but not sure what the protocol is. I did try to register to be able to edit the page but never got a password. I found the proper explanation on the mailing list from Steve Underwood: > fax_ident should be set to the telephone number to be used within the > fax exchange. This will typically appear on an LCD display at the far > end. In theory it should be limited to digits, spaces, + and one or two > other characters appropriate to telephone numbers. In practice FAX > machines are usually happy with any text, up to 20 characters. This > string may also play a part in page headers. > > If fax_header is set to a non-null string, a header line will be > inserted at the start of each page, just like a typical FAX machine > does. The fax_ident, fax_header and the page number will be used to form > the text of this line. If you are forwarding FAXes, you probably don't > want to add a header line, as there will already be one that was > inserted by the original source. If you are sending a locally generated > FAX, you probably do want to add header lines to each page. Regards, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/2d7ad42d/attachment-0001.html From james.mortensen at synclio.com Wed Oct 2 00:17:43 2013 From: james.mortensen at synclio.com (James Mortensen) Date: Tue, 1 Oct 2013 13:17:43 -0700 Subject: [Freeswitch-users] Using G729 licenses purchased from Digium on FreeSWITCH Message-ID: Hello, I followed the instructions here http://wiki.freeswitch.org/wiki/Mod_com_g729 to install the mod_com_g729 module. While these instructions which "*explains the installation **quite perfectly" *might work in the general, vanilla cases where things are exactly the same as the documentation author's situation, I suspect mine is slightly different since I'm moving from Asterisk with existing licenses. I've purchased 40 G729 licenses from Digium, but since WebRTC works a lot better on FreeSWITCH, I'm ditching that Asterisk server in favor of FreeSWITCH. I followed the instructions and paste what I believe might be the "sales code" or "activation code" into the validator. It produces an empty licenses.zip file. In the fs_cli, I see "Unrecognised resource G.729A/0", which according to other mailing list threads mean that the licenses aren't valid. The code I got from Digium looks something like this: Key : G729-3543AV54354397WE (not the real code, just an example of what one might look like based on the pattern I see) If anyone has experience porting G729 licenses from Asterisk to FreeSWITCH, please let me know. Thank you! James - *"Being nice to people, building those relationships early on in your career actually helps in making people want to work with you and believing in your vision." - Girish Mathrubootham, CEO/Founder of Freshdesk* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/51631609/attachment.html From krice at freeswitch.org Wed Oct 2 00:38:05 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 1 Oct 2013 15:38:05 -0500 Subject: [Freeswitch-users] Using G729 licenses purchased from Digium on FreeSWITCH In-Reply-To: References: Message-ID: <251BF5CC-1CFD-4115-B325-02A7C658B48B@freeswitch.org> digium's G729 licenses only work with Asterisk. when you license the G729 codec you are not just licensing the patents but you are also licensing the software that implements G729 codecs. Ken Sent from my iPad On Oct 1, 2013, at 15:17, James Mortensen wrote: > Hello, > > I followed the instructions here http://wiki.freeswitch.org/wiki/Mod_com_g729 to install the mod_com_g729 module. While these instructions which "explains the installation quite perfectly" might work in the general, vanilla cases where things are exactly the same as the documentation author's situation, I suspect mine is slightly different since I'm moving from Asterisk with existing licenses. > > I've purchased 40 G729 licenses from Digium, but since WebRTC works a lot better on FreeSWITCH, I'm ditching that Asterisk server in favor of FreeSWITCH. > > I followed the instructions and paste what I believe might be the "sales code" or "activation code" into the validator. It produces an empty licenses.zip file. In the fs_cli, I see "Unrecognised resource G.729A/0", which according to other mailing list threads mean that the licenses aren't valid. > > The code I got from Digium looks something like this: > > Key : G729-3543AV54354397WE (not the real code, just an example of what one might look like based on the pattern I see) > > If anyone has experience porting G729 licenses from Asterisk to FreeSWITCH, please let me know. > > Thank you! > > James > > > - "Being nice to people, building those relationships early on in your career actually helps in making people want to work with you and believing in your vision." - Girish Mathrubootham, CEO/Founder of Freshdesk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/93509939/attachment.html From mike at jerris.com Wed Oct 2 00:42:43 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Oct 2013 16:42:43 -0400 Subject: [Freeswitch-users] Using G729 licenses purchased from Digium on FreeSWITCH In-Reply-To: References: Message-ID: <8958C3D0-91A3-4D5A-B621-462AAEADA629@jerris.com> The licenses you got from Digium are for their g.729 codec only, and not for other implementations. Our licenses will similarly work only with our implementation. Mike On Oct 1, 2013, at 4:17 PM, James Mortensen wrote: > Hello, > > I followed the instructions here http://wiki.freeswitch.org/wiki/Mod_com_g729 to install the mod_com_g729 module. While these instructions which "explains the installation quite perfectly" might work in the general, vanilla cases where things are exactly the same as the documentation author's situation, I suspect mine is slightly different since I'm moving from Asterisk with existing licenses. > > I've purchased 40 G729 licenses from Digium, but since WebRTC works a lot better on FreeSWITCH, I'm ditching that Asterisk server in favor of FreeSWITCH. > > I followed the instructions and paste what I believe might be the "sales code" or "activation code" into the validator. It produces an empty licenses.zip file. In the fs_cli, I see "Unrecognised resource G.729A/0", which according to other mailing list threads mean that the licenses aren't valid. > > The code I got from Digium looks something like this: > > Key : G729-3543AV54354397WE (not the real code, just an example of what one might look like based on the pattern I see) > > If anyone has experience porting G729 licenses from Asterisk to FreeSWITCH, please let me know. > > Thank you! > > James > > > - "Being nice to people, building those relationships early on in your career actually helps in making people want to work with you and believing in your vision." - Girish Mathrubootham, CEO/Founder of Freshdesk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/b6ab99ea/attachment.html From ssinyagin at yahoo.com Wed Oct 2 01:12:40 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 1 Oct 2013 14:12:40 -0700 (PDT) Subject: [Freeswitch-users] Documentation about available options In-Reply-To: References: Message-ID: <1380661960.68468.YahooMailNeo@web126205.mail.ne1.yahoo.com> Phil, at the bottom of Voicemail wiki page, there's a section called "Voicemail Callback". Does it not explain the details? Also, there is a variable for callback context, so that you don't need to create a separate profile for a new tenant in a multi-tenant setup. Well, sometimes I had to look up in the source code when documentation was not clear. But I must say, there was never a task that I could not implement within FreeSWITCH. I've listed some use cases at http://voxserv.ch if you're interested. So, having read the 1.0.6 book and using Wiki search, and sometimes looking into the sources, I find this product completely manageable, and pretty well documented. You can also easily find consultancy services from the developers team or other guys, and save your time. cheers, stan ________________________________ From: Phil Mickelson To: freeswitch-users Sent: Tuesday, October 1, 2013 8:55 PM Subject: [Freeswitch-users] (no subject) Mick, Thank you for your list. ?I've search those also. ?Let me give you a quick example where I actually found the option but the docs don't supply any additional information. I believe I need to create new VM profiles since I'm running multi-tenant. ?I figured that one out by looking at other commands that expect a voicemail_profile value. ?In searching through the default profile I found the parameter "callback-context." ?Since "context" is generally used in FS in connection with multi-tenant I assume (perhaps incorrectly) that this is important for me. ?So, I go in search of the parameter in the docs. ?And, after much searching (I have gotten better at this over time) I find: callback-context default That's it. ?The exact same line in the XML file. ?Doesn't explain anything. ?And, if you look at most of those parameters you'll see the exact same thing. Regards, Phil Mickelson Hi Phil, In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble!??). API Commands https://wiki.freeswitch.org/wiki/Mod_commands Dialplan Applications https://wiki.freeswitch.org/wiki/Mod_dptools Channel Variables https://wiki.freeswitch.org/wiki/Channel_Variables XML Dialplan https://wiki.freeswitch.org/wiki/Dialplan_XML Documentation https://wiki.freeswitch.org/wiki/Documentation ? Rgds, Mick Tel/SMS. +44(0)7967 594432 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/1de08588/attachment-0001.html From grcamauer at gmail.com Wed Oct 2 01:24:12 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 1 Oct 2013 18:24:12 -0300 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: All, Thanks for the responses, specially for the tips on the tools. I have installed both htop and sysstat (mpstat) and I am monitoring my system. I will post back any relevant findings. Guillermo On Mon, Sep 30, 2013 at 9:15 PM, Steven Ayre wrote: > Personally I prefer htop. You'll see CPU usage broken out for each core. > > top itself also has a similar option - run top and press '1' > > > > On 30 September 2013 20:26, Guillermo Ruiz Camauer wrote: > >> Yes, TOP is giving me the 12.5% reading. I understand that this is an >> average reading of all cores. The fact that it never seems to go over >> 12.5% tells me that I am never using more than 1 core at a time, and that >> is is just the one process jumping around the different cores. Is this a >> valid assumption? I want to test using more threads to see if I a hitting >> a bottleneck with this.. >> >> Guillermo >> >> >> On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: >> >>> If I only have one Voip provider, can I still have more than one SIP >>>> profile? My provider authenticates via IP >>> >>> >>> Profiles bind to a single ip:port combination. >>> >>> You could run multiple profiles each on a separate port. That way >>> they'll all send to the provider using the same IP. >>> >>> >>> >>> >>> >>> On 30 September 2013 18:54, Guillermo Ruiz Camauer wrote: >>> >>>> In the Wiki, under "Performance testing and configurations", one of >>>> the suggestions given under "Recommended SIP Settings" is: >>>> >>>> libsofia only handles 1 thread per profile, so if that is your bottle >>>> neck use more profiles >>>> >>>> If I only have one Voip provider, can I still have more than one SIP >>>> profile? My provider authenticates via IP. I currently run 240 concurrent >>>> calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core >>>> machine. This means that one core is maxing out. How can I get more >>>> threads up to distribute the load? >>>> >>>> >>>> >>>> -- >>>> Guillermo Ruiz Camauer >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/c97ee597/attachment.html From krice at freeswitch.org Wed Oct 2 02:40:12 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 01 Oct 2013 17:40:12 -0500 Subject: [Freeswitch-users] ClueCon Weekly - Oct 2 Tomorrow! Message-ID: Hey Guys, ClueCon Weekly for Oct 2 is rapidly upon us. This week, mod_httapi! We?ve talked about this before, but this week, I?ll be demoing how simple it really is to create an application using something as simple as mod_httapi and PHP! Join us tomorrow at 1PM Eastern (that?s 10AM Pacific) to check out mod_httapi and how easy you can have your WebDevs start creating applications! Where? sip:888 at conference.freeswitch.org for more info see https://wiki.freeswitch.org/wiki/FS_weekly_2013_10_02 or just for the calling instructions see: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/83e751c5/attachment.html From lconroy at insensate.co.uk Wed Oct 2 03:26:04 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 2 Oct 2013 00:26:04 +0100 Subject: [Freeswitch-users] OS X works with tarball, not with 1.2 stable git? Message-ID: <79F5256A-5DBE-4A52-9F83-88F59D65AA51@insensate.co.uk> Hi Folks, has anyone managed to get the latest git 1.2 stable branch to run correctly on OS X 10.6.8? I have no problems getting the 1.2.13 tarball to build and run, but ... With the 1.2 stable git (as of today), whilst fs builds OK, when run sofia seems to accept regs from clients, will accept an INVITE from the client and send back the 100, set up a worker for the call, but then ignore it. Result is the client sits spinning its wheels waiting for a final response, and getting nothing. Trying to shutdown fs meets a wait for the worker thread to quit, and eventually a time out when the parent gives up. Like I said, this works fine with the tarball. Is there a tagged git release equivalent to the tarball? Without one it's a challenge to track back to find where the problem raised its head. [yup, I am a tyro at this new-fangled git stuff; this may be obvious, but not to me ?] all the best, Lawrence From krice at freeswitch.org Wed Oct 2 04:04:57 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 01 Oct 2013 19:04:57 -0500 Subject: [Freeswitch-users] OS X works with tarball, not with 1.2 stable git? In-Reply-To: <79F5256A-5DBE-4A52-9F83-88F59D65AA51@insensate.co.uk> Message-ID: Yes there is a git tag... git checkout v1.2.13 from your git tree to get the version that's in the tarball If that doesn't build theres probably an issue that needs to be looked at in how its getting bootstrapped on your system and a ticket w/ a full log (stderr and stdout) attached to the ticket as a .txt file will hopefully shed some light on the why On 10/1/13 6:26 PM, "Lawrence Conroy" wrote: > Hi Folks, > has anyone managed to get the latest git 1.2 stable branch to run correctly > on OS X 10.6.8? > > I have no problems getting the 1.2.13 tarball to build and run, but ... > > With the 1.2 stable git (as of today), whilst fs builds OK, when run sofia > seems to accept regs from clients, will accept an INVITE from the client and > send back the 100, set up a worker for the call, but then ignore it. > Result is the client sits spinning its wheels waiting for a final response, > and getting nothing. > Trying to shutdown fs meets a wait for the worker thread to quit, and > eventually a time out when the parent gives up. > > Like I said, this works fine with the tarball. Is there a tagged git release > equivalent to the tarball? > Without one it's a challenge to track back to find where the problem raised > its head. > [yup, I am a tyro at this new-fangled git stuff; this may be obvious, but not > to me ?] > > all the best, > Lawrence > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From steveu at coppice.org Wed Oct 2 04:17:43 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 02 Oct 2013 08:17:43 +0800 Subject: [Freeswitch-users] Documentation - fax_ident and fax_header In-Reply-To: References: Message-ID: <524B6627.6060402@coppice.org> You can pretty much use my test verbatim on the wiki page, and you will have a complete description of these options. Regards, Steve On 10/02/2013 03:43 AM, Ali Pey wrote: > Hello, > > I noticed that the wiki is not correct when it comes to fax_ident and > fax_header: > > http://wiki.freeswitch.org/wiki/Mod_spandsp#By_mod_spandsp_fax.c > > - fax_ident - fax identity (numeric only) shown on remote machine > during communication. Also may be printed on each page. > - fax_header - fax identity (alpha-numeric) show on remote machine > during communication. Also may be printed on each page. > > > Fax_ident is the Caller's Station ID (CSID) that can be alpha numeric > as well. Fax_header would be used to add a header on top of each page > on send. > > I would be more than happy to fix the wiki page, but not sure what the > protocol is. I did try to register to be able to edit the page but > never got a password. > > > I found the proper explanation on the mailing list from Steve Underwood: > > > fax_ident should be set to the telephone number to be used within the > > fax exchange. This will typically appear on an LCD display at the far > > end. In theory it should be limited to digits, spaces, + and one or two > > other characters appropriate to telephone numbers. In practice FAX > > machines are usually happy with any text, up to 20 characters. This > > string may also play a part in page headers. > > > > If fax_header is set to a non-null string, a header line will be > > inserted at the start of each page, just like a typical FAX machine > > does. The fax_ident, fax_header and the page number will be used to > form > > the text of this line. If you are forwarding FAXes, you probably don't > > want to add a header line, as there will already be one that was > > inserted by the original source. If you are sending a locally generated > > FAX, you probably do want to add header lines to each page. > > > Regards, > Ali Pey > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Oct 2 05:31:36 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Oct 2013 21:31:36 -0400 Subject: [Freeswitch-users] OS X works with tarball, not with 1.2 stable git? In-Reply-To: References: Message-ID: <9FDCB978-A0F0-4F14-ADF7-7A4EF367F0EE@jerris.com> I pulled down the latest xcode today, I'll be working on getting the build with latest compilers working this week, i know thats an issue, otherwise I have not had anyone say they were having issues with osx, so I'll see if I get the same when I get this all building again. Mike On Oct 1, 2013, at 8:04 PM, Ken Rice wrote: > Yes there is a git tag... > > git checkout v1.2.13 from your git tree to get the version that's in the > tarball > > If that doesn't build theres probably an issue that needs to be looked at in > how its getting bootstrapped on your system and a ticket w/ a full log > (stderr and stdout) attached to the ticket as a .txt file will hopefully > shed some light on the why > > On 10/1/13 6:26 PM, "Lawrence Conroy" wrote: > >> Hi Folks, >> has anyone managed to get the latest git 1.2 stable branch to run correctly >> on OS X 10.6.8? >> >> I have no problems getting the 1.2.13 tarball to build and run, but ... >> >> With the 1.2 stable git (as of today), whilst fs builds OK, when run sofia >> seems to accept regs from clients, will accept an INVITE from the client and >> send back the 100, set up a worker for the call, but then ignore it. >> Result is the client sits spinning its wheels waiting for a final response, >> and getting nothing. >> Trying to shutdown fs meets a wait for the worker thread to quit, and >> eventually a time out when the parent gives up. >> >> Like I said, this works fine with the tarball. Is there a tagged git release >> equivalent to the tarball? >> Without one it's a challenge to track back to find where the problem raised >> its head. >> [yup, I am a tyro at this new-fangled git stuff; this may be obvious, but not >> to me ?] >> >> all the best, >> Lawrence From alipey at gmail.com Wed Oct 2 06:25:27 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 1 Oct 2013 22:25:27 -0400 Subject: [Freeswitch-users] Documentation - fax_ident and fax_header In-Reply-To: <524B6627.6060402@coppice.org> References: <524B6627.6060402@coppice.org> Message-ID: Thank you Steve. My point was that the description on the wiki page is not correct and I was looking a for a way to fix it for people who would read it in the future. I managed to figure out what those are with testing and some research. Regards, Ali Pey On Tue, Oct 1, 2013 at 8:17 PM, Steve Underwood wrote: > You can pretty much use my test verbatim on the wiki page, and you will > have a complete description of these options. > > Regards, > Steve > > On 10/02/2013 03:43 AM, Ali Pey wrote: > > Hello, > > > > I noticed that the wiki is not correct when it comes to fax_ident and > > fax_header: > > > > http://wiki.freeswitch.org/wiki/Mod_spandsp#By_mod_spandsp_fax.c > > > > - fax_ident - fax identity (numeric only) shown on remote machine > > during communication. Also may be printed on each page. > > - fax_header - fax identity (alpha-numeric) show on remote machine > > during communication. Also may be printed on each page. > > > > > > Fax_ident is the Caller's Station ID (CSID) that can be alpha numeric > > as well. Fax_header would be used to add a header on top of each page > > on send. > > > > I would be more than happy to fix the wiki page, but not sure what the > > protocol is. I did try to register to be able to edit the page but > > never got a password. > > > > > > I found the proper explanation on the mailing list from Steve Underwood: > > > > > fax_ident should be set to the telephone number to be used within the > > > fax exchange. This will typically appear on an LCD display at the far > > > end. In theory it should be limited to digits, spaces, + and one or two > > > other characters appropriate to telephone numbers. In practice FAX > > > machines are usually happy with any text, up to 20 characters. This > > > string may also play a part in page headers. > > > > > > If fax_header is set to a non-null string, a header line will be > > > inserted at the start of each page, just like a typical FAX machine > > > does. The fax_ident, fax_header and the page number will be used to > > form > > > the text of this line. If you are forwarding FAXes, you probably don't > > > want to add a header line, as there will already be one that was > > > inserted by the original source. If you are sending a locally generated > > > FAX, you probably do want to add header lines to each page. > > > > > > Regards, > > Ali Pey > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131001/6bb8af12/attachment.html From steveayre at gmail.com Wed Oct 2 14:04:41 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 2 Oct 2013 11:04:41 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: When you see context mentioned it'll mean a dialplan context - how to separate different types of extension. Multitenant systems may share a single context or have a context for each domain, or anywhere between the two. That's up to you On Tuesday, October 1, 2013, Phil Mickelson wrote: > Mick, > > Thank you for your list. I've search those also. Let me give you a quick > example where I actually found the option but the docs don't supply any > additional information. > > I believe I need to create new VM profiles since I'm running multi-tenant. > I figured that one out by looking at other commands that expect a > voicemail_profile value. In searching through the default profile I found > the parameter "callback-context." Since "context" is generally used in FS > in connection with multi-tenant I assume (perhaps incorrectly) that this is > important for me. So, I go in search of the parameter in the docs. And, > after much searching (I have gotten better at this over time) I find: > > callback-context default > > > > > That's it. The exact same line in the XML file. Doesn't explain > anything. And, if you look at most of those parameters you'll see the > exact same thing. > > Regards, > > Phil Mickelson > > > > Hi Phil, > > In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble! ). > > API Commands > https://wiki.freeswitch.org/wiki/Mod_commands > > > Dialplan Applications > https://wiki.freeswitch.org/wiki/Mod_dptools > > > Channel Variables > https://wiki.freeswitch.org/wiki/Channel_Variables > > > XML Dialplan > https://wiki.freeswitch.org/wiki/Dialplan_XML > > > Documentation > https://wiki.freeswitch.org/wiki/Documentation > > > > Rgds, Mick > Tel/SMS. +44(0)7967 594432 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/6700d07c/attachment.html From lconroy at insensate.co.uk Wed Oct 2 14:53:38 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 2 Oct 2013 11:53:38 +0100 Subject: [Freeswitch-users] OS X works with tarball, not with 1.2 stable git? In-Reply-To: <9FDCB978-A0F0-4F14-ADF7-7A4EF367F0EE@jerris.com> References: <9FDCB978-A0F0-4F14-ADF7-7A4EF367F0EE@jerris.com> Message-ID: <5CDF898D-37D9-4303-8E13-85F7A6D8653A@insensate.co.uk> Hi Mike, I had a play with Xcode 5 on 10.8.5 yesterday -- got this to build after a few hacks (compiler is a lot more picky so flagged up things like testing a enumeration-typed variable against 9999, ...). Hacking in changes to the code to quiet those meant it built, but has the same behaviour as described when run -- seems to lose interest in worker thread, whilst still responding to incoming messages as expected. It DID give me good hints on an underlying residual lurky in the portaudio fix of FS-4786, so worth the fun. all the best, Lawrence On 2 Oct 2013, at 02:31, Michael Jerris wrote: > I pulled down the latest xcode today, I'll be working on getting the build with latest compilers working this week, i know thats an issue, otherwise I have not had anyone say they were having issues with osx, so I'll see if I get the same when I get this all building again. > > Mike > > On Oct 1, 2013, at 8:04 PM, Ken Rice wrote: > >> Yes there is a git tag... >> >> git checkout v1.2.13 from your git tree to get the version that's in the >> tarball >> >> If that doesn't build theres probably an issue that needs to be looked at in >> how its getting bootstrapped on your system and a ticket w/ a full log >> (stderr and stdout) attached to the ticket as a .txt file will hopefully >> shed some light on the why >> >> On 10/1/13 6:26 PM, "Lawrence Conroy" wrote: >> >>> Hi Folks, >>> has anyone managed to get the latest git 1.2 stable branch to run correctly >>> on OS X 10.6.8? >>> >>> I have no problems getting the 1.2.13 tarball to build and run, but ... >>> >>> With the 1.2 stable git (as of today), whilst fs builds OK, when run sofia >>> seems to accept regs from clients, will accept an INVITE from the client and >>> send back the 100, set up a worker for the call, but then ignore it. >>> Result is the client sits spinning its wheels waiting for a final response, >>> and getting nothing. >>> Trying to shutdown fs meets a wait for the worker thread to quit, and >>> eventually a time out when the parent gives up. >>> >>> Like I said, this works fine with the tarball. Is there a tagged git release >>> equivalent to the tarball? >>> Without one it's a challenge to track back to find where the problem raised >>> its head. >>> [yup, I am a tyro at this new-fangled git stuff; this may be obvious, but not >>> to me ?] >>> >>> all the best, >>> Lawrence > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From callum.guy at x-on.co.uk Wed Oct 2 15:00:14 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 2 Oct 2013 12:00:14 +0100 Subject: [Freeswitch-users] Conference Configure Multiple Prompts Message-ID: Hi All, I am working with the conference application and controlling it remotely with ESL. I have a requirement to assign multiple prompts to the same event - in this case conference join. I have configured an application to record a users name, and wish to play "NAME..." followed by a default prompt "..has joined the conference". Similarly i plan to use the same idea when callers leave the conference. I am first looking at the application parameters in "autoload_configs/conference.conf.xml" and wondering if there is a way to share a list of prompts - something along the lines of the following which i naively hoped would repeat the prompt twice (obviously it doesn't work, otherwise i wouldn't be asking!): I can advance this by changing strategy and broadcasting an event (play application) to all known conference members however i suspect there may be a better and cleaner way of doing this so i'm putting the question to the community! Any help/pointers would be greatly appreciated. Best Regards, Callum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/1f4afdf1/attachment.html From mike at jerris.com Wed Oct 2 16:08:22 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Oct 2013 08:08:22 -0400 Subject: [Freeswitch-users] Conference Configure Multiple Prompts In-Reply-To: References: Message-ID: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> Check out phrases: https://wiki.freeswitch.org/wiki/Speech_Phrase_Management On Oct 2, 2013, at 7:00 AM, Callum Guy wrote: > Hi All, > > I am working with the conference application and controlling it remotely with ESL. I have a requirement to assign multiple prompts to the same event - in this case conference join. > > I have configured an application to record a users name, and wish to play "NAME..." followed by a default prompt "..has joined the conference". Similarly i plan to use the same idea when callers leave the conference. > > I am first looking at the application parameters in "autoload_configs/conference.conf.xml" and wondering if there is a way to share a list of prompts - something along the lines of the following which i naively hoped would repeat the prompt twice (obviously it doesn't work, otherwise i wouldn't be asking!): > > > > I can advance this by changing strategy and broadcasting an event (play application) to all known conference members however i suspect there may be a better and cleaner way of doing this so i'm putting the question to the community! > > Any help/pointers would be greatly appreciated. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/285999bc/attachment.html From crazygabry at gmail.com Wed Oct 2 18:55:23 2013 From: crazygabry at gmail.com (crazygabry) Date: Wed, 2 Oct 2013 07:55:23 -0700 (PDT) Subject: [Freeswitch-users] best way to change configuration files programmatically Message-ID: <1380725722254-7595412.post@n2.nabble.com> Hello everyone, i'm doing a project and i need to edit xml (dialplan, users and so on) from an external software placed on another server. I saw that with mod_curl can read the dialplan or other configurations from remote places but i don't want to introduce latency when freeswitch parses the dialplan. I came from asterisk and it was possible to change files with Event API, i saw that with mod_event_socket (i think) it isn't possible to change conf files. Can you suggest me the best way to change xml files from remote server? Thanks in advice -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412.html Sent from the freeswitch-users mailing list archive at Nabble.com. From alipey at gmail.com Wed Oct 2 19:00:04 2013 From: alipey at gmail.com (Ali Pey) Date: Wed, 2 Oct 2013 11:00:04 -0400 Subject: [Freeswitch-users] canceling a bridge command Message-ID: Hello, Is there a way to cancel or stop a bridge command? I'd like to listen to dtmf digits and if the user presses a dtmf digit, cacel the bridge command that was already issued to multiple destination and send the call to a diffrent location. Is this possible? How? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/546623e6/attachment.html From callum.guy at x-on.co.uk Wed Oct 2 19:50:32 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 2 Oct 2013 16:50:32 +0100 Subject: [Freeswitch-users] Conference Configure Multiple Prompts In-Reply-To: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> References: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> Message-ID: Thanks Mike, I've now built and tested the phrase macro and it works but i'm having some difficulty working out how to use it as a conference announcement. My goal is to record a users name as they dial the conference number, store that filename in a variable and when they enter the conference i want to play "users name" followed by "has joined the conference". In an ideal world i'd be able to do the following in my conference profile: Firstly phrase is invalid in this context so that's no good. Also this would need access to the file path and name of the file created by the person entering but at present the variable only exists in the channel scope so that won't work either, and going global seems messy. I'm going to try uuid_broadcast unless i'm missing something obvious? ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 2 October 2013 13:08, Michael Jerris wrote: > Check out phrases: > https://wiki.freeswitch.org/wiki/Speech_Phrase_Management > > > On Oct 2, 2013, at 7:00 AM, Callum Guy wrote: > > Hi All, > > I am working with the conference application and controlling it remotely > with ESL. I have a requirement to assign multiple prompts to the same event > - in this case conference join. > > I have configured an application to record a users name, and wish to play > "NAME..." followed by a default prompt "..has joined the conference". > Similarly i plan to use the same idea when callers leave the conference. > > I am first looking at the application parameters > in "autoload_configs/conference.conf.xml" and wondering if there is a way > to share a list of prompts - something along the lines of the following > which i naively hoped would repeat the prompt twice (obviously it doesn't > work, otherwise i wouldn't be asking!): > > value="/usr/share/sounds/ConferencingOnlyPerson.wav > /usr/share/sounds/ConferencingOnlyPerson.wav"/> > > I can advance this by changing strategy and broadcasting an event (play > application) to all known conference members however i suspect there may be > a better and cleaner way of doing this so i'm putting the question to the > community! > > Any help/pointers would be greatly appreciated. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/0dc10930/attachment-0001.html From karl at xtronics.com Wed Oct 2 20:00:34 2013 From: karl at xtronics.com (Karl Schmidt) Date: Wed, 02 Oct 2013 11:00:34 -0500 Subject: [Freeswitch-users] FAX nightmare continues Message-ID: <524C4322.5060406@xtronics.com> This was tested and worked, but fails in the real world. I have a landline that brings in FAX to a grandstream gxw4104 - which gets connected to a gxw4008 - to a modem to receive faxes via hylafax. There must be too much distortion as I'm getting lots of errors in hylafax and it finally drops the call. I have PCMU on both of the grandstream boxes so I'm thinking that freeswitch should just pass the data. Are there any other settings I might fiddle with? ,.,.,.,.,.,.,.,.,.,. This also brings up a question about echo suppression - there are places in the grandstreams for echo suppression (also in the polycom phones). Should both ends do echo cancellation or just one end? What is the best practice for setting echo cancellation? -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Ask any politician in private, It's a blast spending other peoples money. -------------------------------------------------------------------------------- From noc at sonerep.com Wed Oct 2 19:29:03 2013 From: noc at sonerep.com (Groupe SOGO) Date: Wed, 02 Oct 2013 16:29:03 +0100 Subject: [Freeswitch-users] Integrate openfire to freeswitch Message-ID: <524C3BBF.3010306@sonerep.com> Dear Friends, I installed freeswitch latest version on CentOS 6.4 and everything is working well. The default sip internal accounts can make and receive calls. I also installed openfire 3.8.1 on the same machine and it is working well. I can create users with spark client 2.6.3 and they can chat and create conference. So everything works very well. Now I want to integrate openfire to freeswitch. Has anyone tried it? What did he do to achieve that? What can i do to achieve the integration? thanks in advance for your help. Labolinux K. Amouzou From callum.guy at x-on.co.uk Wed Oct 2 20:31:33 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 2 Oct 2013 17:31:33 +0100 Subject: [Freeswitch-users] Conference Configure Multiple Prompts In-Reply-To: References: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> Message-ID: OK so i've run into another issue when trying uuid_broadcast. I'm running the following just after joining an existing conference: api uuid_broadcast [UUID] phrase:conferenceAnnounce:${ConferenceJoinName} both This is having no immediate effect however when either leg closes they hear the broadcast message set by the closing channel! I'm sure i'm doing something wrong and i'll revisit tomorrow! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/58ae4f1d/attachment.html From callum.guy at x-on.co.uk Wed Oct 2 21:02:00 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 2 Oct 2013 18:02:00 +0100 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: <1380725722254-7595412.post@n2.nabble.com> References: <1380725722254-7595412.post@n2.nabble.com> Message-ID: You should be able to remotely run reloadxml using esl On 2 Oct 2013 15:59, "crazygabry" wrote: > Hello everyone, > > i'm doing a project and i need to edit xml (dialplan, users and so on) from > an external software placed on another server. > I saw that with mod_curl can read the dialplan or other configurations from > remote places but i don't want to introduce latency when freeswitch parses > the dialplan. > I came from asterisk and it was possible to change files with Event API, i > saw that with mod_event_socket (i think) it isn't possible to change conf > files. > > Can you suggest me the best way to change xml files from remote server? > > Thanks in advice > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/05f7e930/attachment.html From avi at avimarcus.net Wed Oct 2 21:31:42 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 2 Oct 2013 17:31:42 +0000 Subject: [Freeswitch-users] canceling a bridge command In-Reply-To: References: Message-ID: <000001417a39ad4e-d6d51eff-df65-412d-91f8-cee21ff3b150-000000@email.amazonses.com> Something like: Will send the call to an IVR called new_destination when it catches a **. Even if that's in the middle of a call, the other end will get hung up on. I presume it works during a bridge, too. Check the wiki for how to execute other options. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/99d463c1/attachment.html From luis.daniel.lucio at gmail.com Wed Oct 2 21:34:06 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 2 Oct 2013 13:34:06 -0400 Subject: [Freeswitch-users] FAX nightmare continues In-Reply-To: <524C4322.5060406@xtronics.com> References: <524C4322.5060406@xtronics.com> Message-ID: Try thiese params, it works for me. but i dont have grandstream appliances, so i dont know if it will work for you originate {ignore_early_media=true,absolute_codec_string='PCMU,PCMA',fax_enable_t38=false,fax_verbose=true,fax_use_ecm=false,fax_enable_t38_request=true}sofia/gateway/myGW/myE164num &txfax(//tmp/16138007358 at faxofaxo.com.16138007358.25504/test.pdf.tiff) originate {ignore_early_media=true,absolute_codec_string='PCMU,PCMA',origination_caller_id_name='LDLQ',origination_caller_id_number=1234567890,fax_enable_t38=false,fax_retry_attempts=32,fax_retry_limit=32,fax_retry_sleep=180,fax_verbose=true,fax_use_ecm=off,fax_enable_t38_request=false}sofia/gateway/myGW/myE164NUM &txfax(//tmp/test.pdf.tiff) I doscover that some carriers, they need origination_caller_id for fax in order to work, such as flowroute for example. Maybe 2013/10/2 Karl Schmidt : > > This was tested and worked, but fails in the real world. > > I have a landline that brings in FAX to a grandstream gxw4104 - which gets connected to a gxw4008 - > to a modem to receive faxes via hylafax. > > There must be too much distortion as I'm getting lots of errors in hylafax and it finally drops the > call. > > I have PCMU on both of the grandstream boxes so I'm thinking that freeswitch should just pass the data. > > Are there any other settings I might fiddle with? > > > ,.,.,.,.,.,.,.,.,.,. > > This also brings up a question about echo suppression - there are places in the grandstreams for > echo suppression (also in the polycom phones). Should both ends do echo cancellation or just one > end? What is the best practice for setting echo cancellation? > > > > > -------------------------------------------------------------------------------- > Karl Schmidt EMail Karl at xtronics.com > Transtronics, Inc. WEB http://secure.transtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 FAX (785) 841-0434 > > Ask any politician in private, It's a blast > spending other peoples money. > -------------------------------------------------------------------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From william.king at quentustech.com Wed Oct 2 21:55:00 2013 From: william.king at quentustech.com (William King) Date: Wed, 02 Oct 2013 10:55:00 -0700 Subject: [Freeswitch-users] Freeswitch+Radius+IAS/NPS In-Reply-To: References: Message-ID: <524C5DF4.8050000@quentustech.com> I have not configured mod_xml_radius against either of those systems, but feel free to contact me on or off list and I might be able to assist you. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 10/01/2013 04:50 AM, Elton Machado wrote: > Does anyone have or know how to setup freeswitch radius module to work > against NPS or IAS? > > I'm try to use the documentation for the freeradius but I'm not > succeeding in doing it well. > > TIA, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mario_fs at mgtech.com Wed Oct 2 21:19:54 2013 From: mario_fs at mgtech.com (Mario G) Date: Wed, 2 Oct 2013 10:19:54 -0700 Subject: [Freeswitch-users] OS X works with tarball, not with 1.2 stable git? In-Reply-To: <5CDF898D-37D9-4303-8E13-85F7A6D8653A@insensate.co.uk> References: <9FDCB978-A0F0-4F14-ADF7-7A4EF367F0EE@jerris.com> <5CDF898D-37D9-4303-8E13-85F7A6D8653A@insensate.co.uk> Message-ID: <83CEF362-88EA-422A-9253-4B4AB35339ED@mgtech.com> Lawrence, you should track Jira FS-5819 which I started for Xcode 5. It has patches Mike already made, two more to go to get a clean make. Mario G On Oct 2, 2013, at 3:53 AM, Lawrence Conroy wrote: > Hi Mike, > I had a play with Xcode 5 on 10.8.5 yesterday -- got this to build after a few hacks (compiler is a lot more picky so flagged up things like testing a enumeration-typed variable against 9999, ...). > Hacking in changes to the code to quiet those meant it built, but has the same behaviour as described when run -- seems to lose interest in worker thread, whilst still responding to incoming messages as expected. > It DID give me good hints on an underlying residual lurky in the portaudio fix of FS-4786, so worth the fun. > all the best, > Lawrence > > On 2 Oct 2013, at 02:31, Michael Jerris wrote: >> I pulled down the latest xcode today, I'll be working on getting the build with latest compilers working this week, i know thats an issue, otherwise I have not had anyone say they were having issues with osx, so I'll see if I get the same when I get this all building again. >> >> Mike >> >> On Oct 1, 2013, at 8:04 PM, Ken Rice wrote: >> >>> Yes there is a git tag... >>> >>> git checkout v1.2.13 from your git tree to get the version that's in the >>> tarball >>> >>> If that doesn't build theres probably an issue that needs to be looked at in >>> how its getting bootstrapped on your system and a ticket w/ a full log >>> (stderr and stdout) attached to the ticket as a .txt file will hopefully >>> shed some light on the why >>> >>> On 10/1/13 6:26 PM, "Lawrence Conroy" wrote: >>> >>>> Hi Folks, >>>> has anyone managed to get the latest git 1.2 stable branch to run correctly >>>> on OS X 10.6.8? >>>> >>>> I have no problems getting the 1.2.13 tarball to build and run, but ... >>>> >>>> With the 1.2 stable git (as of today), whilst fs builds OK, when run sofia >>>> seems to accept regs from clients, will accept an INVITE from the client and >>>> send back the 100, set up a worker for the call, but then ignore it. >>>> Result is the client sits spinning its wheels waiting for a final response, >>>> and getting nothing. >>>> Trying to shutdown fs meets a wait for the worker thread to quit, and >>>> eventually a time out when the parent gives up. >>>> >>>> Like I said, this works fine with the tarball. Is there a tagged git release >>>> equivalent to the tarball? >>>> Without one it's a challenge to track back to find where the problem raised >>>> its head. >>>> [yup, I am a tyro at this new-fangled git stuff; this may be obvious, but not >>>> to me ?] >>>> >>>> all the best, >>>> Lawrence >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vipkilla at gmail.com Wed Oct 2 22:43:54 2013 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 2 Oct 2013 14:43:54 -0400 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> Message-ID: Use mod_xml_curl to change configurations, directory, dialplan, etc... On Wed, Oct 2, 2013 at 1:02 PM, Callum Guy wrote: > You should be able to remotely run reloadxml using esl > On 2 Oct 2013 15:59, "crazygabry" wrote: > >> Hello everyone, >> >> i'm doing a project and i need to edit xml (dialplan, users and so on) >> from >> an external software placed on another server. >> I saw that with mod_curl can read the dialplan or other configurations >> from >> remote places but i don't want to introduce latency when freeswitch parses >> the dialplan. >> I came from asterisk and it was possible to change files with Event API, i >> saw that with mod_event_socket (i think) it isn't possible to change conf >> files. >> >> Can you suggest me the best way to change xml files from remote server? >> >> Thanks in advice >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/42cce1b4/attachment.html From crazygabry at gmail.com Wed Oct 2 23:53:28 2013 From: crazygabry at gmail.com (crazygabry) Date: Wed, 2 Oct 2013 12:53:28 -0700 (PDT) Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> Message-ID: <1380743608115-7595424.post@n2.nabble.com> So you suggest to have a server witch Apache that contains all dialplans, directories and so on and let freeswitch ask to him while parsing xml right? Thank you very much -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595424.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vipkilla at gmail.com Thu Oct 3 00:05:06 2013 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 2 Oct 2013 16:05:06 -0400 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: <1380743608115-7595424.post@n2.nabble.com> References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: yeah something like that... i use nginx + php5-fpm to generate the XML though :) On Wed, Oct 2, 2013 at 3:53 PM, crazygabry wrote: > So you suggest to have a server witch Apache that contains all dialplans, > directories and so on and let freeswitch ask to him while parsing xml > right? > > Thank you very much > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595424.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/9b9c512b/attachment.html From grcamauer at gmail.com Thu Oct 3 00:31:35 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 2 Oct 2013 17:31:35 -0300 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: How about a shared directory? Or rcp the files? You can have a program that writes the XML to the shared directory and then connect via ESL and execute a reloadxml as suggested by Callum above. This avoids any delays from http requests or maintaining a HTTP server. I guess it would depend on how many Freeswitch servers you are configuring this way, and how often you are changing the XML files. Guillermo On Wed, Oct 2, 2013 at 5:05 PM, Vik Killa wrote: > yeah something like that... i use nginx + php5-fpm to generate the XML > though :) > > > On Wed, Oct 2, 2013 at 3:53 PM, crazygabry wrote: > >> So you suggest to have a server witch Apache that contains all dialplans, >> directories and so on and let freeswitch ask to him while parsing xml >> right? >> >> Thank you very much >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595424.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/da9fd8fc/attachment-0001.html From crazygabry at gmail.com Thu Oct 3 00:34:30 2013 From: crazygabry at gmail.com (crazygabry) Date: Wed, 2 Oct 2013 13:34:30 -0700 (PDT) Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: <1380746070244-7595426.post@n2.nabble.com> Thank you i'll try this weekend and nginx + php5-fpm :) i'll let you know Thanks to you and to Callum Guy Gabriele -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595426.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ben at langfeld.co.uk Thu Oct 3 00:42:54 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 2 Oct 2013 17:42:54 -0300 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: That's called configuration management and you should use a proper tool from it, such as Chef or Puppet, rather than a homebrew collection of scripts. The next guy will thank you. On 2 October 2013 17:31, Guillermo Ruiz Camauer wrote: > How about a shared directory? Or rcp the files? You can have a program > that writes the XML to the shared directory and then connect via ESL and > execute a reloadxml as suggested by Callum above. This avoids any delays > from http requests or maintaining a HTTP server. I guess it would depend > on how many Freeswitch servers you are configuring this way, and how often > you are changing the XML files. > > Guillermo > > > On Wed, Oct 2, 2013 at 5:05 PM, Vik Killa wrote: > >> yeah something like that... i use nginx + php5-fpm to generate the XML >> though :) >> >> >> On Wed, Oct 2, 2013 at 3:53 PM, crazygabry wrote: >> >>> So you suggest to have a server witch Apache that contains all dialplans, >>> directories and so on and let freeswitch ask to him while parsing xml >>> right? >>> >>> Thank you very much >>> >>> >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595424.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/15885ce6/attachment.html From steveayre at gmail.com Thu Oct 3 01:01:53 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 2 Oct 2013 22:01:53 +0100 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: +1 - I've used SFTP for this in the past to push the config onto the server, verified it's valid XML and uploaded correctly with xmllint & md5sum, and then done reloadxml over ESL. You could either push from the central server or pull from the freeswitch ones (perhaps in a cronjob). I've also heard of people storing their configs in git and doing a git clone into the appropriate directory on the servers. Then it's just 'git pull' + reloadxml (well, depending on the module - but that's all you need for dialplan changes). You can also get FreeSWITCH to use wget to download the dialplan over HTTP. See https://wiki.freeswitch.org/wiki/Mod_xml_curl#Alternative_ways_of_storing_static_configurations- I can't remember if exec actions work during a reloadxml though. Truth is there's a million and one ways to get the file onto the server. :) If the wget one works on reloadxml that would be the simplest I guess though. Might cause problems if there's any HTTP error though (500 Internal Server Error -> blank result -> no dialplan, whereas copying the file you can verify it before you reloadxml). On 2 October 2013 21:31, Guillermo Ruiz Camauer wrote: > How about a shared directory? Or rcp the files? You can have a program > that writes the XML to the shared directory and then connect via ESL and > execute a reloadxml as suggested by Callum above. This avoids any delays > from http requests or maintaining a HTTP server. I guess it would depend > on how many Freeswitch servers you are configuring this way, and how often > you are changing the XML files. > > Guillermo > > > On Wed, Oct 2, 2013 at 5:05 PM, Vik Killa wrote: > >> yeah something like that... i use nginx + php5-fpm to generate the XML >> though :) >> >> >> On Wed, Oct 2, 2013 at 3:53 PM, crazygabry wrote: >> >>> So you suggest to have a server witch Apache that contains all dialplans, >>> directories and so on and let freeswitch ask to him while parsing xml >>> right? >>> >>> Thank you very much >>> >>> >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595424.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/511731d4/attachment-0001.html From avi at avimarcus.net Thu Oct 3 01:22:43 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 2 Oct 2013 21:22:43 +0000 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: <000001417b0d2f38-c25f4aed-b4da-4a6d-875c-bd1eeb24f0c5-000000@email.amazonses.com> Or salt-stack. You can possibly put the configs in GIT and use a hook to trigger a reloadxml. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/55eb8145/attachment.html From andrew at cassidywebservices.co.uk Thu Oct 3 01:27:31 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 2 Oct 2013 22:27:31 +0100 Subject: [Freeswitch-users] vars.xml In-Reply-To: <524B061B.9050501@freeswitch.org> References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> <524B061B.9050501@freeswitch.org> Message-ID: Unrelated, but thanks for highlighting mod_translate, it's solved one of the big problems I was having in needing to translate numbers before they hit the xml dialplan (and therefore xml curl) On 1 October 2013 18:27, Raymond Chandler wrote: > The numbering_plan variable was originally set a few years back for the > concept of adding numbering plans to the default config set (i.e. for doing > local dialing rules for the US, UK, HK, etc). The idea was that we'd > include a US.xml dialplan that you could transfer the call into so that > numbers could be formatted into e.164 according to predefined rules. > > Since then, mod_translate was added, which will translate the numbers for > you in it's dialplan, using a dialplan app for the XML dialplan, or api > function call. Using a module instead of multiple dialplans is arguably > simpler, and more efficient. The sample config in mod_translate's conf/ > directory in the source, will show you an example usage which happens to > include the numbering_plan variable. > > > > > > > > > > > > > > There's also a very minimal wiki page at > https://wiki.freeswitch.org/wiki/Mod_translate that gives basic usage > examples. > > -Ray > > > On 10/01/2013 09:50 AM, Michael Jerris wrote: > > conf/vanilla/directory/default/default.xml:21: name="numbering_plan" value="$${default_country}"/> > conf/vanilla/vars.xml:222: data="default_country=US"/> > > Its used to set the numbering_plan variable which as far as I can tell > is unused. > > On Oct 1, 2013, at 9:42 AM, Richard Mace wrote: > > And, could I ask what it is used for ? > > Richard > On 1 Oct 2013 06:50, "Richard Mace" wrote: > >> Hi All, >> I see that in vars.xml there is an entry for Default Country. >> Where can I find out what each of the available Default Country options >> are please? >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/a0cee1a7/attachment.html From avi at avimarcus.net Thu Oct 3 01:41:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 2 Oct 2013 21:41:24 +0000 Subject: [Freeswitch-users] vars.xml In-Reply-To: References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> <524B061B.9050501@freeswitch.org> Message-ID: <000001417b1e4828-b12b5d93-feee-473d-bdda-0881ac1f817c-000000@email.amazonses.com> Can you update the docs for it, then? The wiki page looks incredibly sparse. -Avi On Thu, Oct 3, 2013 at 12:27 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Unrelated, but thanks for highlighting mod_translate, it's solved one of > the big problems I was having in needing to translate numbers before they > hit the xml dialplan (and therefore xml curl) > > > On 1 October 2013 18:27, Raymond Chandler wrote: > >> The numbering_plan variable was originally set a few years back for the >> concept of adding numbering plans to the default config set (i.e. for doing >> local dialing rules for the US, UK, HK, etc). The idea was that we'd >> include a US.xml dialplan that you could transfer the call into so that >> numbers could be formatted into e.164 according to predefined rules. >> >> Since then, mod_translate was added, which will translate the numbers for >> you in it's dialplan, using a dialplan app for the XML dialplan, or api >> function call. Using a module instead of multiple dialplans is arguably >> simpler, and more efficient. The sample config in mod_translate's conf/ >> directory in the source, will show you an example usage which happens to >> include the numbering_plan variable. >> >> >> >> >> >> >> >> >> >> >> >> >> >> There's also a very minimal wiki page at >> https://wiki.freeswitch.org/wiki/Mod_translate that gives basic usage >> examples. >> >> -Ray >> >> >> On 10/01/2013 09:50 AM, Michael Jerris wrote: >> >> conf/vanilla/directory/default/default.xml:21: > name="numbering_plan" value="$${default_country}"/> >> conf/vanilla/vars.xml:222: > data="default_country=US"/> >> >> Its used to set the numbering_plan variable which as far as I can tell >> is unused. >> >> On Oct 1, 2013, at 9:42 AM, Richard Mace wrote: >> >> And, could I ask what it is used for ? >> >> Richard >> On 1 Oct 2013 06:50, "Richard Mace" wrote: >> >>> Hi All, >>> I see that in vars.xml there is an entry for Default Country. >>> Where can I find out what each of the available Default Country options >>> are please? >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/6db7e978/attachment-0001.html From andrew at cassidywebservices.co.uk Thu Oct 3 02:23:04 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 2 Oct 2013 23:23:04 +0100 Subject: [Freeswitch-users] vars.xml In-Reply-To: <000001417b1e4828-b12b5d93-feee-473d-bdda-0881ac1f817c-000000@email.amazonses.com> References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> <524B061B.9050501@freeswitch.org> <000001417b1e4828-b12b5d93-feee-473d-bdda-0881ac1f817c-000000@email.amazonses.com> Message-ID: I shall indeed now I know how it works! Probably not tonight though, it's getting late. On 2 October 2013 22:41, Avi Marcus wrote: > Can you update the docs for it, then? The wiki page looks incredibly > sparse. > > -Avi > > On Thu, Oct 3, 2013 at 12:27 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> Unrelated, but thanks for highlighting mod_translate, it's solved one of >> the big problems I was having in needing to translate numbers before they >> hit the xml dialplan (and therefore xml curl) >> >> >> On 1 October 2013 18:27, Raymond Chandler wrote: >> >>> The numbering_plan variable was originally set a few years back for >>> the concept of adding numbering plans to the default config set (i.e. for >>> doing local dialing rules for the US, UK, HK, etc). The idea was that we'd >>> include a US.xml dialplan that you could transfer the call into so that >>> numbers could be formatted into e.164 according to predefined rules. >>> >>> Since then, mod_translate was added, which will translate the numbers >>> for you in it's dialplan, using a dialplan app for the XML dialplan, or api >>> function call. Using a module instead of multiple dialplans is arguably >>> simpler, and more efficient. The sample config in mod_translate's conf/ >>> directory in the source, will show you an example usage which happens to >>> include the numbering_plan variable. >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> There's also a very minimal wiki page at >>> https://wiki.freeswitch.org/wiki/Mod_translate that gives basic usage >>> examples. >>> >>> -Ray >>> >>> >>> On 10/01/2013 09:50 AM, Michael Jerris wrote: >>> >>> conf/vanilla/directory/default/default.xml:21: >> name="numbering_plan" value="$${default_country}"/> >>> conf/vanilla/vars.xml:222: >> data="default_country=US"/> >>> >>> Its used to set the numbering_plan variable which as far as I can tell >>> is unused. >>> >>> On Oct 1, 2013, at 9:42 AM, Richard Mace >>> wrote: >>> >>> And, could I ask what it is used for ? >>> >>> Richard >>> On 1 Oct 2013 06:50, "Richard Mace" wrote: >>> >>>> Hi All, >>>> I see that in vars.xml there is an entry for Default Country. >>>> Where can I find out what each of the available Default Country options >>>> are please? >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131002/30e2ecc2/attachment.html From crazygabry at gmail.com Thu Oct 3 03:31:53 2013 From: crazygabry at gmail.com (crazygabry) Date: Wed, 2 Oct 2013 16:31:53 -0700 (PDT) Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> Message-ID: <1380756713562-7595434.post@n2.nabble.com> I really like the SFTP Option... i'll probably try this first! any security suggestion? Thank you -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595434.html Sent from the freeswitch-users mailing list archive at Nabble.com. From richard.mace at gmail.com Thu Oct 3 10:32:24 2013 From: richard.mace at gmail.com (Richard Mace) Date: Thu, 3 Oct 2013 07:32:24 +0100 Subject: [Freeswitch-users] callgroup variable in user.xml Message-ID: Hi, Can anyone tell me what the callgroup variable is used for in the user.xml (1000.xml) file please? Does it relate to a "ring" group, or "call pickup" group? Thanks in advance. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/9253e6de/attachment.html From ssinyagin at yahoo.com Thu Oct 3 11:47:34 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 3 Oct 2013 00:47:34 -0700 (PDT) Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: <1380756713562-7595434.post@n2.nabble.com> References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> <1380756713562-7595434.post@n2.nabble.com> Message-ID: <1380786454.55009.YahooMailNeo@web126206.mail.ne1.yahoo.com> security suggestion is very simple: you need to make it secure :) ________________________________ From: crazygabry To: freeswitch-users at lists.freeswitch.org Sent: Thursday, October 3, 2013 1:31 AM Subject: Re: [Freeswitch-users] best way to change configuration files programmatically I really like the SFTP Option... i'll probably try this first! any security suggestion? Thank you -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595434.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/c147d8b1/attachment-0001.html From callum.guy at x-on.co.uk Thu Oct 3 12:22:58 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 3 Oct 2013 09:22:58 +0100 Subject: [Freeswitch-users] callgroup variable in user.xml In-Reply-To: References: Message-ID: Is this what you're looking for? https://wiki.freeswitch.org/wiki/Callgroup_intercept ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 3 October 2013 07:32, Richard Mace wrote: > Hi, > Can anyone tell me what the callgroup variable is used for in the user.xml > (1000.xml) file please? > Does it relate to a "ring" group, or "call pickup" group? > > Thanks in advance. > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/5d17ce8e/attachment.html From andrew at cassidywebservices.co.uk Thu Oct 3 12:37:11 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 3 Oct 2013 09:37:11 +0100 Subject: [Freeswitch-users] callgroup variable in user.xml In-Reply-To: References: Message-ID: Hi, I think it'd be wise to point out that many of these variables relate directly to the default configuration and in most cases are only used within the default dialplan and other configuration files. I have found one or two exceptions for this rule (for example default_country in vars.xml is used directly by mod_translate) So whilst those variables are used in call pickup as Callum said, they're not required to use call pickup, you could implement that in a completely different way if you wanted to. On 3 October 2013 09:22, Callum Guy wrote: > Is this what you're looking for? > > https://wiki.freeswitch.org/wiki/Callgroup_intercept > > ______________________________ > > Callum Guy > Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 3 October 2013 07:32, Richard Mace wrote: > >> Hi, >> Can anyone tell me what the callgroup variable is used for in the >> user.xml (1000.xml) file please? >> Does it relate to a "ring" group, or "call pickup" group? >> >> Thanks in advance. >> >> Richard >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/3326c928/attachment.html From ssinyagin at yahoo.com Thu Oct 3 12:37:42 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 3 Oct 2013 01:37:42 -0700 (PDT) Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: <1380786454.55009.YahooMailNeo@web126206.mail.ne1.yahoo.com> References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> <1380756713562-7595434.post@n2.nabble.com> <1380786454.55009.YahooMailNeo@web126206.mail.ne1.yahoo.com> Message-ID: <1380789462.15515.YahooMailNeo@web126202.mail.ne1.yahoo.com> speaking more seriously, Git with SSH transport and public key authentication is your best choice: in this case, both Git repository and the client servers can be in the public network without much concern. "Gitolite" is a nice software package for an easy-to-setup private Git repository. Also with Git, you can easily track the changes and roll back quickly if necessary. You can also set up hooks which will notify the client servers that the new update is available and needs to be pulled. ________________________________ From: Stanislav Sinyagin To: FreeSWITCH Users Help Sent: Thursday, October 3, 2013 9:47 AM Subject: Re: [Freeswitch-users] best way to change configuration files programmatically security suggestion is very simple: you need to make it secure :) ________________________________ From: crazygabry To: freeswitch-users at lists.freeswitch.org Sent: Thursday, October 3, 2013 1:31 AM Subject: Re: [Freeswitch-users] best way to change configuration files programmatically I really like the SFTP Option... i'll probably try this first! any security suggestion? Thank you -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595434.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/93865fca/attachment-0001.html From andrew at cassidywebservices.co.uk Thu Oct 3 13:04:30 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 3 Oct 2013 10:04:30 +0100 Subject: [Freeswitch-users] vars.xml In-Reply-To: References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> <524B061B.9050501@freeswitch.org> <000001417b1e4828-b12b5d93-feee-473d-bdda-0881ac1f817c-000000@email.amazonses.com> Message-ID: Ok Updated. As far as I can tell from reading the code it's mostly correct. https://wiki.freeswitch.org/wiki/Mod_translate On 2 October 2013 23:23, Andrew Cassidy wrote: > I shall indeed now I know how it works! Probably not tonight though, it's > getting late. > > > On 2 October 2013 22:41, Avi Marcus wrote: > >> Can you update the docs for it, then? The wiki page looks incredibly >> sparse. >> >> -Avi >> >> On Thu, Oct 3, 2013 at 12:27 AM, Andrew Cassidy < >> andrew at cassidywebservices.co.uk> wrote: >> >>> Unrelated, but thanks for highlighting mod_translate, it's solved one of >>> the big problems I was having in needing to translate numbers before they >>> hit the xml dialplan (and therefore xml curl) >>> >>> >>> On 1 October 2013 18:27, Raymond Chandler wrote: >>> >>>> The numbering_plan variable was originally set a few years back for >>>> the concept of adding numbering plans to the default config set (i.e. for >>>> doing local dialing rules for the US, UK, HK, etc). The idea was that we'd >>>> include a US.xml dialplan that you could transfer the call into so that >>>> numbers could be formatted into e.164 according to predefined rules. >>>> >>>> Since then, mod_translate was added, which will translate the numbers >>>> for you in it's dialplan, using a dialplan app for the XML dialplan, or api >>>> function call. Using a module instead of multiple dialplans is arguably >>>> simpler, and more efficient. The sample config in mod_translate's conf/ >>>> directory in the source, will show you an example usage which happens to >>>> include the numbering_plan variable. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> There's also a very minimal wiki page at >>>> https://wiki.freeswitch.org/wiki/Mod_translate that gives basic usage >>>> examples. >>>> >>>> -Ray >>>> >>>> >>>> On 10/01/2013 09:50 AM, Michael Jerris wrote: >>>> >>>> conf/vanilla/directory/default/default.xml:21: >>> name="numbering_plan" value="$${default_country}"/> >>>> conf/vanilla/vars.xml:222: >>> data="default_country=US"/> >>>> >>>> Its used to set the numbering_plan variable which as far as I can >>>> tell is unused. >>>> >>>> On Oct 1, 2013, at 9:42 AM, Richard Mace >>>> wrote: >>>> >>>> And, could I ask what it is used for ? >>>> >>>> Richard >>>> On 1 Oct 2013 06:50, "Richard Mace" wrote: >>>> >>>>> Hi All, >>>>> I see that in vars.xml there is an entry for Default Country. >>>>> Where can I find out what each of the available Default Country >>>>> options are please? >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>> Managing Director >>> >>> >>> *T *03300 100 960 *F >>> *03300 100 961 >>> *E *andrew at cassidywebservices.co.uk >>> *W *www.cassidywebservices.co.uk >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/b8833f51/attachment.html From andrew at cassidywebservices.co.uk Thu Oct 3 13:10:07 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 3 Oct 2013 10:10:07 +0100 Subject: [Freeswitch-users] best way to change configuration files programmatically In-Reply-To: <1380789462.15515.YahooMailNeo@web126202.mail.ne1.yahoo.com> References: <1380725722254-7595412.post@n2.nabble.com> <1380743608115-7595424.post@n2.nabble.com> <1380756713562-7595434.post@n2.nabble.com> <1380786454.55009.YahooMailNeo@web126206.mail.ne1.yahoo.com> <1380789462.15515.YahooMailNeo@web126202.mail.ne1.yahoo.com> Message-ID: +1 for xml_curl, except I use nginx + uwsgi + django with as postgres backend. On 3 October 2013 09:37, Stanislav Sinyagin wrote: > speaking more seriously, > > Git with SSH transport and public key authentication is your best choice: > in this case, both Git repository and the client servers can be in the > public network without much concern. > > "Gitolite" is a nice software package for an easy-to-setup private Git > repository. > > Also with Git, you can easily track the changes and roll back quickly if > necessary. You can also set up hooks which will notify the client servers > that the new update is available and needs to be pulled. > > > > > ------------------------------ > *From:* Stanislav Sinyagin > *To:* FreeSWITCH Users Help > *Sent:* Thursday, October 3, 2013 9:47 AM > > *Subject:* Re: [Freeswitch-users] best way to change configuration files > programmatically > > security suggestion is very simple: you need to make it secure :) > > > ------------------------------ > *From:* crazygabry > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, October 3, 2013 1:31 AM > *Subject:* Re: [Freeswitch-users] best way to change configuration files > programmatically > > I really like the SFTP Option... i'll probably try this first! any security > suggestion? > > Thank you > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/best-way-to-change-configuration-files-programmatically-tp7595412p7595434.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/35c2a59c/attachment-0001.html From avi at avimarcus.net Thu Oct 3 13:14:48 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 3 Oct 2013 09:14:48 +0000 Subject: [Freeswitch-users] vars.xml In-Reply-To: References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> <524B061B.9050501@freeswitch.org> <000001417b1e4828-b12b5d93-feee-473d-bdda-0881ac1f817c-000000@email.amazonses.com> Message-ID: <000001417d991e63-9899bd95-a2e9-44cb-8beb-d64a336a60b0-000000@email.amazonses.com> Thanks! What happens after you call the translate app, it starts the dial plan over again? (like a transfer?) -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/3a24bc4f/attachment.html From andrew at cassidywebservices.co.uk Thu Oct 3 14:20:35 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 3 Oct 2013 11:20:35 +0100 Subject: [Freeswitch-users] vars.xml In-Reply-To: <000001417d991e63-9899bd95-a2e9-44cb-8beb-d64a336a60b0-000000@email.amazonses.com> References: <35F0F09A-E15D-4C0E-B417-DBB2B0A1D291@jerris.com> <524B061B.9050501@freeswitch.org> <000001417b1e4828-b12b5d93-feee-473d-bdda-0881ac1f817c-000000@email.amazonses.com> <000001417d991e63-9899bd95-a2e9-44cb-8beb-d64a336a60b0-000000@email.amazonses.com> Message-ID: Good question (a point I missed). It stores the result in a channel variable called translated On 3 October 2013 10:14, Avi Marcus wrote: > Thanks! > > What happens after you call the translate app, it starts the dial plan > over again? (like a transfer?) > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/4983da62/attachment.html From mehroz.ashraf85 at gmail.com Thu Oct 3 14:59:08 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 3 Oct 2013 03:59:08 -0700 (PDT) Subject: [Freeswitch-users] Update SDP of Leg-B Message-ID: <1380797948678-7595445.post@n2.nabble.com> Hi All, I am facing audio/video and video/audio switching issue under proxy media mode. It looks as of SDP does not update itself on the B-leg. OR if it does, it doesn't transmit video RTP after a switch from audio only. With the default media mode, this is achievable but , it b-Leg doesn't see any call update and hence pauses the video . 1) is there a way to explicitly change the SDP to be transmitted ? (proxy mode) 2) Any trigger or event at the B-leg to update the call , one A has done so (default mode) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-of-Leg-B-tp7595445.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Thu Oct 3 14:59:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 3 Oct 2013 11:59:57 +0100 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: Message-ID: When you see context mentioned it'll mean a dialplan context - how to separate different types of extension. Multitenant systems may share a single context or have a context for each domain, or anywhere between the two. That's up to you On Tuesday, October 1, 2013, Phil Mickelson wrote: > Mick, > > Thank you for your list. I've search those also. Let me give you a quick > example where I actually found the option but the docs don't supply any > additional information. > > I believe I need to create new VM profiles since I'm running multi-tenant. > I figured that one out by looking at other commands that expect a > voicemail_profile value. In searching through the default profile I found > the parameter "callback-context." Since "context" is generally used in FS > in connection with multi-tenant I assume (perhaps incorrectly) that this is > important for me. So, I go in search of the parameter in the docs. And, > after much searching (I have gotten better at this over time) I find: > > callback-context default > > > > > That's it. The exact same line in the XML file. Doesn't explain > anything. And, if you look at most of those parameters you'll see the > exact same thing. > > Regards, > > Phil Mickelson > > > > Hi Phil, > > In addition to the books you mentioned, do the following wiki pages not provide everything you need (if they don't we are in trouble! ). > > API Commands > https://wiki.freeswitch.org/wiki/Mod_commands > > > Dialplan Applications > https://wiki.freeswitch.org/wiki/Mod_dptools > > > Channel Variables > https://wiki.freeswitch.org/wiki/Channel_Variables > > > XML Dialplan > https://wiki.freeswitch.org/wiki/Dialplan_XML > > > Documentation > https://wiki.freeswitch.org/wiki/Documentation > > > > Rgds, Mick > Tel/SMS. +44(0)7967 594432 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/2416e278/attachment.html From richard.mace at gmail.com Thu Oct 3 15:44:30 2013 From: richard.mace at gmail.com (Richard Mace) Date: Thu, 3 Oct 2013 12:44:30 +0100 Subject: [Freeswitch-users] callgroup variable in user.xml In-Reply-To: References: Message-ID: Thanks Andrew, Yes, I am trying to build a GUI based PBX using the default config files as a base, hence the way I am putting it together is by looking at these variables and settings and trying to translate them into a GUI in the easiest way from the users point of view. Thanks for your input. Richard On 3 Oct 2013 09:38, "Andrew Cassidy" wrote: > Hi, > > I think it'd be wise to point out that many of these variables relate > directly to the default configuration and in most cases are only used > within the default dialplan and other configuration files. I have found one > or two exceptions for this rule (for example default_country in vars.xml is > used directly by mod_translate) > > So whilst those variables are used in call pickup as Callum said, they're > not required to use call pickup, you could implement that in a completely > different way if you wanted to. > > > On 3 October 2013 09:22, Callum Guy wrote: > >> Is this what you're looking for? >> >> https://wiki.freeswitch.org/wiki/Callgroup_intercept >> >> ______________________________ >> >> Callum Guy >> Developer >> >> X-on >> Framlingham Technology Centre >> Station Road, Framlingham, >> Suffolk, IP13 9EZ >> >> T 0333 332 0116 >> E callum.guy at x-on.co.uk >> >> >> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales >> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD >> Company Registration No. 2578478 >> >> This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message >> is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from >> your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of >> the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have >> been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on >> are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. >> Please consider the environment before printing this email. >> >> >> >> On 3 October 2013 07:32, Richard Mace wrote: >> >>> Hi, >>> Can anyone tell me what the callgroup variable is used for in the >>> user.xml (1000.xml) file please? >>> Does it relate to a "ring" group, or "call pickup" group? >>> >>> Thanks in advance. >>> >>> Richard >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/c541c78c/attachment-0001.html From alipey at gmail.com Thu Oct 3 17:03:26 2013 From: alipey at gmail.com (Ali Pey) Date: Thu, 3 Oct 2013 09:03:26 -0400 Subject: [Freeswitch-users] canceling a bridge command In-Reply-To: <000001417a39ad4e-d6d51eff-df65-412d-91f8-cee21ff3b150-000000@email.amazonses.com> References: <000001417a39ad4e-d6d51eff-df65-412d-91f8-cee21ff3b150-000000@email.amazonses.com> Message-ID: Thank you Avi. I will give it a try and provide update here. Regards, Ali Pey On Wed, Oct 2, 2013 at 1:31 PM, Avi Marcus wrote: > Something like: > > > > Will send the call to an IVR called new_destination when it catches a **. > Even if that's in the middle of a call, the other end will get hung up on. > I presume it works during a bridge, too. > Check the wiki for how to execute other options. > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/e3be3e8f/attachment.html From dnsingh.dns at gmail.com Thu Oct 3 11:21:35 2013 From: dnsingh.dns at gmail.com (DN Singh) Date: Thu, 3 Oct 2013 12:51:35 +0530 Subject: [Freeswitch-users] Automate wancfg Message-ID: Hello list, Has anyone tried to automate the wancfg configuration with freeswitch? I have tried, but gets automatically configured with T1, even if I give it parameters for E1. I have done this by using wancfg_zaptel with some arguments. Please let me know if more details would be required. Cheers!! DN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/8c2931d7/attachment.html From haakan.eriksson at ipadio.com Thu Oct 3 15:41:47 2013 From: haakan.eriksson at ipadio.com (Viking) Date: Thu, 3 Oct 2013 04:41:47 -0700 (PDT) Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: <7972438173691683594@unknownmsgid> References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> <7972438173691683594@unknownmsgid> Message-ID: <1380800507493-7595446.post@n2.nabble.com> Hi, I had the same problem but the /solution/ was different. I checked the ports weren't in use etc and everything looked fine. So in desperation I changed the static IP addresses to local 127.0.0.1. And it worked sort of... Loads the profile but I still can't make any outgoing calls: 2013-10-03 12:22:57.831669 [DEBUG] sofia.c:5793 Channel sofia/outbound/077113062586 at sip.node4.co.uk entering state [calling][0] sres.c:2978 sres_query_report_error() sres(q=0x7f7da4047100): reporting error RECORD_ERR for NAPTR sip.node4.co.uk nta.c:10482 outgoing_query_srv() nta: for "sip.node4.co.uk" query "_sip._udp.sip.node4.co.uk" SRV sres.c:2810 sres_send_dns_query() sres_send_dns_query(0x7f7da4002960, 0x7f7da402a5d0) id=16542 SRV _sip._udp.sip.node4.co.uk (to [83.138.151.81]:53) nta.c:10578 outgoing_answer_srv() nta: _sip._udp.sip.node4.co.uk IN SRV 0 0 5060 83.166.160.240. (udp) nta.c:10687 outgoing_query_a() nta: for "sip.node4.co.uk" query "83.166.160.240." A sres.c:2810 sres_send_dns_query() sres_send_dns_query(0x7f7da4002960, 0x7f7da4044b50) id=16543 A 83.166.160.240. (to [83.138.151.81]:53) sres.c:2978 sres_query_report_error() sres(q=0x7f7da4044b50): reporting error NAME_ERR for A 83.166.160.240. nta.c:10687 outgoing_query_a() nta: for "sip.node4.co.uk" query "sip.node4.co.uk" A sres.c:2810 sres_send_dns_query() sres_send_dns_query(0x7f7da4002960, 0x7f7da4045f30) id=16544 A sip.node4.co.uk (to [83.138.151.81]:53) nta.c:10740 outgoing_answer_a() nta: sip.node4.co.uk. IN A 83.166.160.240 tport.c:3253 tport_tsend() tport_tsend(0x7f7da4004600) tpn = */83.166.160.240:5060 tport.c:4675 tport_by_addrinfo() tport_by_addrinfo(0x7f7da4004600): not found by name */83.166.160.240:5060 tport.c:3631 tport_send_fatal() tport_vsend(0x7f7da4004600): Invalid argument with (s=26 */83.166.160.240:5060) nta.c:8379 outgoing_print_tport_error() nta: INVITE (50063624): Invalid argument (22) with */[83.166.160.240]:5060 nua_session.c:4135 signal_call_state_change() nua(0x7f7dc4009480): call state changed: calling -> init nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f7dc4009480): removing session usage 2013-10-03 12:22:57.871683 [DEBUG] switch_core_session.c:1006 Send signal sofia/outbound/07713062586 at sip.node4.co.uk [BREAK] 2013-10-03 12:22:57.871683 [DEBUG] switch_core_session.c:1006 Send signal sofia/outbound/07713062586 at sip.node4.co.uk [BREAK] 2013-10-03 12:22:57.871683 [DEBUG] switch_core_session.c:1006 Send signal sofia/outbound/07713062586 at sip.node4.co.uk [BREAK] 2013-10-03 12:22:57.871683 [DEBUG] sofia.c:5793 Channel sofia/outbound/07713062586 at sip.node4.co.uk entering state [terminated][503] 2013-10-03 12:22:57.871683 [NOTICE] sofia.c:6601 Hangup sofia/outbound/07713062586 at sip.node4.co.uk [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] I can ping the server etc so I really have no idea what the problem is..... Anyone have any idea? I am running Centos 6 and FreeSwitch 1.2.7. reg./H -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Creating-SIP-UA-for-profile-X-tp7201299p7595446.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Thu Oct 3 17:55:18 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Oct 2013 09:55:18 -0400 Subject: [Freeswitch-users] Update SDP of Leg-B In-Reply-To: <1380797948678-7595445.post@n2.nabble.com> References: <1380797948678-7595445.post@n2.nabble.com> Message-ID: <69A4FC56-864E-4E8C-A87A-6589CC82BF56@jerris.com> Proxy mode should probably never be used anymore. There is no reason for it. Can you open a bug for the issue in default media mode. Make sure to include sip trace and full debug logs. Thanks Mike On Oct 3, 2013, at 6:59 AM, mehroz wrote: > Hi All, > > I am facing audio/video and video/audio switching issue under proxy media > mode. > It looks as of SDP does not update itself on the B-leg. OR if it does, it > doesn't transmit video RTP after a switch from audio only. > > With the default media mode, this is achievable but , it b-Leg doesn't see > any call update and hence pauses the video . > > 1) is there a way to explicitly change the SDP to be transmitted ? (proxy > mode) > 2) Any trigger or event at the B-leg to update the call , one A has done so > (default mode) > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-of-Leg-B-tp7595445.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lloyd.aloysius at gmail.com Thu Oct 3 18:00:50 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 3 Oct 2013 10:00:50 -0400 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: <1380800507493-7595446.post@n2.nabble.com> References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> <7972438173691683594@unknownmsgid> <1380800507493-7595446.post@n2.nabble.com> Message-ID: Try the following, in your vars.xml, add the following line and replace A.B.C.D with your ip address. stop and start the switch Lloyd * * On Thu, Oct 3, 2013 at 7:41 AM, Viking wrote: > Hi, > I had the same problem but the /solution/ was different. > I checked the ports weren't in use etc and everything looked fine. > > So in desperation I changed the static IP addresses to local 127.0.0.1. And > it worked sort of... > Loads the profile but I still can't make any outgoing calls: > > 2013-10-03 12:22:57.831669 [DEBUG] sofia.c:5793 Channel > sofia/outbound/077113062586 at sip.node4.co.uk entering state [calling][0] > sres.c:2978 sres_query_report_error() sres(q=0x7f7da4047100): reporting > error RECORD_ERR for NAPTR sip.node4.co.uk > nta.c:10482 outgoing_query_srv() nta: for "sip.node4.co.uk" query > "_sip._udp.sip.node4.co.uk" SRV > sres.c:2810 sres_send_dns_query() sres_send_dns_query(0x7f7da4002960, > 0x7f7da402a5d0) id=16542 SRV _sip._udp.sip.node4.co.uk (to > [83.138.151.81]:53) > nta.c:10578 outgoing_answer_srv() nta: _sip._udp.sip.node4.co.uk IN SRV 0 > 0 > 5060 83.166.160.240. (udp) > nta.c:10687 outgoing_query_a() nta: for "sip.node4.co.uk" query > "83.166.160.240." A > sres.c:2810 sres_send_dns_query() sres_send_dns_query(0x7f7da4002960, > 0x7f7da4044b50) id=16543 A 83.166.160.240. (to [83.138.151.81]:53) > sres.c:2978 sres_query_report_error() sres(q=0x7f7da4044b50): reporting > error NAME_ERR for A 83.166.160.240. > nta.c:10687 outgoing_query_a() nta: for "sip.node4.co.uk" query > "sip.node4.co.uk" A > sres.c:2810 sres_send_dns_query() sres_send_dns_query(0x7f7da4002960, > 0x7f7da4045f30) id=16544 A sip.node4.co.uk (to [83.138.151.81]:53) > nta.c:10740 outgoing_answer_a() nta: sip.node4.co.uk. IN A 83.166.160.240 > tport.c:3253 tport_tsend() tport_tsend(0x7f7da4004600) tpn = > */83.166.160.240:5060 > tport.c:4675 tport_by_addrinfo() tport_by_addrinfo(0x7f7da4004600): not > found by name */83.166.160.240:5060 > tport.c:3631 tport_send_fatal() tport_vsend(0x7f7da4004600): Invalid > argument with (s=26 */83.166.160.240:5060) > nta.c:8379 outgoing_print_tport_error() nta: INVITE (50063624): Invalid > argument (22) with */[83.166.160.240]:5060 > nua_session.c:4135 signal_call_state_change() nua(0x7f7dc4009480): call > state changed: calling -> init > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f7dc4009480): removing > session usage > 2013-10-03 12:22:57.871683 [DEBUG] switch_core_session.c:1006 Send signal > sofia/outbound/07713062586 at sip.node4.co.uk [BREAK] > 2013-10-03 12:22:57.871683 [DEBUG] switch_core_session.c:1006 Send signal > sofia/outbound/07713062586 at sip.node4.co.uk [BREAK] > 2013-10-03 12:22:57.871683 [DEBUG] switch_core_session.c:1006 Send signal > sofia/outbound/07713062586 at sip.node4.co.uk [BREAK] > 2013-10-03 12:22:57.871683 [DEBUG] sofia.c:5793 Channel > sofia/outbound/07713062586 at sip.node4.co.uk entering state > [terminated][503] > 2013-10-03 12:22:57.871683 [NOTICE] sofia.c:6601 Hangup > sofia/outbound/07713062586 at sip.node4.co.uk [CS_CONSUME_MEDIA] > [NORMAL_TEMPORARY_FAILURE] > > I can ping the server etc so I really have no idea what the problem is..... > Anyone have any idea? > I am running Centos 6 and FreeSwitch 1.2.7. > > reg./H > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Error-Creating-SIP-UA-for-profile-X-tp7201299p7595446.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/20a637bd/attachment-0001.html From callum.guy at x-on.co.uk Thu Oct 3 18:04:54 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 3 Oct 2013 15:04:54 +0100 Subject: [Freeswitch-users] Conference Configure Multiple Prompts In-Reply-To: References: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> Message-ID: Hi All, I am still having problems implementing a conference announcement where i need to play two files in sequence. For clarity i want to play the joiners (or leavers) name, followed by "has joined/left the conference". I have tried channel variables in the conference XML. This works great for a single file so i can play the name of the joiner/leaver to the conference but not both files in sequence. I have tried phrases as suggested by Michael but these are incompatible with the conference XML (if i understand correctly). I also attempted to use these with uuid_broadcast however that does not interrupt the conference. My most recent attempt has been to use the conference play API (i.e. "api conference test play /usr/share/sounds/ConferenceJoined.wav async") where I am having some success. The problem is that this configuration is awkward because it won't play a set of files either and often due to suspected timing issues one of the play events doesn't happen at all! Overall i have to say that using channel variables in the XML for the enter sound () is the only stable option but I really do need to be able to play two files if at all possible. I have considered concatenating the audio files but this doesn't seem like a clean solution. Does anyone have any ideas for me to pursue? Thanks, Callum ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 2 October 2013 17:31, Callum Guy wrote: > OK so i've run into another issue when trying uuid_broadcast. > > I'm running the following just after joining an existing conference: > > api uuid_broadcast [UUID] phrase:conferenceAnnounce:${ConferenceJoinName} > both > > This is having no immediate effect however when either leg closes they > hear the broadcast message set by the closing channel! > > I'm sure i'm doing something wrong and i'll revisit tomorrow! > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/1c9c45c1/attachment.html From ssinyagin at yahoo.com Thu Oct 3 18:28:26 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 3 Oct 2013 07:28:26 -0700 (PDT) Subject: [Freeswitch-users] Conference Configure Multiple Prompts In-Reply-To: References: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> Message-ID: <1380810506.48449.YahooMailNeo@web126206.mail.ne1.yahoo.com> isn't the example at the bottom of http://wiki.freeswitch.org/wiki/Mod_conference doing exactly that? ________________________________ From: Callum Guy To: FreeSWITCH Users Help Sent: Thursday, October 3, 2013 4:04 PM Subject: Re: [Freeswitch-users] Conference Configure Multiple Prompts Hi All, I am still having problems implementing a conference announcement where i need to play two files in sequence. For clarity i want to play the joiners (or leavers) name, followed by "has joined/left the conference". I have tried channel variables in the conference XML. This works great for a single file so i can play the name of the joiner/leaver to the conference but not both files in sequence. I have tried phrases as suggested by Michael but these are incompatible with the conference XML (if i understand correctly). I also attempted to use these with uuid_broadcast however that does not interrupt the conference. My most recent attempt has been to use the conference play API (i.e. "api conference test play /usr/share/sounds/ConferenceJoined.wav async") where I am having some success. The problem is that this configuration is awkward because it won't play a set of files either and often due to suspected timing issues one of the play events doesn't happen at all! Overall i have to say that using channel variables in the XML for the enter sound () is the only stable option but I really do need to be able to play two files if at all possible. I have considered concatenating the audio files but this doesn't seem like a clean solution. Does anyone have any ideas for me to pursue? Thanks, Callum ______________________________Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 2 October 2013 17:31, Callum Guy wrote: OK so i've run into another issue when trying uuid_broadcast. > > >I'm running the following just after joining an existing conference: > > >api uuid_broadcast [UUID] phrase:conferenceAnnounce:${ConferenceJoinName} both > > > >This is having no immediate effect however when either leg closes they hear the broadcast message set by the closing channel! > > >I'm sure i'm doing something wrong and i'll revisit tomorrow! > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/bb135793/attachment-0001.html From dnsingh.dns at gmail.com Thu Oct 3 18:45:35 2013 From: dnsingh.dns at gmail.com (DN Singh) Date: Thu, 03 Oct 2013 20:15:35 +0530 Subject: [Freeswitch-users] FreeTDM rpm issue Message-ID: <524D830F.1030704@gmail.com> Hi list, I am installing freetdm library via yum repo in Centos 5. The problem is that even though it gets installed, I do not get ftmod_wanpipe.so which this file is supposed to provide. Can anybody help me on this issue? Would I have to compile and install freeswitch, just for this particular library? NOTE: I do have wanpipe and libsng_isdn already installed on it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/458dc6bb/attachment.html From mehroz.ashraf85 at gmail.com Thu Oct 3 18:47:37 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 3 Oct 2013 07:47:37 -0700 (PDT) Subject: [Freeswitch-users] Update SDP of Leg-B In-Reply-To: <69A4FC56-864E-4E8C-A87A-6589CC82BF56@jerris.com> References: <1380797948678-7595445.post@n2.nabble.com> <69A4FC56-864E-4E8C-A87A-6589CC82BF56@jerris.com> Message-ID: <1380811657483-7595455.post@n2.nabble.com> Yes Mike, Ill post traces. But lets say you want to carry your communication in proxy mode (when you want to communicate on codecs not supported by FS and when you want minimum processing of system, actually that is the requirement).. . I believe this is more a bug, than in default mode. Default media mode has FS in between and it is not passing the SDP to B-leg. I dont know whether it should or not as default media policies. Whats your say about that? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-of-Leg-B-tp7595445p7595455.html Sent from the freeswitch-users mailing list archive at Nabble.com. From manish.kumar.fs at gmail.com Thu Oct 3 18:51:07 2013 From: manish.kumar.fs at gmail.com (Manish Kumar) Date: Thu, 3 Oct 2013 20:21:07 +0530 Subject: [Freeswitch-users] FreeTDM rpm issue In-Reply-To: <524D830F.1030704@gmail.com> References: <524D830F.1030704@gmail.com> Message-ID: I too had this similar issue when I was configuring freeswitch with freetdm. But, unfortunately could not solve it. Maybe someone from the list can help. On Thu, Oct 3, 2013 at 8:15 PM, DN Singh wrote: > Hi list, > > I am installing freetdm library via yum repo in Centos 5. The problem is > that even though it gets installed, I do not get ftmod_wanpipe.so which > this file is supposed to provide. > > Can anybody help me on this issue? Would I have to compile and install > freeswitch, just for this particular library? > > NOTE: I do have wanpipe and libsng_isdn already installed on it. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/e326116d/attachment.html From mike at jerris.com Thu Oct 3 18:53:56 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Oct 2013 10:53:56 -0400 Subject: [Freeswitch-users] Update SDP of Leg-B In-Reply-To: <1380811657483-7595455.post@n2.nabble.com> References: <1380797948678-7595445.post@n2.nabble.com> <69A4FC56-864E-4E8C-A87A-6589CC82BF56@jerris.com> <1380811657483-7595455.post@n2.nabble.com> Message-ID: <20E751AC-3D82-4FEA-AEF9-AF3DDC790AAE@jerris.com> we can make stub passthrough codecs for any unsupported codecs, Proxy mode really doesn't in any real way reduce processing. On Oct 3, 2013, at 10:47 AM, mehroz wrote: > Yes Mike, Ill post traces. But lets say you want to carry your communication > in proxy mode (when you want to communicate on codecs not supported by FS > and when you want minimum processing of system, actually that is the > requirement).. . I believe this is more a bug, than in default mode. > > Default media mode has FS in between and it is not passing the SDP to B-leg. > I dont know whether it should or not as default media policies. Whats your > say about that? > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-of-Leg-B-tp7595445p7595455.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lloyd.aloysius at gmail.com Thu Oct 3 18:53:44 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 3 Oct 2013 10:53:44 -0400 Subject: [Freeswitch-users] mod_voicemail and non_numeric id Message-ID: Hi All I use non numeric id at user configuration(). All my dial plans in lua. *Problem :* when a user logged into his voicemail and try forward a message to another user's voicemail, mod_voicemail do not understand the extension number and always say invalid. Is there any way map extension number to user id , so that mod_voicemail can get the non_numeric id and forward the voicemail? Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/07ff3796/attachment.html From callum.guy at x-on.co.uk Thu Oct 3 19:01:04 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 3 Oct 2013 16:01:04 +0100 Subject: [Freeswitch-users] Conference Configure Multiple Prompts In-Reply-To: <1380810506.48449.YahooMailNeo@web126206.mail.ne1.yahoo.com> References: <168A1A13-2ECD-48BE-8551-CD4BA267E8ED@jerris.com> <1380810506.48449.YahooMailNeo@web126206.mail.ne1.yahoo.com> Message-ID: OK, wow, given the amount of time i've spent searching the wiki i'm surprised i hadn't already seen that! Presumably i had overlooked it as i'm working purely with ESL rather than XML dialplans. Anyway your suggestion pointed me in the right direction and the file string syntax gave me everything i needed to get things working. As such my conference XML shows: This solved the issue and is the nice clean solution i have been looking for. Thank you! ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 3 October 2013 15:28, Stanislav Sinyagin wrote: > isn't the example at the bottom of > http://wiki.freeswitch.org/wiki/Mod_conference > doing exactly that? > > > > > ------------------------------ > *From:* Callum Guy > *To:* FreeSWITCH Users Help > *Sent:* Thursday, October 3, 2013 4:04 PM > *Subject:* Re: [Freeswitch-users] Conference Configure Multiple Prompts > > Hi All, > > I am still having problems implementing a conference announcement where i > need to play two files in sequence. For clarity i want to play the joiners > (or leavers) name, followed by "has joined/left the conference". > > I have tried channel variables in the conference XML. This works great for > a single file so i can play the name of the joiner/leaver to the conference > but not both files in sequence. > > I have tried phrases as suggested by Michael but these are incompatible > with the conference XML (if i understand correctly). I also attempted to > use these with uuid_broadcast however that does not interrupt the > conference. > > My most recent attempt has been to use the conference play API (i.e. "api > conference test play /usr/share/sounds/ConferenceJoined.wav async") where I > am having some success. The problem is that this configuration is awkward > because it won't play a set of files either and often due to suspected > timing issues one of the play events doesn't happen at all! > > Overall i have to say that using channel variables in the XML for the > enter sound () is > the only stable option but I really do need to be able to play two files if > at all possible. I have considered concatenating the audio files but this > doesn't seem like a clean solution. > > Does anyone have any ideas for me to pursue? > > Thanks, > > Callum > > ______________________________ > > Callum Guy > Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 2 October 2013 17:31, Callum Guy wrote: > > OK so i've run into another issue when trying uuid_broadcast. > > I'm running the following just after joining an existing conference: > > api uuid_broadcast [UUID] phrase:conferenceAnnounce:${ConferenceJoinName} > both > > This is having no immediate effect however when either leg closes they > hear the broadcast message set by the closing channel! > > I'm sure i'm doing something wrong and i'll revisit tomorrow! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/5a8c96a9/attachment-0001.html From vipkilla at gmail.com Thu Oct 3 19:23:22 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 3 Oct 2013 11:23:22 -0400 Subject: [Freeswitch-users] sofia's SLA confusing caller_profiles Message-ID: im may be wrong about this but it is confusing.... when an outbound call (B-leg) is "stolen" via SLA (sofia's intercept, meaning that A-leg pushes hold and a new leg, C-leg, is created when another phone hit's the line key to bridge to the B-leg) the B-leg's caller_profile contains an originator_caller_profile of the new C-leg's caller_profile and the C-leg's caller_profile contains an originatee_caller_profile of B-leg's caller_profile this seems counter-intuitive and backwards compared to how it functions in all other scenarios shouldn't the original B-leg have an originatee_caller_profile and the new C-leg have an originator_caller_profile? since the C-leg is the NEW leg, it should have an originator, no? and since the B-leg already exists, it should have an originatee? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/98fe6831/attachment.html From msc at freeswitch.org Thu Oct 3 20:23:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Oct 2013 09:23:22 -0700 Subject: [Freeswitch-users] mod_voicemail and non_numeric id In-Reply-To: References: Message-ID: I would start here: https://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping Give that a whirl and let us know if you have success. -MC On Thu, Oct 3, 2013 at 7:53 AM, Lloyd Aloysius wrote: > Hi All > > I use non numeric id at user configuration(). All my dial > plans in lua. > > *Problem :* when a user logged into his voicemail and try forward a > message to another user's voicemail, mod_voicemail do not understand the > extension number and always say invalid. > > Is there any way map extension number to user id , so that mod_voicemail > can get the non_numeric id and forward the voicemail? > > > Thanks > > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/10bc9dd6/attachment.html From lloyd.aloysius at gmail.com Thu Oct 3 20:30:52 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 3 Oct 2013 12:30:52 -0400 Subject: [Freeswitch-users] mod_voicemail and non_numeric id In-Reply-To: References: Message-ID: Hi MC Thank you. I try that no luck.Then I look into the code, the code is not checking the number-alias. Can we add a check if id lookup failed check the number-alias? http://jira.freeswitch.org/browse/FS-5841 1726 status =* switch_xml_locate_user_merged(*"id"*, vm_cc, cbt->domain, NULL, &x_user, my_params); * 1727 switch_event_destroy(&my_params); 1728 1729 if (status != SWITCH_STATUS_SUCCESS) { 1730 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, 1731 "Failed to forward message - Cannot locate user %s@%s\n", vm_cc, cbt->domain); 9203183 1723 1732 TRY_CODE(switch_ivr_phrase_macro(session, VM_INVALID_EXTENSION_MACRO, vm_cc, NULL, NULL)); Thank you Lloyd On Thu, Oct 3, 2013 at 12:23 PM, Michael Collins wrote: > I would start here: > > > https://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping > > Give that a whirl and let us know if you have success. > -MC > > > > On Thu, Oct 3, 2013 at 7:53 AM, Lloyd Aloysius wrote: > >> Hi All >> >> I use non numeric id at user configuration(). All my >> dial plans in lua. >> >> *Problem :* when a user logged into his voicemail and try forward a >> message to another user's voicemail, mod_voicemail do not understand the >> extension number and always say invalid. >> >> Is there any way map extension number to user id , so that mod_voicemail >> can get the non_numeric id and forward the voicemail? >> >> >> Thanks >> >> Lloyd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/c0ce1091/attachment.html From jackal at cybershroud.net Thu Oct 3 20:48:20 2013 From: jackal at cybershroud.net (Carlos Flor) Date: Thu, 3 Oct 2013 12:48:20 -0400 Subject: [Freeswitch-users] Update SDP of Leg-B In-Reply-To: <20E751AC-3D82-4FEA-AEF9-AF3DDC790AAE@jerris.com> References: <1380797948678-7595445.post@n2.nabble.com> <69A4FC56-864E-4E8C-A87A-6589CC82BF56@jerris.com> <1380811657483-7595455.post@n2.nabble.com> <20E751AC-3D82-4FEA-AEF9-AF3DDC790AAE@jerris.com> Message-ID: Not to hijack this thread, but proxy mode is still needed for end-to-end zrtp, is it not? Your previous comment about there being no reason for it made and that it should probably never be used anymore made me wonder if there are plans of phasing support for proxy mode out of FS. On Thu, Oct 3, 2013 at 10:53 AM, Michael Jerris wrote: > we can make stub passthrough codecs for any unsupported codecs, Proxy mode > really doesn't in any real way reduce processing. > > On Oct 3, 2013, at 10:47 AM, mehroz wrote: > > > Yes Mike, Ill post traces. But lets say you want to carry your > communication > > in proxy mode (when you want to communicate on codecs not supported by FS > > and when you want minimum processing of system, actually that is the > > requirement).. . I believe this is more a bug, than in default mode. > > > > Default media mode has FS in between and it is not passing the SDP to > B-leg. > > I dont know whether it should or not as default media policies. Whats > your > > say about that? > > > > > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Update-SDP-of-Leg-B-tp7595445p7595455.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/33d696af/attachment-0001.html From msc at freeswitch.org Thu Oct 3 21:05:21 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Oct 2013 10:05:21 -0700 Subject: [Freeswitch-users] mod_voicemail and non_numeric id In-Reply-To: References: Message-ID: That's a bit outside my area of expertise, but I would suggest that you grep around and see if number-alias is used anywhere else in the code base. If it is then maybe you could emulate what it's doing and work up a patch for the devs to review. -MC On Thu, Oct 3, 2013 at 9:30 AM, Lloyd Aloysius wrote: > Hi MC > > Thank you. I try that no luck.Then I look into the code, the code is not > checking the number-alias. Can we add a check if id lookup failed check > the number-alias? > > http://jira.freeswitch.org/browse/FS-5841 > > > 1726 status =* switch_xml_locate_user_merged(*"id"*, vm_cc, cbt->domain, > NULL, &x_user, my_params); * > 1727 switch_event_destroy(&my_params); > 1728 > 1729 if (status != SWITCH_STATUS_SUCCESS) { > 1730 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), > SWITCH_LOG_WARNING, > 1731 "Failed to forward message - Cannot locate user %s@%s\n", vm_cc, > cbt->domain); > 9203183 > 1723 1732 TRY_CODE(switch_ivr_phrase_macro(session, > VM_INVALID_EXTENSION_MACRO, vm_cc, NULL, NULL)); > > > Thank you > Lloyd > > > On Thu, Oct 3, 2013 at 12:23 PM, Michael Collins wrote: > >> I would start here: >> >> >> https://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping >> >> Give that a whirl and let us know if you have success. >> -MC >> >> >> >> On Thu, Oct 3, 2013 at 7:53 AM, Lloyd Aloysius wrote: >> >>> Hi All >>> >>> I use non numeric id at user configuration(). All my >>> dial plans in lua. >>> >>> *Problem :* when a user logged into his voicemail and try forward a >>> message to another user's voicemail, mod_voicemail do not understand the >>> extension number and always say invalid. >>> >>> Is there any way map extension number to user id , so that mod_voicemail >>> can get the non_numeric id and forward the voicemail? >>> >>> >>> Thanks >>> >>> Lloyd >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/02456eee/attachment.html From andretodd at verizon.net Thu Oct 3 23:54:11 2013 From: andretodd at verizon.net (Andre) Date: Thu, 03 Oct 2013 15:54:11 -0400 Subject: [Freeswitch-users] park_after_bridge not working Message-ID: <0aaa01cec072$5538beb0$ffaa3c10$@verizon.net> Hi, My Park_after_bridge doesn't work if I add bypass_media=true. How can I get Park to work when I'm bypass_media mode? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/fe513e60/attachment.html From fcastelco at gmail.com Fri Oct 4 01:05:22 2013 From: fcastelco at gmail.com (Federico Castro) Date: Thu, 3 Oct 2013 18:05:22 -0300 Subject: [Freeswitch-users] Problem with NAT configuration. Trying to understand autonat. Message-ID: Hi everybody, please someone could help me to understand what "autonat:" makes when is used to modify ext-rtp-ip and ext-sip-ip in profiles. I faced the problem I'm going to explain and I solved it using "autonat" but I couldn't understand why it fixed my problem: *Problem:* Calls from SIP phones are dropped after 32 seconds *Scenario (not real IPs):* FS IP: 172.23.9.4 IP phone "A": 172.23.9.5 IP phone "B": 172.23.9.10 FS is behind NAT to call some VoIP providers. Public IP: 190.190.190.190 *What happens:* IP phone "A" calls IP phone "B", no NAT is necesary. All devices are in the same LAN, FS and both IP phones. I copy below profile configuration and 200 OK message that freeswitch sends to IP phone "A" when call is answered. Contact header is wrong, it is using Public IP instead of local IP. It produces that ACK from IP phone "A" never reaches FS and call is dropped by timeout. In profile configuration file: Then, I changed to: And it solved my problem. *My Questions:* Why when I dont use autonat "contact header" contains public IP if all devices are in the local network? And why autonat change this behaviour? Thanks for your help. Federico Castro *Profile configuration:* http://pastebin.com/X4DBg3yv *SIP 200 OK:* * * U 2013/10/03 16:22:35.536073 172.23.9.4:5060 -> 172.23.9.5:59453 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.23.9.4:59453 ;branch=z9hG4bK-d8754z-c07981061a655f7f-1---d8754z-;rport=59453. From: "somebody" ;tag=87348974. To: ;tag=1BQZX56QaNgZQ. Call-ID: M2RhY2EzNDA0YmQ5Y2E5YmJkYWIxYzI5ZTM0ZjkxZWE.. CSeq: 2 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.2.11+git~20130703T205557Z~60adf50f86. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 269. Remote-Party-ID: "2340" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1380800075 1380800076 IN IP4 172.23.9.4. s=FreeSWITCH. c=IN IP4 172.23.9.4. t=0 0. m=audio 28080 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. m=video 0 RTP/AVP 19. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/33fffaea/attachment-0001.html From rafaelstnoliveira at gmail.com Fri Oct 4 03:44:56 2013 From: rafaelstnoliveira at gmail.com (Rafael Santana) Date: Thu, 3 Oct 2013 20:44:56 -0300 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio Message-ID: Hi, I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. I've been trying to setup an environment where It can be possible to make a call through Google Chrome Browser using JsSIP to a standard phone device on PSTN. In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 instance and to get access to PSTN via this instance I had to register my Asterisk instance as a gateway on my Sofia's external profile. This part of my scenario works fine. I'm able to make calls using a softphone registered on FreeSwitch to standard phones on PSTN with no problems. What I wasn't able to do until now was the JsSIP + FreeSwitch integration. To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml. I really don't know if just this is sufficient. Am I missing something important? To connect on my FreeSwitch instance from Chrome, I'm using the Tryit JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a call to a PSTN phone. The connection is established but I don't get any audio in both endpoints. The same happens when I try to call the 5000 ivr extension or an user on a softphone at the same network from my Chrome browser. Assuming that all the services I've mentioned here are running on the same network, do you have any idea why I can't get audio in both endpoints of my experiment? Additional information: Ubuntu 12.04 64 bits FreeSwitch version 1.5.5 default install configuration Tryit JsSIP Demo with jssip-0.3.0.js Thanks in advance, Rafael. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/7af20f2d/attachment.html From ibk at labhijau.net Fri Oct 4 05:24:21 2013 From: ibk at labhijau.net (Iwan Budi Kusnanto) Date: Fri, 4 Oct 2013 08:24:21 +0700 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: On Friday, October 4, 2013, Rafael Santana wrote: > Hi, > > I'm new to telephony and FreeSwitch's world, so I apologize in advance for > any nonsense I speak here. > > I've been trying to setup an environment where It can be possible to make > a call through Google Chrome Browser using JsSIP to a standard phone device > on PSTN. > > In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't > chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 > instance and to get access to PSTN via this instance I had to register my > Asterisk instance as a gateway on my Sofia's external profile. This part of > my scenario works fine. I'm able to make calls using a softphone registered > on FreeSwitch to standard phones on PSTN with no problems. What I wasn't > able to do until now was the JsSIP + FreeSwitch integration. > > To setup FreeSwitch to comunicate with JsSIP, the only thing I did was > uncomment the line below on sip_profiles/internal.xml. > > > > I really don't know if just this is sufficient. Am I missing something > important? > > To connect on my FreeSwitch instance from Chrome, I'm using the Tryit > JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a > call to a PSTN phone. The connection is established but I don't get any > audio in both endpoints. The same happens when I try to call the 5000 ivr > extension or an user on a softphone at the same network from my Chrome > browser. > > Assuming that all the services I've mentioned here are running on the same > network, do you have any idea why I can't get audio in both endpoints of my > experiment? > > Additional information: > Ubuntu 12.04 64 bits > FreeSwitch version 1.5.5 default install configuration > Tryit JsSIP Demo with jssip-0.3.0.js > Try to add --with-openssl when doing ./configure > > Thanks in advance, > Rafael. > -- Iwan Budi Kusnanto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/69d5358d/attachment.html From max at nysolutions.com Fri Oct 4 05:35:36 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 4 Oct 2013 01:35:36 +0000 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: Did you open the websocket ports on your firewall? https://wiki.freeswitch.org/wiki/Firewall Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Iwan Budi Kusnanto Sent: Thursday, October 03, 2013 9:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio On Friday, October 4, 2013, Rafael Santana wrote: Hi, I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. I've been trying to setup an environment where It can be possible to make a call through Google Chrome Browser using JsSIP to a standard phone device on PSTN. In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 instance and to get access to PSTN via this instance I had to register my Asterisk instance as a gateway on my Sofia's external profile. This part of my scenario works fine. I'm able to make calls using a softphone registered on FreeSwitch to standard phones on PSTN with no problems. What I wasn't able to do until now was the JsSIP + FreeSwitch integration. To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml. I really don't know if just this is sufficient. Am I missing something important? To connect on my FreeSwitch instance from Chrome, I'm using the Tryit JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a call to a PSTN phone. The connection is established but I don't get any audio in both endpoints. The same happens when I try to call the 5000 ivr extension or an user on a softphone at the same network from my Chrome browser. Assuming that all the services I've mentioned here are running on the same network, do you have any idea why I can't get audio in both endpoints of my experiment? Additional information: Ubuntu 12.04 64 bits FreeSwitch version 1.5.5 default install configuration Tryit JsSIP Demo with jssip-0.3.0.js Try to add --with-openssl when doing ./configure Thanks in advance, Rafael. -- Iwan Budi Kusnanto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/ce78831d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/ce78831d/attachment-0001.jpg From karl at xtronics.com Fri Oct 4 06:41:39 2013 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 03 Oct 2013 21:41:39 -0500 Subject: [Freeswitch-users] Problem with NAT configuration. Trying to understand autonat. In-Reply-To: References: Message-ID: <524E2AE3.5040800@xtronics.com> On 10/03/2013 04:05 PM, Federico Castro wrote: > Hi everybody, please someone could help me to understand what "autonat:" makes when is used to > modify ext-rtp-ip and ext-sip-ip in profiles. > > I faced the problem I'm going to explain and I solved it using "autonat" but I couldn't understand > why it fixed my problem: > > I second the question. -- but first, I think auto-nat has a hyphen in it? I dug into this once and still have no idea what is going on - thus I have little hope of believing I have a secure configuration. particularly : sip_profiles/internal.xml: sip_profiles/internal.xml: and sip_profiles/external.xml: sip_profiles/external.xml: I guessed that auto-nat would return the correct IP address - but filling in the real IP address values didn't work as far as I could tell??? Thus, the following comment in the default file does not appear to be correct?? And global_getvar shows external_rtp_ip and external_sip_ip set to the correct IP, but external_sip_ip is not ext_sip_ip I'm not using NAT-PMP or UPNP - but it appears to work - I have no idea why. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 "If you don't read the newspaper you are uninformed, if you do read the newspaper you are misinformed." -- Mark Twain -------------------------------------------------------------------------------- From james.mortensen at synclio.com Fri Oct 4 08:51:14 2013 From: james.mortensen at synclio.com (James Mortensen) Date: Thu, 3 Oct 2013 21:51:14 -0700 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: Hi Rafael, You didn't mention whether the server was in the cloud. If you're server is on Amazon EC2, make sure you're following the guide here: https://wiki.freeswitch.org/wiki/Amazon_EC2 Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk server, do you see audio flowing? Also, in Chrome, startup chrome from the command line with the options to enable debug logging: chrome --enable-logging --v=11 Then look to see if there are STUN binding errors. Also, check chrome://webrtc-internals, which will also tell you if Chrome is trying to send audio. Is the server behind NAT or is it on the public Internet with it's own public IP bound to the eth0 interface? Hope this helps! James On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana wrote: > Hi, > > I'm new to telephony and FreeSwitch's world, so I apologize in advance for > any nonsense I speak here. > > I've been trying to setup an environment where It can be possible to make > a call through Google Chrome Browser using JsSIP to a standard phone device > on PSTN. > > In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't > chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 > instance and to get access to PSTN via this instance I had to register my > Asterisk instance as a gateway on my Sofia's external profile. This part of > my scenario works fine. I'm able to make calls using a softphone registered > on FreeSwitch to standard phones on PSTN with no problems. What I wasn't > able to do until now was the JsSIP + FreeSwitch integration. > > To setup FreeSwitch to comunicate with JsSIP, the only thing I did was > uncomment the line below on sip_profiles/internal.xml. > > > > I really don't know if just this is sufficient. Am I missing something > important? > > To connect on my FreeSwitch instance from Chrome, I'm using the Tryit > JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a > call to a PSTN phone. The connection is established but I don't get any > audio in both endpoints. The same happens when I try to call the 5000 ivr > extension or an user on a softphone at the same network from my Chrome > browser. > > Assuming that all the services I've mentioned here are running on the same > network, do you have any idea why I can't get audio in both endpoints of my > experiment? > > Additional information: > Ubuntu 12.04 64 bits > FreeSwitch version 1.5.5 default install configuration > Tryit JsSIP Demo with jssip-0.3.0.js > > Thanks in advance, > Rafael. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131003/d65cbd09/attachment.html From haakan.eriksson at ipadio.com Fri Oct 4 11:23:32 2013 From: haakan.eriksson at ipadio.com (Viking) Date: Fri, 4 Oct 2013 00:23:32 -0700 (PDT) Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> <7972438173691683594@unknownmsgid> <1380800507493-7595446.post@n2.nabble.com> Message-ID: <1380871412460-7595473.post@n2.nabble.com> Thanks! I kept comparing with our other installation and that line is not in that one. Saved my day / V -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Creating-SIP-UA-for-profile-X-tp7201299p7595473.html Sent from the freeswitch-users mailing list archive at Nabble.com. From khuenm at vega.com.vn Fri Oct 4 12:40:36 2013 From: khuenm at vega.com.vn (Khue Nguyen Minh) Date: Fri, 4 Oct 2013 15:40:36 +0700 Subject: [Freeswitch-users] get BYE event in javascripts Message-ID: Hi all, I have a simple javascripts with simple scenario. In this scenario, user can listen audio and send dtmf to play next or previous audio file. When user send dtmf to play other audio file, I will store play time of this audio to a text file (time.txt), but, if user hangup this call, the play time of last file cannot store in time.txt. How I can capture event user hangup to store play time into time.txt before javascripts session destroyed? Brs, Khue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/ca709049/attachment.html From regis.freeswitch.org at tornad.net Fri Oct 4 13:02:01 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Fri, 4 Oct 2013 11:02:01 +0200 Subject: [Freeswitch-users] get BYE event in javascripts In-Reply-To: References: Message-ID: Hi, You can use the hangup hook like : http://wiki.freeswitch.org/wiki/Variable_api_hangup_hook 2013/10/4 Khue Nguyen Minh > Hi all, > > I have a simple javascripts with simple scenario. In this scenario, user > can listen audio and send dtmf to play next or previous audio file. When > user send dtmf to play other audio file, I will store play time of this > audio to a text file (time.txt), but, if user hangup this call, the play > time of last file cannot store in time.txt. > > How I can capture event user hangup to store play time into time.txt > before javascripts session destroyed? > > Brs, > Khue. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/4e170946/attachment.html From noc at sonerep.com Fri Oct 4 12:56:52 2013 From: noc at sonerep.com (Groupe SOGO) Date: Fri, 04 Oct 2013 09:56:52 +0100 Subject: [Freeswitch-users] freeswitch openfire Message-ID: <524E82D4.5060902@sonerep.com> Dear Friends, I am working on freeswitch.Everything is going well. I want to integrate openfire 3.7.1 to freeswitch for video conference part. In my search, I find there is asterisk plugin for openfire. But I don't know if freeswitch plugin for openfire exists. Has anyone tried it? Where can I find this plugin? Thanks in advance for your help. Labolinux K. Amouzou From luis.azedo at factorlusitano.com Fri Oct 4 14:03:20 2013 From: luis.azedo at factorlusitano.com (Luis Azedo) Date: Fri, 4 Oct 2013 11:03:20 +0100 Subject: [Freeswitch-users] SRTP instructions on wiki wrong/outdated Message-ID: Hi, i have followed the instructions on wiki to setup srtp encryption and everything failed until i looked up in code and found that wiki refers to "sip_secure_media" but code uses "rtp_secure_media". cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/ea72ea40/attachment.html From bigx333 at gmail.com Fri Oct 4 14:14:07 2013 From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado) Date: Fri, 4 Oct 2013 12:14:07 +0200 Subject: [Freeswitch-users] SRTP instructions on wiki wrong/outdated In-Reply-To: References: Message-ID: What about updating it? On 4 Oct 2013 12:04, "Luis Azedo" wrote: > Hi, > i have followed the instructions on wiki to setup srtp encryption and > everything failed until i looked up in code and found that wiki refers to > "sip_secure_media" but code uses "rtp_secure_media". > > cheers > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/e01032d5/attachment.html From kris at kriskinc.com Fri Oct 4 18:16:14 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 4 Oct 2013 10:16:14 -0400 Subject: [Freeswitch-users] SRTP instructions on wiki wrong/outdated In-Reply-To: References: Message-ID: 1.2 and prior use sip_ master and later use rtp_ On Friday, October 4, 2013, Nelson Luiz Ferraz de Camargo Penteado wrote: > What about updating it? > On 4 Oct 2013 12:04, "Luis Azedo" > > wrote: > >> Hi, >> i have followed the instructions on wiki to setup srtp encryption and >> everything failed until i looked up in code and found that wiki refers to >> "sip_secure_media" but code uses "rtp_secure_media". >> >> cheers >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org > 'consulting at freeswitch.org');> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org > 'FreeSWITCH-users at lists.freeswitch.org');> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -- Sent from mobile device -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/40a619c1/attachment.html From brian at freeswitch.org Fri Oct 4 18:22:41 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Oct 2013 09:22:41 -0500 Subject: [Freeswitch-users] freeswitch openfire In-Reply-To: <524E82D4.5060902@sonerep.com> References: <524E82D4.5060902@sonerep.com> Message-ID: <580CC3C0-6A9B-49EC-B37C-80CCC2559112@freeswitch.org> Doesn't exist. Nobody has done any integration with FreeSWITCH and OpenFire as far as I'm aware of. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 4, 2013, at 3:56 AM, Groupe SOGO wrote: > Has anyone tried it? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/bb6a0286/attachment.bin From brian at freeswitch.org Fri Oct 4 18:35:11 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Oct 2013 09:35:11 -0500 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: <1380871412460-7595473.post@n2.nabble.com> References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> <7972438173691683594@unknownmsgid> <1380800507493-7595446.post@n2.nabble.com> <1380871412460-7595473.post@n2.nabble.com> Message-ID: That means your auto detection of local_ip_v4 is detecting the wrong IP, You should really just be setting the IP in the profile anyway. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 4, 2013, at 2:23 AM, Viking wrote: > Thanks! > I kept comparing with our other installation and that line is not in that > one. > Saved my day / V > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Creating-SIP-UA-for-profile-X-tp7201299p7595473.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/4f47f164/attachment.bin From vipkilla at gmail.com Fri Oct 4 18:43:20 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 4 Oct 2013 10:43:20 -0400 Subject: [Freeswitch-users] mod_callcenter del / add agent Message-ID: callcenter_config agent del agent1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/8a901f70/attachment.html From vipkilla at gmail.com Fri Oct 4 18:45:21 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 4 Oct 2013 10:45:21 -0400 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: References: Message-ID: stupid gmail sent this before i was finished writing.... Why does mod_callcenter not read the config XML when adding an agent back with the API? callcenter_config agent del agent1 callcenter_config agent add agent1 callback ^ when i run this command, the agent is added back but all the agent's parameters are defaulted... shouldn't it read the callcenter.conf.xml for the default values of the agent? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/de07297f/attachment-0001.html From ms4esl at gmail.com Fri Oct 4 12:09:43 2013 From: ms4esl at gmail.com (Marcin S) Date: Fri, 4 Oct 2013 10:09:43 +0200 Subject: [Freeswitch-users] esl dialer Message-ID: Hello, I'm trying to switch from dialogic api to freeswitch/esl. Dealing with inbound calls is easy (using esl outbound socket), however i have some problems with outbound calls. My application is written in C. I create a handle to esl (esl_connect), then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in order to handle this connection just like inbound call. So far so good. Here come the problems: 1. esl creates new thread for my call - thats ok, it was expected - but I do not receive any events on this new handle... 2. esl_events fails - here is some strace output: [pid 31588] send(5, "filter unique-id a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = 110 [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR command not found\n\n", 65535, 0) = 64 [pid 31588] send(5, "linger", 6, 0) = 6 [pid 31588] send(5, "\n\n", 2, 0) = 2 [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR command not found\n\n", 65535, 0) = 64 3. how can i distinguish, whether this call is successfull or not? What am I doing wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/d5a0c0d5/attachment.html From vbvbrj at gmail.com Fri Oct 4 20:40:51 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 04 Oct 2013 19:40:51 +0300 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: References: Message-ID: <524EEF93.4010600@gmail.com> On 04.10.2013 17:45, Vik Killa wrote: > callcenter_config agent add agent1 callback > ^ when i run this command, the agent is added back but all the agent's > parameters are defaulted... shouldn't it read the callcenter.conf.xml > for the default values of the agent? Think of agents like UUIDs. Names are just aliases. Every time an agent is added to the system via agent add, or from callcenter config file, it is a new UUID. That's why parameters are default. Add needed parameters again from CLI. -- Mimiko desu. From rgenthner at symplicity.com Fri Oct 4 21:17:39 2013 From: rgenthner at symplicity.com (Richard Genthner) Date: Fri, 4 Oct 2013 13:17:39 -0400 Subject: [Freeswitch-users] Mod_Callcenter callers Message-ID: <02F42845-33C2-4746-ABB8-6F7A16A145BE@symplicity.com> Is there a way in the dial plan for mod call center make it listen for a key press and break out and go to voicemail like if max-wait-time is hit ? -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/db81fefa/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 496 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/db81fefa/attachment.bin From max at nysolutions.com Fri Oct 4 21:25:45 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 4 Oct 2013 17:25:45 +0000 Subject: [Freeswitch-users] Mod_Callcenter callers In-Reply-To: <02F42845-33C2-4746-ABB8-6F7A16A145BE@symplicity.com> References: <02F42845-33C2-4746-ABB8-6F7A16A145BE@symplicity.com> Message-ID: I see mod_Fifo has that option documented http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo#Dialplan , would be interesting if call center has something similar and it just was not documented. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Genthner Sent: Friday, October 04, 2013 1:18 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Mod_Callcenter callers Is there a way in the dial plan for mod call center make it listen for a key press and break out and go to voicemail like if max-wait-time is hit ? -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/0d84d947/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/0d84d947/attachment.jpg From max at nysolutions.com Fri Oct 4 21:28:31 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 4 Oct 2013 17:28:31 +0000 Subject: [Freeswitch-users] Mod_Callcenter callers In-Reply-To: References: <02F42845-33C2-4746-ABB8-6F7A16A145BE@symplicity.com> Message-ID: It looks doable see http://freeswitch-users.2379917.n2.nabble.com/How-to-enable-dtmf-prompt-while-in-callcenter-queue-td7587914.html If I have some free time I will add it to the wiki. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein Sent: Friday, October 04, 2013 1:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Mod_Callcenter callers I see mod_Fifo has that option documented http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo#Dialplan , would be interesting if call center has something similar and it just was not documented. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Genthner Sent: Friday, October 04, 2013 1:18 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Mod_Callcenter callers Is there a way in the dial plan for mod call center make it listen for a key press and break out and go to voicemail like if max-wait-time is hit ? -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/3e9c9757/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/3e9c9757/attachment-0001.jpg From vipkilla at gmail.com Fri Oct 4 21:55:00 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 4 Oct 2013 13:55:00 -0400 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: <524EEF93.4010600@gmail.com> References: <524EEF93.4010600@gmail.com> Message-ID: this seems silly since if you do reload mod_callcenter, the XML is read from the agent data and the agent is added with those attributes.... why not have the add agent function same as if the module is being reloaded? On Fri, Oct 4, 2013 at 12:40 PM, Mimiko wrote: > On 04.10.2013 17:45, Vik Killa wrote: > > callcenter_config agent add agent1 callback > > ^ when i run this command, the agent is added back but all the agent's > > parameters are defaulted... shouldn't it read the callcenter.conf.xml > > for the default values of the agent? > > Think of agents like UUIDs. Names are just aliases. Every time an agent > is added to the system via agent add, or from callcenter config file, it > is a new UUID. That's why parameters are default. Add needed parameters > again from CLI. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/63582317/attachment.html From krice at freeswitch.org Fri Oct 4 22:45:35 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 04 Oct 2013 13:45:35 -0500 Subject: [Freeswitch-users] Friday Free For All! Message-ID: Hey Guys, Get in here! Friday Free For All is go!! SIP:888 at conference.freeswitch.org or use the https:/webrtc.freeswitch.org webrtc Demo to call 888! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/2e4a2baa/attachment.html From vbvbrj at gmail.com Fri Oct 4 23:02:28 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 04 Oct 2013 22:02:28 +0300 Subject: [Freeswitch-users] Mod_Callcenter callers In-Reply-To: <02F42845-33C2-4746-ABB8-6F7A16A145BE@symplicity.com> References: <02F42845-33C2-4746-ABB8-6F7A16A145BE@symplicity.com> Message-ID: <524F10C4.903@gmail.com> On 04.10.2013 20:17, Richard Genthner wrote: > Is there a way in the dial plan for mod call center make it listen for a > key press and break out and go to voicemail like if max-wait-time is hit ? -- Mimiko desu. From vbvbrj at gmail.com Fri Oct 4 23:06:49 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 04 Oct 2013 22:06:49 +0300 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: References: <524EEF93.4010600@gmail.com> Message-ID: <524F11C9.2050903@gmail.com> On 04.10.2013 20:55, Vik Killa wrote: > this seems silly since if you do reload mod_callcenter, the XML is read > from the agent data and the agent is added with those attributes.... why > not have the add agent function same as if the module is being reloaded? It's not silly. When you do a reload mod_callcenter, it's actually deletes all config an adds again agents in the same way as you do from CLI. In your case isn't better to just set agent's status to 'Loggd Off' and then 'Available'? Thus retaining agent's settings. -- Mimiko desu. From vipkilla at gmail.com Fri Oct 4 23:20:14 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 4 Oct 2013 15:20:14 -0400 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: <524F11C9.2050903@gmail.com> References: <524EEF93.4010600@gmail.com> <524F11C9.2050903@gmail.com> Message-ID: it's not just logged off to available, it's all of the agent's settings if you 'add' an agent, you need to 'set' each parameter maybe there should be a 'reload' agent API that does what im talking about. like the 'reload queue' api On Fri, Oct 4, 2013 at 3:06 PM, Mimiko wrote: > On 04.10.2013 20:55, Vik Killa wrote: > > this seems silly since if you do reload mod_callcenter, the XML is read > > from the agent data and the agent is added with those attributes.... why > > not have the add agent function same as if the module is being reloaded? > > It's not silly. When you do a reload mod_callcenter, it's actually > deletes all config an adds again agents in the same way as you do from > CLI. In your case isn't better to just set agent's status to 'Loggd Off' > and then 'Available'? Thus retaining agent's settings. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/5fe00d11/attachment.html From rafaelstnoliveira at gmail.com Fri Oct 4 23:19:38 2013 From: rafaelstnoliveira at gmail.com (Rafael Santana) Date: Fri, 4 Oct 2013 16:19:38 -0300 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status. @James My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server. []'s 2013/10/4 James Mortensen > Hi Rafael, > > You didn't mention whether the server was in the cloud. If you're server > is on Amazon EC2, make sure you're following the guide here: > https://wiki.freeswitch.org/wiki/Amazon_EC2 > > Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk > server, do you see audio flowing? Also, in Chrome, startup chrome from the > command line with the options to enable debug logging: > > chrome --enable-logging --v=11 > > Then look to see if there are STUN binding errors. Also, check > chrome://webrtc-internals, which will also tell you if Chrome is trying to > send audio. > > Is the server behind NAT or is it on the public Internet with it's own > public IP bound to the eth0 interface? > > Hope this helps! > > > > James > > > > On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana < > rafaelstnoliveira at gmail.com> wrote: > >> Hi, >> >> I'm new to telephony and FreeSwitch's world, so I apologize in advance >> for any nonsense I speak here. >> >> I've been trying to setup an environment where It can be possible to make >> a call through Google Chrome Browser using JsSIP to a standard phone device >> on PSTN. >> >> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't >> chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 >> instance and to get access to PSTN via this instance I had to register my >> Asterisk instance as a gateway on my Sofia's external profile. This part of >> my scenario works fine. I'm able to make calls using a softphone registered >> on FreeSwitch to standard phones on PSTN with no problems. What I wasn't >> able to do until now was the JsSIP + FreeSwitch integration. >> >> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was >> uncomment the line below on sip_profiles/internal.xml. >> >> >> >> I really don't know if just this is sufficient. Am I missing something >> important? >> >> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit >> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a >> call to a PSTN phone. The connection is established but I don't get any >> audio in both endpoints. The same happens when I try to call the 5000 ivr >> extension or an user on a softphone at the same network from my Chrome >> browser. >> >> Assuming that all the services I've mentioned here are running on the >> same network, do you have any idea why I can't get audio in both endpoints >> of my experiment? >> >> Additional information: >> Ubuntu 12.04 64 bits >> FreeSwitch version 1.5.5 default install configuration >> Tryit JsSIP Demo with jssip-0.3.0.js >> >> Thanks in advance, >> Rafael. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Rafael Santana Oliveira Mestre em Ci?ncia da Computa??o -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/df843244/attachment-0001.html From james.mortensen at synclio.com Sat Oct 5 00:21:18 2013 From: james.mortensen at synclio.com (James Mortensen) Date: Fri, 4 Oct 2013 13:21:18 -0700 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server. Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D) Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too. Hope this helps! James On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana wrote: > Thanks for the replies! I couldn't test or check anything today. As soon > as I do the tests I will inform here the new status. > > @James > My application server (nginx) is on the same network my FreeSwitch and > Asterisk are, so I'm not using a Stun server. > > []'s > > > 2013/10/4 James Mortensen > >> Hi Rafael, >> >> You didn't mention whether the server was in the cloud. If you're server >> is on Amazon EC2, make sure you're following the guide here: >> https://wiki.freeswitch.org/wiki/Amazon_EC2 >> >> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk >> server, do you see audio flowing? Also, in Chrome, startup chrome from the >> command line with the options to enable debug logging: >> >> chrome --enable-logging --v=11 >> >> Then look to see if there are STUN binding errors. Also, check >> chrome://webrtc-internals, which will also tell you if Chrome is trying to >> send audio. >> >> Is the server behind NAT or is it on the public Internet with it's own >> public IP bound to the eth0 interface? >> >> Hope this helps! >> >> >> >> James >> >> >> >> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana < >> rafaelstnoliveira at gmail.com> wrote: >> >>> Hi, >>> >>> I'm new to telephony and FreeSwitch's world, so I apologize in advance >>> for any nonsense I speak here. >>> >>> I've been trying to setup an environment where It can be possible to >>> make a call through Google Chrome Browser using JsSIP to a standard phone >>> device on PSTN. >>> >>> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't >>> chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 >>> instance and to get access to PSTN via this instance I had to register my >>> Asterisk instance as a gateway on my Sofia's external profile. This part of >>> my scenario works fine. I'm able to make calls using a softphone registered >>> on FreeSwitch to standard phones on PSTN with no problems. What I wasn't >>> able to do until now was the JsSIP + FreeSwitch integration. >>> >>> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was >>> uncomment the line below on sip_profiles/internal.xml. >>> >>> >>> >>> I really don't know if just this is sufficient. Am I missing something >>> important? >>> >>> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit >>> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a >>> call to a PSTN phone. The connection is established but I don't get any >>> audio in both endpoints. The same happens when I try to call the 5000 ivr >>> extension or an user on a softphone at the same network from my Chrome >>> browser. >>> >>> Assuming that all the services I've mentioned here are running on the >>> same network, do you have any idea why I can't get audio in both endpoints >>> of my experiment? >>> >>> Additional information: >>> Ubuntu 12.04 64 bits >>> FreeSwitch version 1.5.5 default install configuration >>> Tryit JsSIP Demo with jssip-0.3.0.js >>> >>> Thanks in advance, >>> Rafael. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Rafael Santana Oliveira > Mestre em Ci?ncia da Computa??o > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/31a64db9/attachment.html From grcamauer at gmail.com Sat Oct 5 02:34:12 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 4 Oct 2013 19:34:12 -0300 Subject: [Freeswitch-users] esl dialer In-Reply-To: References: Message-ID: I have a Dialer that uses ESL from C, but I make an inbound connection to Freeswitch. It is a socket which I keep open permanently. I recommend reading chapter 4 of the FreeSwitch Cookbook. Guillermo On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: > Hello, > > I'm trying to switch from dialogic api to freeswitch/esl. Dealing with > inbound calls is easy (using esl outbound socket), however i have some > problems with outbound calls. > My application is written in C. I create a handle to esl (esl_connect), > then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in > order to handle this connection just like inbound call. So far so good. > Here come the problems: > > 1. esl creates new thread for my call - thats ok, it was expected - but I > do not receive any events on this new handle... > > 2. esl_events fails - here is some strace output: > > [pid 31588] send(5, "filter unique-id > a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 > [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter > added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = > 110 > [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 > [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR command > not found\n\n", 65535, 0) = 64 > [pid 31588] send(5, "linger", 6, 0) = 6 > [pid 31588] send(5, "\n\n", 2, 0) = 2 > [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR command > not found\n\n", 65535, 0) = 64 > > 3. how can i distinguish, whether this call is successfull or not? > > What am I doing wrong? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/9d2677bc/attachment.html From rafaelstnoliveira at gmail.com Sat Oct 5 00:58:46 2013 From: rafaelstnoliveira at gmail.com (Rafael Santana) Date: Fri, 4 Oct 2013 17:58:46 -0300 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: James, Yes, Chrome is on a machine at the same network. This log file (http://pastebin.freeswitch.org/21480) is from a test scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset. I got this error during this test. [11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented reached in virtual void content::WebRtcAudioRenderer::Start() [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice value of thread to -10 I'm checking for clues right now. Any idea about it? Thanks for the attention! 2013/10/4 James Mortensen > But Chrome isn't on the same network, right? Also, I'm not an expert on > this, but from what I understand, STUN binding is something that occurs > between Chrome and the media server, not a STUN server. See Example 17 > here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. > The STUN binding occurs between the two user agents, where one is the SIP > user and the other could be your media server. > > Chrome will complain about STUN binding errors or receiving unknown > packets. If audio isn't flowing, all I'm trying to say is it might not be a > FreeSWITCH issue and you should make sure Chrome isn't the culprit before > changing too many things in FreeSWITCH. In one instance, my server's > network was the problem, and setting up the same exact FreeSWITCH (and even > Asterisk) configuration resulted in two way audio. :D) > > Not saying this is your problem, just that you should definitely be > watching what's happening in the Chrome debug logs too. > > Hope this helps! > > > James > > > > On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < > rafaelstnoliveira at gmail.com> wrote: > >> Thanks for the replies! I couldn't test or check anything today. As soon >> as I do the tests I will inform here the new status. >> >> @James >> My application server (nginx) is on the same network my FreeSwitch and >> Asterisk are, so I'm not using a Stun server. >> >> []'s >> >> >> 2013/10/4 James Mortensen >> >>> Hi Rafael, >>> >>> You didn't mention whether the server was in the cloud. If you're >>> server is on Amazon EC2, make sure you're following the guide here: >>> https://wiki.freeswitch.org/wiki/Amazon_EC2 >>> >>> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk >>> server, do you see audio flowing? Also, in Chrome, startup chrome from the >>> command line with the options to enable debug logging: >>> >>> chrome --enable-logging --v=11 >>> >>> Then look to see if there are STUN binding errors. Also, check >>> chrome://webrtc-internals, which will also tell you if Chrome is trying to >>> send audio. >>> >>> Is the server behind NAT or is it on the public Internet with it's own >>> public IP bound to the eth0 interface? >>> >>> Hope this helps! >>> >>> >>> >>> James >>> >>> >>> >>> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana < >>> rafaelstnoliveira at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> I'm new to telephony and FreeSwitch's world, so I apologize in advance >>>> for any nonsense I speak here. >>>> >>>> I've been trying to setup an environment where It can be possible to >>>> make a call through Google Chrome Browser using JsSIP to a standard phone >>>> device on PSTN. >>>> >>>> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I >>>> can't chance it today). To communicate with Chrome I have a FreeSwitch >>>> 1.5.5 instance and to get access to PSTN via this instance I had to >>>> register my Asterisk instance as a gateway on my Sofia's external profile. >>>> This part of my scenario works fine. I'm able to make calls using a >>>> softphone registered on FreeSwitch to standard phones on PSTN with no >>>> problems. What I wasn't able to do until now was the JsSIP + FreeSwitch >>>> integration. >>>> >>>> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was >>>> uncomment the line below on sip_profiles/internal.xml. >>>> >>>> >>>> >>>> I really don't know if just this is sufficient. Am I missing something >>>> important? >>>> >>>> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit >>>> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a >>>> call to a PSTN phone. The connection is established but I don't get any >>>> audio in both endpoints. The same happens when I try to call the 5000 ivr >>>> extension or an user on a softphone at the same network from my Chrome >>>> browser. >>>> >>>> Assuming that all the services I've mentioned here are running on the >>>> same network, do you have any idea why I can't get audio in both endpoints >>>> of my experiment? >>>> >>>> Additional information: >>>> Ubuntu 12.04 64 bits >>>> FreeSwitch version 1.5.5 default install configuration >>>> Tryit JsSIP Demo with jssip-0.3.0.js >>>> >>>> Thanks in advance, >>>> Rafael. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Rafael Santana Oliveira >> Mestre em Ci?ncia da Computa??o >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Rafael Santana Oliveira Mestre em Ci?ncia da Computa??o -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131004/91580062/attachment-0001.html From mike at jerris.com Sat Oct 5 19:54:42 2013 From: mike at jerris.com (Michael Jerris) Date: Sat, 5 Oct 2013 11:54:42 -0400 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: <2E965F77-B6C1-4407-BC02-2E2383BE9AAB@jerris.com> Sounds like some part of the audio renderer is not implemented on your version of chrome? On Oct 4, 2013, at 4:58 PM, Rafael Santana wrote: > James, > > Yes, Chrome is on a machine at the same network. > > This log file (http://pastebin.freeswitch.org/21480) is from a test scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset. I got this error during this test. > > [11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented reached in virtual void content::WebRtcAudioRenderer::Start() > [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice value of thread to -10 > > I'm checking for clues right now. Any idea about it? > > Thanks for the attention! > > > 2013/10/4 James Mortensen > But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server. See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server. > > Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. :D) > > Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too. > > Hope this helps! > > > James > > > > On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana wrote: > Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status. > > @James > My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server. > > []'s > > > 2013/10/4 James Mortensen > Hi Rafael, > > You didn't mention whether the server was in the cloud. If you're server is on Amazon EC2, make sure you're following the guide here: https://wiki.freeswitch.org/wiki/Amazon_EC2 > > Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk server, do you see audio flowing? Also, in Chrome, startup chrome from the command line with the options to enable debug logging: > > chrome --enable-logging --v=11 > > Then look to see if there are STUN binding errors. Also, check chrome://webrtc-internals, which will also tell you if Chrome is trying to send audio. > > Is the server behind NAT or is it on the public Internet with it's own public IP bound to the eth0 interface? > > Hope this helps! > > > > James > > > > On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana wrote: > Hi, > > I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. > > I've been trying to setup an environment where It can be possible to make a call through Google Chrome Browser using JsSIP to a standard phone device on PSTN. > > In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 instance and to get access to PSTN via this instance I had to register my Asterisk instance as a gateway on my Sofia's external profile. This part of my scenario works fine. I'm able to make calls using a softphone registered on FreeSwitch to standard phones on PSTN with no problems. What I wasn't able to do until now was the JsSIP + FreeSwitch integration. > > To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml. > > > > I really don't know if just this is sufficient. Am I missing something important? > > To connect on my FreeSwitch instance from Chrome, I'm using the Tryit JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a call to a PSTN phone. The connection is established but I don't get any audio in both endpoints. The same happens when I try to call the 5000 ivr extension or an user on a softphone at the same network from my Chrome browser. > > Assuming that all the services I've mentioned here are running on the same network, do you have any idea why I can't get audio in both endpoints of my experiment? > > Additional information: > Ubuntu 12.04 64 bits > FreeSwitch version 1.5.5 default install configuration > Tryit JsSIP Demo with jssip-0.3.0.js > > Thanks in advance, > Rafael. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Rafael Santana Oliveira > Mestre em Ci?ncia da Computa??o > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Rafael Santana Oliveira > Mestre em Ci?ncia da Computa??o > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/790f687c/attachment.html From jh.zhou at outlook.com Sun Oct 6 03:39:50 2013 From: jh.zhou at outlook.com (ZhouJianhua) Date: Sat, 5 Oct 2013 23:39:50 +0000 Subject: [Freeswitch-users] Does FS support sigcomp Message-ID: Seems the sofia-sip support sigcomp, and how to turn on it for freeswitch? One sigcomp lib is here:https://code.google.com/p/libsigcomp/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/65566154/attachment.html From oak.hidden at gmail.com Sun Oct 6 06:16:03 2013 From: oak.hidden at gmail.com (woot root) Date: Sat, 5 Oct 2013 21:16:03 -0500 Subject: [Freeswitch-users] how to get console warning via events? Message-ID: seems just set "event plain all" could not get the console logs. so how to get these console warning/error by esl events? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/fa5e84be/attachment-0001.html From max at nysolutions.com Sun Oct 6 06:21:12 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 6 Oct 2013 02:21:12 +0000 Subject: [Freeswitch-users] Registered(UDP-NAT) although no NAT Message-ID: I have a system where everything is working OK, however Freeswitch is showing the registrations as if they are behind Nat, although there is no NAT. sofia status profile Lan reg Call-ID: 5926c3a2-798304e0-f228d939 at 10.21.10.73 User: 5381 at 10.21.10.10 Contact: "user" Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.8.0070 Status: Registered(UDP-NAT)(unknown) EXP(2013-10-05 21:58:44) EXPSECS(106) Host: Fusion IP: 10.21.10.73 Port: 5060 Auth-User: 5381 Auth-Realm: 10.21.10.10 MWI-Account: 5381 at 10.21.10.10 show registrations 5381,10.21.10.10,5926c3a2-798304e0-f228d939 at 10.21.10.73,sofia/Lan/sip:5381 at 10.21.10.73;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060,1381025309,10.21.10.73,5060,udp,Fusion, Version FreeSWITCH Version 1.2.11+git~20130720T040554Z~39dfa5e422 (git 39dfa5e 2013-07-20 04:05:54Z) Win64 Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/a0070696/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/a0070696/attachment.jpg From anthony.minessale at gmail.com Sun Oct 6 08:19:03 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 5 Oct 2013 23:19:03 -0500 Subject: [Freeswitch-users] how to get console warning via events? In-Reply-To: References: Message-ID: You send the command "log 7" change 7 to desired log level. On Sat, Oct 5, 2013 at 9:16 PM, woot root wrote: > seems just set "event plain all" could not get the console logs. > > so how to get these console warning/error by esl events? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/9feb0f00/attachment.html From anthony.minessale at gmail.com Sun Oct 6 08:20:16 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 5 Oct 2013 23:20:16 -0500 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: <2E965F77-B6C1-4407-BC02-2E2383BE9AAB@jerris.com> References: <2E965F77-B6C1-4407-BC02-2E2383BE9AAB@jerris.com> Message-ID: Try setting the headset as the default audio device in your OS before you start chrome. Does it work on that same headset on webrtc.freeswitch.org? On Sat, Oct 5, 2013 at 10:54 AM, Michael Jerris wrote: > Sounds like some part of the audio renderer is not implemented on your > version of chrome? > > On Oct 4, 2013, at 4:58 PM, Rafael Santana > wrote: > > James, > > Yes, Chrome is on a machine at the same network. > > This log file (http://pastebin.freeswitch.org/21480) is from a test > scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On > the FS console I can see that the ivr call follow is executed perfectly, > but I can't hear anything on my headset. I got this error during this test. > > [11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented > reached in virtual void content::WebRtcAudioRenderer::Start() > [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice > value of thread to -10 > > I'm checking for clues right now. Any idea about it? > > Thanks for the attention! > > > 2013/10/4 James Mortensen > >> But Chrome isn't on the same network, right? Also, I'm not an expert on >> this, but from what I understand, STUN binding is something that occurs >> between Chrome and the media server, not a STUN server. See Example 17 >> here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. >> The STUN binding occurs between the two user agents, where one is the SIP >> user and the other could be your media server. >> >> Chrome will complain about STUN binding errors or receiving unknown >> packets. If audio isn't flowing, all I'm trying to say is it might not be a >> FreeSWITCH issue and you should make sure Chrome isn't the culprit before >> changing too many things in FreeSWITCH. In one instance, my server's >> network was the problem, and setting up the same exact FreeSWITCH (and even >> Asterisk) configuration resulted in two way audio. :D) >> >> Not saying this is your problem, just that you should definitely be >> watching what's happening in the Chrome debug logs too. >> >> Hope this helps! >> >> >> James >> >> >> >> On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < >> rafaelstnoliveira at gmail.com> wrote: >> >>> Thanks for the replies! I couldn't test or check anything today. As soon >>> as I do the tests I will inform here the new status. >>> >>> @James >>> My application server (nginx) is on the same network my FreeSwitch and >>> Asterisk are, so I'm not using a Stun server. >>> >>> []'s >>> >>> >>> 2013/10/4 James Mortensen >>> >>>> Hi Rafael, >>>> >>>> You didn't mention whether the server was in the cloud. If you're >>>> server is on Amazon EC2, make sure you're following the guide here: >>>> https://wiki.freeswitch.org/wiki/Amazon_EC2 >>>> >>>> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk >>>> server, do you see audio flowing? Also, in Chrome, startup chrome from the >>>> command line with the options to enable debug logging: >>>> >>>> chrome --enable-logging --v=11 >>>> >>>> Then look to see if there are STUN binding errors. Also, check >>>> chrome://webrtc-internals, which will also tell you if Chrome is >>>> trying to send audio. >>>> >>>> Is the server behind NAT or is it on the public Internet with it's own >>>> public IP bound to the eth0 interface? >>>> >>>> Hope this helps! >>>> >>>> >>>> >>>> James >>>> >>>> >>>> >>>> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana < >>>> rafaelstnoliveira at gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm new to telephony and FreeSwitch's world, so I apologize in advance >>>>> for any nonsense I speak here. >>>>> >>>>> I've been trying to setup an environment where It can be possible to >>>>> make a call through Google Chrome Browser using JsSIP to a standard phone >>>>> device on PSTN. >>>>> >>>>> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I >>>>> can't chance it today). To communicate with Chrome I have a FreeSwitch >>>>> 1.5.5 instance and to get access to PSTN via this instance I had to >>>>> register my Asterisk instance as a gateway on my Sofia's external profile. >>>>> This part of my scenario works fine. I'm able to make calls using a >>>>> softphone registered on FreeSwitch to standard phones on PSTN with no >>>>> problems. What I wasn't able to do until now was the JsSIP + FreeSwitch >>>>> integration. >>>>> >>>>> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was >>>>> uncomment the line below on sip_profiles/internal.xml. >>>>> >>>>> >>>>> >>>>> I really don't know if just this is sufficient. Am I missing something >>>>> important? >>>>> >>>>> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit >>>>> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a >>>>> call to a PSTN phone. The connection is established but I don't get any >>>>> audio in both endpoints. The same happens when I try to call the 5000 ivr >>>>> extension or an user on a softphone at the same network from my Chrome >>>>> browser. >>>>> >>>>> Assuming that all the services I've mentioned here are running on the >>>>> same network, do you have any idea why I can't get audio in both endpoints >>>>> of my experiment? >>>>> >>>>> Additional information: >>>>> Ubuntu 12.04 64 bits >>>>> FreeSwitch version 1.5.5 default install configuration >>>>> Tryit JsSIP Demo with jssip-0.3.0.js >>>>> >>>>> Thanks in advance, >>>>> Rafael. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Rafael Santana Oliveira >>> Mestre em Ci?ncia da Computa??o >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Rafael Santana Oliveira > Mestre em Ci?ncia da Computa??o > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/09268000/attachment-0001.html From anthony.minessale at gmail.com Sun Oct 6 08:22:36 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 5 Oct 2013 23:22:36 -0500 Subject: [Freeswitch-users] Registered(UDP-NAT) although no NAT In-Reply-To: References: Message-ID: Look at your sip register packet, there may be something in the via header that is triggering nat detection. On Sat, Oct 5, 2013 at 9:21 PM, Moishe Grunstein wrote: > I have a system where everything is working OK, however Freeswitch is > showing the registrations as if they are behind Nat, although there is no > NAT.**** > > ** ** > > *sofia status profile Lan reg* **** > > Call-ID: 5926c3a2-798304e0-f228d939 at 10.21.10.73 User: 5381 at 10.21.10.10Contact: "user" < > sip:5381 at 10.21.10.73;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060> > Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.8.0070 Status: > Registered(UDP-NAT)(unknown) EXP(2013-10-05 21:58:44) EXPSECS(106) Host: > Fusion IP: 10.21.10.73 Port: 5060 Auth-User: 5381 Auth-Realm: 10.21.10.10 > MWI-Account: 5381 at 10.21.10.10 **** > > ** ** > > *show registrations* **** > > 5381,10.21.10.10,5926c3a2-798304e0-f228d939 at 10.21.10.73,sofia/Lan/ > sip:5381 at 10.21.10.73;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060,1381025309,10.21.10.73,5060,udp,Fusion, > **** > > ** ** > > Version**** > > FreeSWITCH Version 1.2.11+git~20130720T040554Z~39dfa5e422 (git 39dfa5e > 2013-07-20 04:05:54Z)**** > > ** ** > > Win64**** > > ** ** > > ** ** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/f74da64b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/f74da64b/attachment.jpe From anthony.minessale at gmail.com Sun Oct 6 08:23:19 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 5 Oct 2013 23:23:19 -0500 Subject: [Freeswitch-users] Registered(UDP-NAT) although no NAT In-Reply-To: References: Message-ID: Or if the nat-acl is set, anything matching that acl will count as nat automatically. On Sat, Oct 5, 2013 at 11:22 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Look at your sip register packet, there may be something in the via header > that is triggering nat detection. > > > > On Sat, Oct 5, 2013 at 9:21 PM, Moishe Grunstein wrote: > >> I have a system where everything is working OK, however Freeswitch is >> showing the registrations as if they are behind Nat, although there is no >> NAT.**** >> >> ** ** >> >> *sofia status profile Lan reg* **** >> >> Call-ID: 5926c3a2-798304e0-f228d939 at 10.21.10.73 User: 5381 at 10.21.10.10Contact: "user" < >> sip:5381 at 10.21.10.73;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060> >> Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.8.0070 Status: >> Registered(UDP-NAT)(unknown) EXP(2013-10-05 21:58:44) EXPSECS(106) Host: >> Fusion IP: 10.21.10.73 Port: 5060 Auth-User: 5381 Auth-Realm: 10.21.10.10 >> MWI-Account: 5381 at 10.21.10.10 **** >> >> ** ** >> >> *show registrations* **** >> >> 5381,10.21.10.10,5926c3a2-798304e0-f228d939 at 10.21.10.73,sofia/Lan/ >> sip:5381 at 10.21.10.73;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060,1381025309,10.21.10.73,5060,udp,Fusion, >> **** >> >> ** ** >> >> Version**** >> >> FreeSWITCH Version 1.2.11+git~20130720T040554Z~39dfa5e422 (git 39dfa5e >> 2013-07-20 04:05:54Z)**** >> >> ** ** >> >> Win64**** >> >> ** ** >> >> ** ** >> >> ** ** >> >> ** ** >> >> Thanks,**** >> >> ** ** >> >> Moishe Grunstein**** >> >> Tornado Computer Systems, Inc.**** >> >> 212.400.7650 888.IPPBX.US >> *Service Request Email: support at nysolutions.com ***** >> >> Polycom Certified VAR >> Microsoft Small Business Specialist, Cisco SMB Select Certified**** >> >> [image: cid:image001.jpg at 01C72F94.9EE45D60] >> **** >> >> Computer Networking * Managed Services * IP Video Surveillance * Network >> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network >> Security * Site Surveys * CMS**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/a38e1904/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131005/a38e1904/attachment-0001.jpe From max at nysolutions.com Sun Oct 6 09:04:53 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 6 Oct 2013 05:04:53 +0000 Subject: [Freeswitch-users] Registered(UDP-NAT) although no NAT In-Reply-To: References: Message-ID: Thanks, commenting out fixed it. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, October 06, 2013 12:23 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Registered(UDP-NAT) although no NAT Or if the nat-acl is set, anything matching that acl will count as nat automatically. On Sat, Oct 5, 2013 at 11:22 PM, Anthony Minessale > wrote: Look at your sip register packet, there may be something in the via header that is triggering nat detection. On Sat, Oct 5, 2013 at 9:21 PM, Moishe Grunstein > wrote: I have a system where everything is working OK, however Freeswitch is showing the registrations as if they are behind Nat, although there is no NAT. sofia status profile Lan reg Call-ID: 5926c3a2-798304e0-f228d939 at 10.21.10.73 User: 5381 at 10.21.10.10 Contact: "user" ;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060> Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.8.0070 Status: Registered(UDP-NAT)(unknown) EXP(2013-10-05 21:58:44) EXPSECS(106) Host: Fusion IP: 10.21.10.73 Port: 5060 Auth-User: 5381 Auth-Realm: 10.21.10.10 MWI-Account: 5381 at 10.21.10.10 show registrations 5381,10.21.10.10,5926c3a2-798304e0-f228d939 at 10.21.10.73,sofia/Lan/sip:5381 at 10.21.10.73;fs_nat=yes;fs_path=sip%3A5381%4010.21.10.73%3A5060,1381025309,10.21.10.73,5060,udp,Fusion, Version FreeSWITCH Version 1.2.11+git~20130720T040554Z~39dfa5e422 (git 39dfa5e 2013-07-20 04:05:54Z) Win64 Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/9cfd7ecf/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/9cfd7ecf/attachment.jpg From guga.salazar.loor at gmail.com Sun Oct 6 14:50:35 2013 From: guga.salazar.loor at gmail.com (Gustavo Salazar) Date: Sun, 6 Oct 2013 05:50:35 -0500 Subject: [Freeswitch-users] hot hit message Message-ID: Hi, Recently in my freeswitch cli I see messages "hot hit 1" or "hot hit x" What is the meaning of those messages? -- Gustavo Salazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/b6f32eb4/attachment.html From steveayre at gmail.com Sun Oct 6 18:07:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 6 Oct 2013 15:07:06 +0100 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: STUN allows a client behind NAT to find the IP:port its packets are leaving externally on so that it knows the location to tell the server (FreeSWITCH) to send audio back to. In short STUN is used at whichever end is using NAT (which could be none, one or both). If FreeSWITCH is on a public IP but your PC running Chrome is on NAT (extremely likely) then Chrome still needs to use STUN. On 4 October 2013 21:21, James Mortensen wrote: > But Chrome isn't on the same network, right? Also, I'm not an expert on > this, but from what I understand, STUN binding is something that occurs > between Chrome and the media server, not a STUN server. See Example 17 > here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. > The STUN binding occurs between the two user agents, where one is the SIP > user and the other could be your media server. > > Chrome will complain about STUN binding errors or receiving unknown > packets. If audio isn't flowing, all I'm trying to say is it might not be a > FreeSWITCH issue and you should make sure Chrome isn't the culprit before > changing too many things in FreeSWITCH. In one instance, my server's > network was the problem, and setting up the same exact FreeSWITCH (and even > Asterisk) configuration resulted in two way audio. :D) > > Not saying this is your problem, just that you should definitely be > watching what's happening in the Chrome debug logs too. > > Hope this helps! > > > James > > > > On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < > rafaelstnoliveira at gmail.com> wrote: > >> Thanks for the replies! I couldn't test or check anything today. As soon >> as I do the tests I will inform here the new status. >> >> @James >> My application server (nginx) is on the same network my FreeSwitch and >> Asterisk are, so I'm not using a Stun server. >> >> []'s >> >> >> 2013/10/4 James Mortensen >> >>> Hi Rafael, >>> >>> You didn't mention whether the server was in the cloud. If you're >>> server is on Amazon EC2, make sure you're following the guide here: >>> https://wiki.freeswitch.org/wiki/Amazon_EC2 >>> >>> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk >>> server, do you see audio flowing? Also, in Chrome, startup chrome from the >>> command line with the options to enable debug logging: >>> >>> chrome --enable-logging --v=11 >>> >>> Then look to see if there are STUN binding errors. Also, check >>> chrome://webrtc-internals, which will also tell you if Chrome is trying to >>> send audio. >>> >>> Is the server behind NAT or is it on the public Internet with it's own >>> public IP bound to the eth0 interface? >>> >>> Hope this helps! >>> >>> >>> >>> James >>> >>> >>> >>> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana < >>> rafaelstnoliveira at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> I'm new to telephony and FreeSwitch's world, so I apologize in advance >>>> for any nonsense I speak here. >>>> >>>> I've been trying to setup an environment where It can be possible to >>>> make a call through Google Chrome Browser using JsSIP to a standard phone >>>> device on PSTN. >>>> >>>> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I >>>> can't chance it today). To communicate with Chrome I have a FreeSwitch >>>> 1.5.5 instance and to get access to PSTN via this instance I had to >>>> register my Asterisk instance as a gateway on my Sofia's external profile. >>>> This part of my scenario works fine. I'm able to make calls using a >>>> softphone registered on FreeSwitch to standard phones on PSTN with no >>>> problems. What I wasn't able to do until now was the JsSIP + FreeSwitch >>>> integration. >>>> >>>> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was >>>> uncomment the line below on sip_profiles/internal.xml. >>>> >>>> >>>> >>>> I really don't know if just this is sufficient. Am I missing something >>>> important? >>>> >>>> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit >>>> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a >>>> call to a PSTN phone. The connection is established but I don't get any >>>> audio in both endpoints. The same happens when I try to call the 5000 ivr >>>> extension or an user on a softphone at the same network from my Chrome >>>> browser. >>>> >>>> Assuming that all the services I've mentioned here are running on the >>>> same network, do you have any idea why I can't get audio in both endpoints >>>> of my experiment? >>>> >>>> Additional information: >>>> Ubuntu 12.04 64 bits >>>> FreeSwitch version 1.5.5 default install configuration >>>> Tryit JsSIP Demo with jssip-0.3.0.js >>>> >>>> Thanks in advance, >>>> Rafael. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Rafael Santana Oliveira >> Mestre em Ci?ncia da Computa??o >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/e479ba34/attachment-0001.html From oak.hidden at gmail.com Sun Oct 6 19:02:01 2013 From: oak.hidden at gmail.com (woot root) Date: Sun, 6 Oct 2013 10:02:01 -0500 Subject: [Freeswitch-users] how to get console warning via events? In-Reply-To: References: Message-ID: Anthony, thanks for the reply, i tried 'log 7' command, and i got +OK back. But i still don't see these Console logging in the events received, what i received are still "CHANNEL_ANSWER"/"CHANNEL_EXECUTE_COMPLETE" /"CUSTOM: conference::maintenance"/"BACKGROUND_JOB"/"CHANNEL_STATE" ......... I still don't get these logging showing on the Console screen :( ...... Please let me know how to receiveing these events after set log level thanks a ton!! On Sat, Oct 5, 2013 at 11:19 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You send the command "log 7" change 7 to desired log level. > > > > On Sat, Oct 5, 2013 at 9:16 PM, woot root wrote: > >> seems just set "event plain all" could not get the console logs. >> >> so how to get these console warning/error by esl events? >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/f9d2a5e2/attachment.html From togullu at gmail.com Sun Oct 6 23:23:56 2013 From: togullu at gmail.com (Jeff Ullman) Date: Sun, 6 Oct 2013 12:23:56 -0700 Subject: [Freeswitch-users] SIP return code changed from 480 to 408 Message-ID: Hi All: I realized during originate command SIP return code 480 is set to 408 by FS. Wireshark capture shows 480 from carrier and capture is attached. Pastebin for FS log : http://pastebin.freeswitch.org/21485 I'm using stable version of FS 1.2.3 Can someone please take a look ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/9d2c7fb8/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: wshark.png Type: image/png Size: 128149 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/9d2c7fb8/attachment-0001.png From brian at freeswitch.org Tue Oct 8 17:40:44 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Oct 2013 08:40:44 -0500 Subject: [Freeswitch-users] TEST Message-ID: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> TEST -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/446f704f/attachment.bin From lists at telefaks.de Mon Oct 7 17:03:12 2013 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 07 Oct 2013 15:03:12 +0200 Subject: [Freeswitch-users] JsSIP and wss In-Reply-To: <522A2C5E.3080105@telefaks.de> References: <5227686B.5030404@telefaks.de> <52279757.3050809@telefaks.de> <5228AFFA.6090306@telefaks.de> <5229A421.4060402@telefaks.de> <522A2C5E.3080105@telefaks.de> Message-ID: <5252B110.3020501@telefaks.de> I finally got it, For those who run into the same problem: I missed the --with-openssl option in ./configure --with-openssl Best regards Peter On 09/06/13 21:26, Peter Steinbach wrote: > I downloaded and installed > > * the latest JsSip 0.3.0 Development (uncompressed code, 564KB): > jssip-0.3.0.js and related "tryit" libs (2 days old now) > * the lastest Freeswitch GIT (2 days old now) > * Chromium is 28.0.1500 (Firefox shows the same behaviour) > > > Best regards > Peter > > > On 09/06/13 18:25, Anthony Minessale wrote: >> Out of curiosity, are you on latest master? >> >> >> On Fri, Sep 6, 2013 at 4:45 AM, Peter Steinbach > > wrote: >> >> On 09/06/13 00:52, Anthony Minessale wrote: >>> SSL stuff still shows handshake etc well in a pcap. >>> you need sofia loglevel all 9 to see any connection stuff. >> Thanks for the hint. Freeswitch log shows: >> tport.c:2773 tport_wakeup() tport_wakeup(0x7f59ccddbb70): events IN >> tport.c:2864 tport_recv_event() tport_recv_event(0x7f59ccddbb70) >> tport_type_ws.c:232 tport_recv_stream_ws() >> tport_recv_stream_ws(0x7f59ccddbb70): su_getmsgsize(): Host is >> down (112) >> tport.c:2159 tport_shutdown0() tport_shutdown0(0x7f59ccddbb70, 2) >> tport.c:2096 tport_close() tport_close(0x7f59ccddbb70): >> ws/192.168.178.102:40843/sip >> nua_registrar.c:200 registrar_tport_error() tport error 0: Success >> tport.c:4221 tport_release() tport_release(0x7f59ccddbb70): (nil) >> by 0x7f59cce033e0 with (nil) >> tport.c:2265 tport_set_secondary_timer() tport(0x7f59ccddbb70): >> set timer at 0 ms because zap >> nua_stack.c:271 nua_stack_event() nua(0x7f59cce033e0): event >> i_media_error 500 Transport error detected >> nua_stack.c:359 nua_application_event() nua: >> nua_application_event: entering >> tport.c:2265 tport_set_secondary_timer() tport(0x7f59ccddbb70): >> set timer at 0 ms because zap >> tport_type_ws.c:480 tport_ws_deinit_secondary() 0x7f59ccddbb70 >> destroy ws transport 0x7f59ccddbd60. >> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >> 2013-09-06 11:02:17.085874 [ALERT] sofia.c:1461 SOCKET >> DISCONNECT: rfg21joso6g9bf1oblmtqg 192.168.178.102:40843 >> >> >> I am wondering about 3 things: >> >> * Host is down (112) (but netstat shows below that 7443 is open) >> * ws/192.168.178.102:40843/sip >> (I was expecting wss here >> instead of ws) >> * 500 Transport error detected >> >> Do you see where this behaviour may come from? >> >> netstat -anlpe |grep 7443 >> tcp 0 0 192.168.178.220:7443 >> 0.0.0.0:* >> LISTEN 0 1460052941 25340/freeswitch >> >> Jssip shows: >> >> JsSIP | TRANSPORT | connecting to WebSocket >> wss://mydomain.com:7443 jssip-devel.js:556 >> JsSIP | TRANSPORT | WebSocket connection error: [object Event] >> jssip-devel.js:717 >> JsSIP | TRANSPORT | WebSocket disconnected (code: 1006) >> jssip-devel.js:611 >> JsSIP | TRANSPORT | WebSocket abrupt disconnection jssip-devel.js:614 >> JsSIP | UA | transport wss://mydomain.com:7443 failed | >> connection state set to 2 jssip-devel.js:5292 >> >> By the way, if I open http://192.168.178.220:7443 manually in the >> Chrome Browser, I read in Freeswitch's log: >> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x1feebb0): >> events IN >> tport.c:869 tport_alloc_secondary() >> tport_alloc_secondary(0x1feebb0): new secondary tport 0x7f59b842cdd0 >> tport.c:2622 tport_accept() tport_accept(0x1feebb0): incoming >> secondary on wss/192.168.178.220:7443/sips >> failed. reason = WS_INIT >> >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> mailto:lists (att) telefaks.de >> Internet: www.telefaks.de >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131007/fcbc1464/attachment.html From nneul at mst.edu Tue Oct 8 17:48:00 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 08 Oct 2013 08:48:00 -0500 Subject: [Freeswitch-users] Anyone know how to get a polycom to reload directory? Message-ID: <52540D10.7050906@mst.edu> I've got the config reload working fine by flushing registration, but it appears that doesn't result in it reading -directory.xml, it only reloads the base config files. Any suggestions beyond just teling the phone to do a full reboot? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From callum.guy at x-on.co.uk Tue Oct 8 17:49:44 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 8 Oct 2013 14:49:44 +0100 Subject: [Freeswitch-users] TEST In-Reply-To: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> References: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> Message-ID: Hello. ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 8 October 2013 14:40, Brian West wrote: > TEST > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/d8521a4c/attachment-0001.html From mike at jerris.com Tue Oct 8 17:25:55 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Oct 2013 09:25:55 -0400 Subject: [Freeswitch-users] test Message-ID: <76113B41-7308-43DB-B735-C1BE341C6242@jerris.com> test From ms4esl at gmail.com Tue Oct 8 17:06:10 2013 From: ms4esl at gmail.com (Marcin S) Date: Tue, 8 Oct 2013 15:06:10 +0200 Subject: [Freeswitch-users] esl socket inbound + playback In-Reply-To: References: Message-ID: Some more info, output from strace: create_uid/filter/myevents/event - all OK [pid 32639] recv(6, "Content-Type: auth/request\n\n", 65535, 0) = 28 [pid 32639] send(6, "auth ClueCon\n\n", 14, 0) = 14 [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK accepted\n\n", 65535, 0) = 54 [pid 32639] send(6, "api create_uuid", 15, 0) = 15 [pid 32639] send(6, "\n\n", 2, 0) = 2 [pid 32639] recv(6, "Content-Type: api/response\nContent-Length: 36\n\n", 65535, 0) = 47 [pid 32639] recv(6, "67b9c26c-3018-11e3-87c1-311d1641c437", 65535, 0) = 36 [pid 32639] send(6, "filter unique-id 67b9c26c-3018-11e3-87c1-311d1641c437\n\n", 55, 0) = 55 [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK filter added. [unique-id]=[67b9c26c-3018-11e3-87c1-311d1641c437]\n\n", 65535, 0) = 110 [pid 32639] send(6, "myevents 67b9c26c-3018-11e3-87c1-311d1641c437", 45, 0) = 45 [pid 32639] send(6, "\n\n", 2, 0) = 2 [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK Events Enabled\n\n", 65535, 0) = 60 [pid 32639] send(6, "event plain ALL\n\n", 17, 0) = 17 [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK event listener enabled plain\n\n", 65535, 0) = 74 linger fails: [pid 32639] send(6, "linger", 6, 0) = 6 [pid 32639] send(6, "\n\n", 2, 0) = 2 [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: -ERR not controlling a session\n\n", 65535, 0) = 72 originate - OK [pid 32635] send(6, "api originate {origination_uuid=67b9c26c-3018-11e3-87c1-311d1641c437}sofia/gateway/aster/473 &park", 98, 0) = 98 [pid 32635] send(6, "\n\n", 2, 0) = 2 [pid 32635] recv(6, "Content-Type: api/response\nContent-Length: 41\n\n", 65535, 0) = 47 [pid 32635] recv(6, "+OK 67b9c26c-3018-11e3-87c1-311d1641c437\n", 65535, 0) = 41 After CHANNEL_ANSWER there is a call to "playback" [pid 32635] send(6, "sendmsg\ncall-command: execute\nexecute-app-name: playback\nexecute-app-arg: /usrCC/CC/scenariusze/ScenTest-test/play/POLSKI/EWA//menu.wav\n\n", 137, 0) = 137 [pid 32635] recv(6, "Content-Type: command/reply\nReply-Text: -ERR invalid session id []\n\n", 65535, 0) = 68 [pid 32639] recv(6, "Content-Length: 1925\nContent-Type: text/event-plain\n\n", 65535, 0) = 53 [pid 32639] recv(6, "Event-Name: CHANNEL_CALLSTATE\nCore-UUID: 00fda624-2a7d-11e3-b4b2-311d1641c437\nFreeSWITCH-Hostname: fs-devel.altar\nFreeSWITCH-Switchname: fs-devel.altar\nFreeSWITCH-IPv4: 192.168.2.44\nFreeSWITCH-IPv6: %3A%3A1\nEvent-Date-Local: 2013-10-08%2014%3A52%3A53\nEvent-Date-GMT: Tue,%2008%20Oct%202013%2012%3A52%3A53%20GMT\nEvent-Date-Timestamp: 1381236773305888\nEvent-Calling-File: switch_channel.c\nEvent-Calling-Function: switch_channel_perform_set_callstate\nEvent-Calling-Line-Number: 242\nEvent-Sequence: 274033\nOriginal-Channel-Call-State: ACTIVE\nChannel-Call-State-Number: 6\nChannel-State: CS_EXECUTE\nChannel-Call-State: HANGUP\nChannel-State-Number: 10\nChannel-Name: sofia/external/473\nUnique-ID: 67b9c26c-3018-11e3-87c1-311d1641c437\nCall-Direction: outbound\nPresence-Call-Direction: outbound\nChannel-HIT-Dialplan: true\nChannel-Call-UUID: 67b9c26c-3018-11e3-87c1-311d1641c437\nAnswer-State: hangup\nChannel-Read-Codec-Name: PCMA\nChannel-Read-Codec-Rate: 8000\nChannel-Read-Codec-Bit-Rate: 64000\nChannel-Write-Codec-Name: PCMA\nChannel-Write-Codec-Rate: 8000\nChannel-Write-Codec-Bit-Rate: 64000\nCaller-Direction: outbound\nCaller-Caller-ID-Name: Outbound%20Call\nCaller-Caller-ID-Number: 473\nCaller-Network-Addr: 192.168.2.97\nCaller-Destination-Number: 473\nCaller-Unique-ID: 67b9c26c-3018-11e3-87c1-311d1641c437\nCaller-Source: src/switch_ivr_originate.c\nCaller-Context: default\nCaller-Channel-Name: sofia/external/473\nCaller-Profile-Index: 1\nCaller-Profile-Created-Time: 1381236712904967\nCaller-Channel-Created-Time: 1381236712904967\nCaller-Channel-Answered-Time: 1381236723045094\nCaller-Channel-Progress-Time: 1381236712904967\nCaller-Channel-Progress-Media-Time: 0\nCaller-Channel-Hangup-Time: 0\nCaller-Channel-Transfer-Time: 0\nCaller-Channel-Resurrect-Time: 0\nCaller-Channel-Bridged-Time: 0\nCaller-Channel-Last-Hold: 0\nCaller-Channel-Hold-Accum: 0\nCaller-Screen-Bit: true\nCaller-Privacy-Hide-Name: false\nCaller-Privacy-Hide-Number: false\n\n", 65535, 0) = 1925 How can I control the session in the same way as with socket outbound? 2013/10/8 Marcin S > Hello, > > I wrote simple C application, wich opens connection to esl - freeswitch > and makes call (originate ... &park). So far so good. I get > ESL_EVENT_CHANNEL_ORIGINATE, ESL_EVENT_CHANNEL_ANSWER and > ESL_EVENT_CHANNEL_PARK. Then I wan't to play wav file - but playback > command returns "-ERR invalid session id []". What is wrong? > > Specifying "originate ... &playback" is out of question. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/e1282a25/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Oct 8 17:56:11 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 8 Oct 2013 14:56:11 +0100 Subject: [Freeswitch-users] TEST In-Reply-To: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> References: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> Message-ID: ping On Tue, Oct 8, 2013 at 2:40 PM, Brian West wrote: > TEST > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/399144ae/attachment.html From mike at jerris.com Mon Oct 7 17:01:49 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 7 Oct 2013 09:01:49 -0400 Subject: [Freeswitch-users] how to get console warning via events? In-Reply-To: References: Message-ID: <31C71903-7931-4BCC-8408-DCD0A15501C3@jerris.com> Its possible you have your log levels turned down in configuration? On Oct 6, 2013, at 11:02 AM, woot root wrote: > Anthony, thanks for the reply, i tried 'log 7' command, and i got +OK back. > > But i still don't see these Console logging in the events received, what i received are still "CHANNEL_ANSWER"/"CHANNEL_EXECUTE_COMPLETE" /"CUSTOM: conference::maintenance"/"BACKGROUND_JOB"/"CHANNEL_STATE" ......... > > I still don't get these logging showing on the Console screen :( ...... > > Please let me know how to receiveing these events after set log level > > > thanks a ton!! > > > > > > On Sat, Oct 5, 2013 at 11:19 PM, Anthony Minessale wrote: > You send the command "log 7" change 7 to desired log level. > > > > On Sat, Oct 5, 2013 at 9:16 PM, woot root wrote: > seems just set "event plain all" could not get the console logs. > > so how to get these console warning/error by esl events? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131007/d791a6f8/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Tue Oct 8 18:07:00 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 8 Oct 2013 15:07:00 +0100 Subject: [Freeswitch-users] Optimizing profile In-Reply-To: References: Message-ID: Did you manage to track down the source of your problem? Any info you can post back will be useful to help others in the future that stumble across this thread! Cal On Tue, Oct 1, 2013 at 10:24 PM, Guillermo Ruiz Camauer wrote: > All, > > Thanks for the responses, specially for the tips on the tools. I have > installed both htop and sysstat (mpstat) and I am monitoring my system. I > will post back any relevant findings. > > Guillermo > > > On Mon, Sep 30, 2013 at 9:15 PM, Steven Ayre wrote: > >> Personally I prefer htop. You'll see CPU usage broken out for each core. >> >> top itself also has a similar option - run top and press '1' >> >> >> >> On 30 September 2013 20:26, Guillermo Ruiz Camauer wrote: >> >>> Yes, TOP is giving me the 12.5% reading. I understand that this is an >>> average reading of all cores. The fact that it never seems to go over >>> 12.5% tells me that I am never using more than 1 core at a time, and that >>> is is just the one process jumping around the different cores. Is this a >>> valid assumption? I want to test using more threads to see if I a hitting >>> a bottleneck with this.. >>> >>> Guillermo >>> >>> >>> On Mon, Sep 30, 2013 at 4:11 PM, Steven Ayre wrote: >>> >>>> If I only have one Voip provider, can I still have more than one SIP >>>>> profile? My provider authenticates via IP >>>> >>>> >>>> Profiles bind to a single ip:port combination. >>>> >>>> You could run multiple profiles each on a separate port. That way >>>> they'll all send to the provider using the same IP. >>>> >>>> >>>> >>>> >>>> >>>> On 30 September 2013 18:54, Guillermo Ruiz Camauer >>> > wrote: >>>> >>>>> In the Wiki, under "Performance testing and configurations", one of >>>>> the suggestions given under "Recommended SIP Settings" is: >>>>> >>>>> libsofia only handles 1 thread per profile, so if that is your bottle >>>>> neck use more profiles >>>>> >>>>> If I only have one Voip provider, can I still have more than one SIP >>>>> profile? My provider authenticates via IP. I currently run 240 concurrent >>>>> calls through this Sip trunk, I I see CPU close to 12.5% on a 8 core >>>>> machine. This means that one core is maxing out. How can I get more >>>>> threads up to distribute the load? >>>>> >>>>> >>>>> >>>>> -- >>>>> Guillermo Ruiz Camauer >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/519c9618/attachment.html From victor.chukalovskiy at gmail.com Tue Oct 8 07:14:45 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 07 Oct 2013 23:14:45 -0400 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions Message-ID: <525378A5.40706@gmail.com> Greetings! Everyone who runs some sort of billing for freeswitch, please provide your feedback here: http://www.surveymonkey.com/s/J8SL9TS Just 4 question. Less then a minute to complete! Survey runs for 2-3 days. I'll post final results in this thread for your benefit. Cheers, -Victor From jason.holden at start.ca Mon Oct 7 17:42:16 2013 From: jason.holden at start.ca (Jason Holden) Date: Mon, 7 Oct 2013 09:42:16 -0400 Subject: [Freeswitch-users] channels hung when placed on hold Message-ID: Currently running fs 1.2.13 and finding that when SLA extensions place calls on hold they are hung. Any one have any recommendations on resolving this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131007/0244fcb3/attachment.html From andrew at cassidywebservices.co.uk Tue Oct 8 18:25:13 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 8 Oct 2013 15:25:13 +0100 Subject: [Freeswitch-users] test In-Reply-To: <76113B41-7308-43DB-B735-C1BE341C6242@jerris.com> References: <76113B41-7308-43DB-B735-C1BE341C6242@jerris.com> Message-ID: It works! On 8 October 2013 14:25, Michael Jerris wrote: > test > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/a7adf820/attachment-0001.html From callum.guy at x-on.co.uk Tue Oct 8 18:29:07 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 8 Oct 2013 15:29:07 +0100 Subject: [Freeswitch-users] esl socket inbound + playback In-Reply-To: References: Message-ID: You are using inbound sockets so you need to tell the system the UUID of the call you wish to execute the application on. See: [pid 32635] recv(6, "Content-Type: command/reply\nReply-Text: -ERR invalid session id []\n\n", 65535, 0) = 68 You are missing a parameter. Message should be: SendMsg 2b06b28a-3018-11e3-959e-8b8d1e220731 call-command: execute execute-app-name: playback execute-app-arg: e.wav event-lock: true Does that help? ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 8 October 2013 14:06, Marcin S wrote: > Some more info, output from strace: > > create_uid/filter/myevents/event - all OK > > [pid 32639] recv(6, "Content-Type: auth/request\n\n", 65535, 0) = 28 > [pid 32639] send(6, "auth ClueCon\n\n", 14, 0) = 14 > [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK > accepted\n\n", 65535, 0) = 54 > [pid 32639] send(6, "api create_uuid", 15, 0) = 15 > [pid 32639] send(6, "\n\n", 2, 0) = 2 > [pid 32639] recv(6, "Content-Type: api/response\nContent-Length: 36\n\n", > 65535, 0) = 47 > [pid 32639] recv(6, "67b9c26c-3018-11e3-87c1-311d1641c437", 65535, 0) = 36 > [pid 32639] send(6, "filter unique-id > 67b9c26c-3018-11e3-87c1-311d1641c437\n\n", 55, 0) = 55 > [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK filter > added. [unique-id]=[67b9c26c-3018-11e3-87c1-311d1641c437]\n\n", 65535, 0) = > 110 > [pid 32639] send(6, "myevents 67b9c26c-3018-11e3-87c1-311d1641c437", 45, > 0) = 45 > [pid 32639] send(6, "\n\n", 2, 0) = 2 > [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK Events > Enabled\n\n", 65535, 0) = 60 > [pid 32639] send(6, "event plain ALL\n\n", 17, 0) = 17 > [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK event > listener enabled plain\n\n", 65535, 0) = 74 > > > linger fails: > [pid 32639] send(6, "linger", 6, 0) = 6 > [pid 32639] send(6, "\n\n", 2, 0) = 2 > [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: -ERR not > controlling a session\n\n", 65535, 0) = 72 > > > originate - OK > [pid 32635] send(6, "api originate > {origination_uuid=67b9c26c-3018-11e3-87c1-311d1641c437}sofia/gateway/aster/473 > &park", 98, 0) = 98 > [pid 32635] send(6, "\n\n", 2, 0) = 2 > [pid 32635] recv(6, "Content-Type: api/response\nContent-Length: 41\n\n", > 65535, 0) = 47 > [pid 32635] recv(6, "+OK 67b9c26c-3018-11e3-87c1-311d1641c437\n", 65535, > 0) = 41 > > > After CHANNEL_ANSWER there is a call to "playback" > > [pid 32635] send(6, "sendmsg\ncall-command: execute\nexecute-app-name: > playback\nexecute-app-arg: > /usrCC/CC/scenariusze/ScenTest-test/play/POLSKI/EWA//menu.wav\n\n", 137, 0) > = 137 > [pid 32635] recv(6, "Content-Type: command/reply\nReply-Text: -ERR invalid > session id []\n\n", 65535, 0) = 68 > [pid 32639] recv(6, "Content-Length: 1925\nContent-Type: > text/event-plain\n\n", 65535, 0) = 53 > [pid 32639] recv(6, "Event-Name: CHANNEL_CALLSTATE\nCore-UUID: > 00fda624-2a7d-11e3-b4b2-311d1641c437\nFreeSWITCH-Hostname: > fs-devel.altar\nFreeSWITCH-Switchname: fs-devel.altar\nFreeSWITCH-IPv4: > 192.168.2.44\nFreeSWITCH-IPv6: %3A%3A1\nEvent-Date-Local: > 2013-10-08%2014%3A52%3A53\nEvent-Date-GMT: > Tue,%2008%20Oct%202013%2012%3A52%3A53%20GMT\nEvent-Date-Timestamp: > 1381236773305888\nEvent-Calling-File: > switch_channel.c\nEvent-Calling-Function: > switch_channel_perform_set_callstate\nEvent-Calling-Line-Number: > 242\nEvent-Sequence: 274033\nOriginal-Channel-Call-State: > ACTIVE\nChannel-Call-State-Number: 6\nChannel-State: > CS_EXECUTE\nChannel-Call-State: HANGUP\nChannel-State-Number: > 10\nChannel-Name: sofia/external/473\nUnique-ID: > 67b9c26c-3018-11e3-87c1-311d1641c437\nCall-Direction: > outbound\nPresence-Call-Direction: outbound\nChannel-HIT-Dialplan: > true\nChannel-Call-UUID: > 67b9c26c-3018-11e3-87c1-311d1641c437\nAnswer-State: > hangup\nChannel-Read-Codec-Name: PCMA\nChannel-Read-Codec-Rate: > 8000\nChannel-Read-Codec-Bit-Rate: 64000\nChannel-Write-Codec-Name: > PCMA\nChannel-Write-Codec-Rate: 8000\nChannel-Write-Codec-Bit-Rate: > 64000\nCaller-Direction: outbound\nCaller-Caller-ID-Name: > Outbound%20Call\nCaller-Caller-ID-Number: 473\nCaller-Network-Addr: > 192.168.2.97\nCaller-Destination-Number: 473\nCaller-Unique-ID: > 67b9c26c-3018-11e3-87c1-311d1641c437\nCaller-Source: > src/switch_ivr_originate.c\nCaller-Context: default\nCaller-Channel-Name: > sofia/external/473\nCaller-Profile-Index: 1\nCaller-Profile-Created-Time: > 1381236712904967\nCaller-Channel-Created-Time: > 1381236712904967\nCaller-Channel-Answered-Time: > 1381236723045094\nCaller-Channel-Progress-Time: > 1381236712904967\nCaller-Channel-Progress-Media-Time: > 0\nCaller-Channel-Hangup-Time: 0\nCaller-Channel-Transfer-Time: > 0\nCaller-Channel-Resurrect-Time: 0\nCaller-Channel-Bridged-Time: > 0\nCaller-Channel-Last-Hold: 0\nCaller-Channel-Hold-Accum: > 0\nCaller-Screen-Bit: true\nCaller-Privacy-Hide-Name: > false\nCaller-Privacy-Hide-Number: false\n\n", 65535, 0) = 1925 > > How can I control the session in the same way as with socket outbound? > > > > > > > > > > > > > > 2013/10/8 Marcin S > >> Hello, >> >> I wrote simple C application, wich opens connection to esl - freeswitch >> and makes call (originate ... &park). So far so good. I get >> ESL_EVENT_CHANNEL_ORIGINATE, ESL_EVENT_CHANNEL_ANSWER and >> ESL_EVENT_CHANNEL_PARK. Then I wan't to play wav file - but playback >> command returns "-ERR invalid session id []". What is wrong? >> >> Specifying "originate ... &playback" is out of question. >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/e0a0edcd/attachment.html From callum.guy at x-on.co.uk Tue Oct 8 18:29:47 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 8 Oct 2013 15:29:47 +0100 Subject: [Freeswitch-users] test In-Reply-To: <76113B41-7308-43DB-B735-C1BE341C6242@jerris.com> References: <76113B41-7308-43DB-B735-C1BE341C6242@jerris.com> Message-ID: pong ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 8 October 2013 14:25, Michael Jerris wrote: > test > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/5ac0c49a/attachment-0001.html From miha at softnet.si Tue Oct 8 10:31:18 2013 From: miha at softnet.si (Miha) Date: Tue, 08 Oct 2013 08:31:18 +0200 Subject: [Freeswitch-users] Diversion header In-Reply-To: <5252A838.3050408@softnet.si> References: <5252A838.3050408@softnet.si> Message-ID: <5253A6B6.7030307@softnet.si> Hi, I am sending invite throught FS with diversion header. Is it possible to set that FS will not remove diversion header and it will send it further? tnx! br, miha From nneul at mst.edu Tue Oct 8 00:22:54 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 07 Oct 2013 15:22:54 -0500 Subject: [Freeswitch-users] Anyone know how to get a polycom to reload directory? Message-ID: <5253181E.60303@mst.edu> I've got the config reload working fine by flushing registration, but it appears that doesn't result in it reading -directory.xml, it only reloads the base config files. Any suggestions beyond just teling the phone to do a full reboot? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nickolayr at gmail.com Mon Oct 7 23:50:51 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Mon, 7 Oct 2013 15:50:51 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure In-Reply-To: <68A9174972294D3DB4061464B242508A@owner397fa27d2> References: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> <68A9174972294D3DB4061464B242508A@owner397fa27d2> Message-ID: I am receiving a transferred call. Is it possible to obtain in dialplan variables all numbers - sip_refer_to and sip_h_referred-by? Thank you. -- Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131007/da563a4f/attachment.html From mario_fs at mgtech.com Tue Oct 8 18:49:43 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 8 Oct 2013 07:49:43 -0700 Subject: [Freeswitch-users] test In-Reply-To: References: <76113B41-7308-43DB-B735-C1BE341C6242@jerris.com> Message-ID: <6D5223FF-47F5-4721-BECF-D84B74AB5173@mgtech.com> I wondered why I didn't get the ML for a couple of days?. On Oct 8, 2013, at 7:29 AM, Callum Guy wrote: > pong > > > ______________________________ > > Callum Guy > Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > > Please consider the environment before printing this email. > > > On 8 October 2013 14:25, Michael Jerris wrote: > test > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/2b4ccdeb/attachment.html From GB at cm.nl Tue Oct 8 18:51:39 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 8 Oct 2013 16:51:39 +0200 Subject: [Freeswitch-users] Multiple contexts per sip profile Message-ID: Hello, I'm trying to configure FreeSWITCH to have multiple contexts per sip profile like below. But this does not seem to work. The reason I want this, is to have each incoming called number to have its own context in which extensions can be created. The problem with using extensions and only one context is I can't defined more than one extension with the same name. FS seems to overwrite or ignore ones that do. Is this even possible? If so, what do I need to change? Is there perhaps a better way for doing this? sip_profile/inbound.xml: dialplan/inbound.xml: dialplan/inbound/1000.xml: dialplan/inbound/1001.xml: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/bc5f461d/attachment-0001.html From lloyd.aloysius at gmail.com Tue Oct 8 19:00:05 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 8 Oct 2013 11:00:05 -0400 Subject: [Freeswitch-users] Multiple contexts per sip profile Message-ID: Please have a look into here http://wiki.freeswitch.org/wiki/Multi-tenant Lloyd * * On Tue, Oct 8, 2013 at 10:51 AM, Grant Bagdasarian wrote: > Hello,**** > > ** ** > > I?m trying to configure FreeSWITCH to have multiple contexts per sip > profile like below.**** > > ** ** > > But this does not seem to work.**** > > The reason I want this, is to have each incoming called number to have its > own context in which extensions can be created. **** > > The problem with using extensions and only one context is I can?t defined > more than one extension with the same name. FS seems to overwrite or ignore > ones that do.**** > > ** ** > > Is this even possible? If so, what do I need to change? Is there perhaps a > better way for doing this?**** > > ** ** > > *sip_profile/inbound.xml:* > > ** ** > > **** > > ** ** > > **** > > ** ** > > **** > > **** > > ** ** > > **** > > **** > > ** ** > > *dialplan/inbound.xml:* > > ** ** > > **** > > **** > > **** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > *dialplan/inbound/1000.xml:* > > ** ** > > **** > > ** ** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > ** ** > > **** > > **** > > data="/usr/src/freeswitch/sounds/fs-hello-world-for-1000.wav"/>**** > > **** > > **** > > ** ** > > **** > > **** > > ** ** > > *dialplan/inbound/1001.xml:* > > ** ** > > **** > > ** ** > > **** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > ** ** > > **** > > **** > > data="/usr/src/freeswitch/sounds/fs-hello-world-for-1001.wav"/>**** > > **** > > **** > > ** ** > > **** > > **** > > ** ** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/8b716a16/attachment.html From rgenthner at symplicity.com Tue Oct 8 19:01:02 2013 From: rgenthner at symplicity.com (Richard Genthner) Date: Tue, 8 Oct 2013 11:01:02 -0400 Subject: [Freeswitch-users] TEST In-Reply-To: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> References: <5340F878-542A-4C02-91AB-60932BF9D8FC@freeswitch.org> Message-ID: <18714655-7FE2-45EB-8D88-028A83996B53@symplicity.com> SYN -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com On Oct 8, 2013, at 9:40 AM, Brian West wrote: > TEST > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/55ddf0ea/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 496 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/55ddf0ea/attachment.bin From nickolayr at gmail.com Tue Oct 8 19:05:22 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Tue, 8 Oct 2013 11:05:22 -0400 Subject: [Freeswitch-users] REFER and channel variables Message-ID: Hello, I am receiving a transferred call in fs. Is it possible in to get the number (in dialplan) who did the transfer? Thank you. -- Rogoshchenkov Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/9b6332b3/attachment.html From jleung at v10networks.ca Tue Oct 8 19:05:01 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Tue, 08 Oct 2013 23:05:01 +0800 Subject: [Freeswitch-users] TEST Message-ID: <001601cec437$c63769d1$7c07000a@smb.curriegrad2004.ca> SYN ACK Richard Genthner wrote: SYN -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com On Oct 8, 2013, at 9:40 AM, Brian West wrote: > TEST > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From acrow at integrafin.co.uk Tue Oct 8 19:07:30 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 08 Oct 2013 16:07:30 +0100 Subject: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions In-Reply-To: References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch> <9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch> Message-ID: <52541FB2.9090602@integrafin.co.uk> Hi, I can confirm that the issue affects me in version: FreeSWITCH Version 1.2.11-n20130817T172515Z-1~wheezy+1+git~20130816T225403Z~8566ffa82a (-n20130817T172515Z-1~wheezy+1git 8566ffa 2013-08-16 22:54:03Z) but not in: FreeSWITCH Version 1.2.8+git~20130405T052920Z~57fb368b32 (git 57fb368 2013-04-05 05:29:20Z) if that helps anyone. These are production systems though so I've had to disable the relevant codecs. Cheers Alex ----Original Message---- *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions *Date:* Sun, 4 Aug 2013 02:43:27 +0000 *From:* Moishe Grunstein *To:* FreeSWITCH Users Help > I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related. > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah > Sent: Saturday, August 03, 2013 10:34 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions > > I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. > Thanks and I'll make sure I get it through the process. > > Cheers > On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson wrote: > >> On Sat, Aug 3, 2013 at 3:31 PM, Emrah wrote: >>> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy. >>> >>> The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end? >> This is what you need to know: >> https://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> >> Best, >> Gabe >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc. 29 Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 5300. (Registered office: as above; Registered in England and Wales under number: 3727592). Authorised and regulated by the Financial Conduct Authority (entered on the Financial Services Register; no. 190856). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/eed92354/attachment-0001.html From oak.hidden at gmail.com Mon Oct 7 07:05:26 2013 From: oak.hidden at gmail.com (woot root) Date: Sun, 6 Oct 2013 22:05:26 -0500 Subject: [Freeswitch-users] how to get console warning via events? In-Reply-To: References: Message-ID: I figured out, thanks! On Sun, Oct 6, 2013 at 10:02 AM, woot root wrote: > Anthony, thanks for the reply, i tried 'log 7' command, and i got +OK back. > > But i still don't see these Console logging in the events received, what i > received are still "CHANNEL_ANSWER"/"CHANNEL_EXECUTE_COMPLETE" /"CUSTOM: > conference::maintenance"/"BACKGROUND_JOB"/"CHANNEL_STATE" ......... > > I still don't get these logging showing on the Console screen :( ...... > > Please let me know how to receiveing these events after set log level > > > thanks a ton!! > > > > > > On Sat, Oct 5, 2013 at 11:19 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> You send the command "log 7" change 7 to desired log level. >> >> >> >> On Sat, Oct 5, 2013 at 9:16 PM, woot root wrote: >> >>> seems just set "event plain all" could not get the console logs. >>> >>> so how to get these console warning/error by esl events? >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131006/2cb6d29c/attachment.html From brian at freeswitch.org Tue Oct 8 19:39:12 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Oct 2013 10:39:12 -0500 Subject: [Freeswitch-users] Diversion header In-Reply-To: <5253A6B6.7030307@softnet.si> References: <5252A838.3050408@softnet.si> <5253A6B6.7030307@softnet.si> Message-ID: <52EA0C6C-84A2-4812-ADAE-8F70D1358A14@freeswitch.org> You have to remember this one very key fact? we aren't a proxy, if you want the diversion header on the b-leg you MUST export it/set it on the b-leg. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 8, 2013, at 1:31 AM, Miha wrote: > > Hi, > > I am sending invite throught FS with diversion header. Is it possible to > set that FS will not remove diversion header and it will send it further? > > tnx! > > br, > miha > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/ba4c20b6/attachment.bin From brian at freeswitch.org Tue Oct 8 19:40:16 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Oct 2013 10:40:16 -0500 Subject: [Freeswitch-users] channels hung when placed on hold In-Reply-To: References: Message-ID: <4E695AF9-B147-4205-A7F4-FDA589E06467@freeswitch.org> Sir, Sounds like you need to do a clean build on the latest stable and master code to see if the problem is gone. JIRA Maybe? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 7, 2013, at 8:42 AM, Jason Holden wrote: > Currently running fs 1.2.13 and finding that when SLA extensions place calls on hold they are hung. > Any one have any recommendations on resolving this? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/a908d8ad/attachment.bin From brian at freeswitch.org Tue Oct 8 19:42:07 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Oct 2013 10:42:07 -0500 Subject: [Freeswitch-users] REFER and channel variables In-Reply-To: References: Message-ID: Load the XML cdr and inspect the variables of a call, referred_by and possibly sip_referred_by depending on stable vs head. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 8, 2013, at 10:05 AM, Nikolay Rogoshchenkov wrote: > Hello, > > I am receiving a transferred call in fs. Is it possible in to get the number (in dialplan) who did the transfer? > Thank you. > > -- > Rogoshchenkov Nikolay > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/49cdd87a/attachment.bin From anthony.minessale at gmail.com Tue Oct 8 19:45:05 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Oct 2013 10:45:05 -0500 Subject: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions In-Reply-To: <52541FB2.9090602@integrafin.co.uk> References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch> <9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch> <52541FB2.9090602@integrafin.co.uk> Message-ID: So 4 months and nobody takes the hint to file a JIRA. Bugs that are reported here have little chance of being dealt with. I can say on latest HEAD of stable branch the eyebeam I tested with did not exhibit this problem. Anything more on this thread needs to be moved to JIRA.... On Tue, Oct 8, 2013 at 10:07 AM, Alex Crow wrote: > Hi, > > I can confirm that the issue affects me in version: > > FreeSWITCH Version > 1.2.11-n20130817T172515Z-1~wheezy+1+git~20130816T225403Z~8566ffa82a > (-n20130817T172515Z-1~wheezy+1git 8566ffa 2013-08-16 22:54:03Z) > > but not in: > > FreeSWITCH Version 1.2.8+git~20130405T052920Z~57fb368b32 (git 57fb368 > 2013-04-05 05:29:20Z) > > if that helps anyone. > > These are production systems though so I've had to disable the relevant > codecs. > > Cheers > > Alex > > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in > all recent FreeSWITCH versions > *Date:* Sun, 4 Aug 2013 02:43:27 +0000 > *From:* Moishe Grunstein > *To:* FreeSWITCH Users Help > > > > I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related. > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc.212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Emrah > Sent: Saturday, August 03, 2013 10:34 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions > > I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. > Thanks and I'll make sure I get it through the process. > > Cheers > On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson wrote: > > > On Sat, Aug 3, 2013 at 3:31 PM, Emrah wrote: > > I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy. > > The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end? > > This is what you need to know:https://wiki.freeswitch.org/wiki/Reporting_Bugs > > > Best, > Gabe > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rshttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > "Transact" is operated by Integrated Financial Arrangements plc. 29 > Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 > 5300. (Registered office: as above; Registered in England and Wales > under number: 3727592). Authorised and regulated by the Financial > Conduct Authority (entered on the Financial Services Register; no. 190856). > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/1bb7baae/attachment-0001.html From peter at hartmanncomputer.com Tue Oct 8 00:55:31 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Mon, 7 Oct 2013 16:55:31 -0400 Subject: [Freeswitch-users] old calls hanging around Message-ID: Hi, First, thanks for Freeswitch! I'm experiencing an issue where 'show calls' returns several calls that aren't actually happening both inbound and outbound. Has anyone seen this before? Rebooting the handset (Polycom IP 550) associated with that extension has no effect so it seems in FS. freeswitch at internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,unknown ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 9 total. Running: FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 2013-10-02 21:30:46Z) Thanks much! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com From nickolayr at gmail.com Tue Oct 8 20:04:46 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Tue, 8 Oct 2013 12:04:46 -0400 Subject: [Freeswitch-users] REFER and channel variables In-Reply-To: References: Message-ID: Thank you Brian. I just enable this module. Is it possible to use this variables in dialplan for external app? -- Rogoshchenkov Nikolay On Tue, Oct 8, 2013 at 11:42 AM, Brian West wrote: > Load the XML cdr and inspect the variables of a call, referred_by and > possibly sip_referred_by depending on stable vs head. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > On Oct 8, 2013, at 10:05 AM, Nikolay Rogoshchenkov > wrote: > > > Hello, > > > > I am receiving a transferred call in fs. Is it possible in to get the > number (in dialplan) who did the transfer? > > Thank you. > > > > -- > > Rogoshchenkov Nikolay > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/dfb7ce59/attachment.html From vipkilla at gmail.com Mon Oct 7 19:38:21 2013 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Oct 2013 11:38:21 -0400 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: References: <524EEF93.4010600@gmail.com> <524F11C9.2050903@gmail.com> Message-ID: or even better do this with tiers as well.... http://jira.freeswitch.org/secure/attachment/19368/callcenter_config.tier.and.agent.reload.c.diff.rev1.txt On Mon, Oct 7, 2013 at 10:57 AM, Vik Killa wrote: > turns out to be a simple patch.... > > http://jira.freeswitch.org/secure/attachment/19367/callcenter_config.agent.reload.c.diff.rev1.txt > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131007/c9eeae25/attachment.html From ms4esl at gmail.com Tue Oct 8 16:04:25 2013 From: ms4esl at gmail.com (Marcin S) Date: Tue, 8 Oct 2013 14:04:25 +0200 Subject: [Freeswitch-users] esl socket inbound + playback Message-ID: Hello, I wrote simple C application, wich opens connection to esl - freeswitch and makes call (originate ... &park). So far so good. I get ESL_EVENT_CHANNEL_ORIGINATE, ESL_EVENT_CHANNEL_ANSWER and ESL_EVENT_CHANNEL_PARK. Then I wan't to play wav file - but playback command returns "-ERR invalid session id []". What is wrong? Specifying "originate ... &playback" is out of question. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/cddbc9a4/attachment.html From victor.chukalovskiy at gmail.com Tue Oct 8 20:16:45 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 08 Oct 2013 12:16:45 -0400 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: <525378A5.40706@gmail.com> References: <525378A5.40706@gmail.com> Message-ID: <52542FED.1050506@gmail.com> Bump! Not sure if my message reached the list yesterday (given all the "tests" today) :-) http://www.surveymonkey.com/s/J8SL9TS On 13-10-07 11:14 PM, Victor Chukalovskiy wrote: > Greetings! > > Everyone who runs some sort of billing for freeswitch, please provide > your feedback here: > > http://www.surveymonkey.com/s/J8SL9TS > > Just 4 question. Less then a minute to complete! > > Survey runs for 2-3 days. I'll post final results in this thread for > your benefit. > > Cheers, > -Victor From acrow at integrafin.co.uk Tue Oct 8 20:29:40 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 08 Oct 2013 17:29:40 +0100 Subject: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions In-Reply-To: References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch> <9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch> <52541FB2.9090602@integrafin.co.uk> Message-ID: <525432F4.70401@integrafin.co.uk> Sorry, I was just informing the other sufferers (and I am happy to disable it on my prod systems anyway). I did a search on Jira to no avail but wasn't sure if someone might have posted it already in a way that my search missed. I can file a Jira but I doubt I have time to build a test lab with the requisite phones to follow it up, at least in the next month or two - but if you think it might be useful to have it up there I will file it. Cheers Alex On 08/10/13 16:45, Anthony Minessale wrote: > So 4 months and nobody takes the hint to file a JIRA. > Bugs that are reported here have little chance of being dealt with. > > I can say on latest HEAD of stable branch the eyebeam I tested with > did not exhibit this problem. > Anything more on this thread needs to be moved to JIRA.... > > > > On Tue, Oct 8, 2013 at 10:07 AM, Alex Crow > wrote: > > Hi, > > I can confirm that the issue affects me in version: > > FreeSWITCH Version > 1.2.11-n20130817T172515Z-1~wheezy+1+git~20130816T225403Z~8566ffa82a (-n20130817T172515Z-1~wheezy+1git > 8566ffa 2013-08-16 22:54:03Z) > > but not in: > > FreeSWITCH Version 1.2.8+git~20130405T052920Z~57fb368b32 (git > 57fb368 2013-04-05 05:29:20Z) > > if that helps anyone. > > These are production systems though so I've had to disable the > relevant codecs. > > Cheers > > Alex > > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in > all recent FreeSWITCH versions > *Date:* Sun, 4 Aug 2013 02:43:27 +0000 > *From:* Moishe Grunstein > > *To:* FreeSWITCH Users Help > > > > > >> I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related. >> >> >> Thanks, >> >> Moishe Grunstein >> Tornado Computer Systems, Inc. >> 212.400.7650 888.IPPBX.US >> Service Request Email:support at nysolutions.com >> Polycom Certified VAR >> Microsoft Small Business Specialist, Cisco SMB Select Certified >> >> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS >> >> -----Original Message----- >> From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah >> Sent: Saturday, August 03, 2013 10:34 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions >> >> I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. >> Thanks and I'll make sure I get it through the process. >> >> Cheers >> On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson wrote: >> >>> On Sat, Aug 3, 2013 at 3:31 PM, Emrah wrote: >>>> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy. >>>> >>>> The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end? >>> This is what you need to know: >>> https://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >>> >>> Best, >>> Gabe >>> >>> ______________________________________________________________________ >>> ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> rs >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > "Transact" is operated by Integrated Financial Arrangements plc. 29 > Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 > 5300. (Registered office: as above; Registered in England and Wales > under number: 3727592). Authorised and regulated by the Financial > Conduct Authority (entered on the Financial Services Register; no. 190856). > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/a30ba9ef/attachment-0001.html From vipkilla at gmail.com Mon Oct 7 18:57:15 2013 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 Oct 2013 10:57:15 -0400 Subject: [Freeswitch-users] mod_callcenter del / add agent In-Reply-To: References: <524EEF93.4010600@gmail.com> <524F11C9.2050903@gmail.com> Message-ID: turns out to be a simple patch.... http://jira.freeswitch.org/secure/attachment/19367/callcenter_config.agent.reload.c.diff.rev1.txt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131007/6ba08250/attachment.html From robert.hadley at teotech.com Tue Oct 8 20:35:43 2013 From: robert.hadley at teotech.com (Robert Hadley) Date: Tue, 8 Oct 2013 16:35:43 +0000 Subject: [Freeswitch-users] Diversion header In-Reply-To: <5253A6B6.7030307@softnet.si> References: <5252A838.3050408@softnet.si> <5253A6B6.7030307@softnet.si> Message-ID: <54e7e2ac6aa84998af778802cfd5de1e@CO1PR04MB524.namprd04.prod.outlook.com> Hi Miha, FS works for me. If there is a Diversion header in the invite coming in, sofia includes the header in the invite that is forwarded out to a SIP trunk. Regards, Robert -----Original Message----- From: Miha [mailto:miha at softnet.si] Sent: Monday, October 07, 2013 11:31 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Diversion header Hi, I am sending invite throught FS with diversion header. Is it possible to set that FS will not remove diversion header and it will send it further? tnx! br, miha From ms4esl at gmail.com Tue Oct 8 11:20:37 2013 From: ms4esl at gmail.com (Marcin S) Date: Tue, 8 Oct 2013 09:20:37 +0200 Subject: [Freeswitch-users] esl dialer In-Reply-To: References: Message-ID: Ok, I will try this approach. Am I right, that in order to handle oubound calls similiar to inbound, I will have to spawn new thread for each call? 2013/10/5 Guillermo Ruiz Camauer > I have a Dialer that uses ESL from C, but I make an inbound connection to > Freeswitch. It is a socket which I keep open permanently. > I recommend reading chapter 4 of the FreeSwitch Cookbook. > > Guillermo > > > On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: > >> Hello, >> >> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >> inbound calls is easy (using esl outbound socket), however i have some >> problems with outbound calls. >> My application is written in C. I create a handle to esl (esl_connect), >> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >> order to handle this connection just like inbound call. So far so good. >> Here come the problems: >> >> 1. esl creates new thread for my call - thats ok, it was expected - but I >> do not receive any events on this new handle... >> >> 2. esl_events fails - here is some strace output: >> >> [pid 31588] send(5, "filter unique-id >> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter >> added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = >> 110 >> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >> command not found\n\n", 65535, 0) = 64 >> [pid 31588] send(5, "linger", 6, 0) = 6 >> [pid 31588] send(5, "\n\n", 2, 0) = 2 >> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >> command not found\n\n", 65535, 0) = 64 >> >> 3. how can i distinguish, whether this call is successfull or not? >> >> What am I doing wrong? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/1c1634ac/attachment.html From grcamauer at gmail.com Tue Oct 8 21:26:08 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 8 Oct 2013 14:26:08 -0300 Subject: [Freeswitch-users] esl dialer In-Reply-To: References: Message-ID: I handle all ESL communications from just one thread which receives and processes or generates all events and commands. I do have a timer for each call which I use for certain timeouts (user inputs and ring time). Guillermo On Tue, Oct 8, 2013 at 4:20 AM, Marcin S wrote: > Ok, I will try this approach. Am I right, that in order to handle oubound > calls similiar to inbound, I will have to spawn new thread for each call? > > > 2013/10/5 Guillermo Ruiz Camauer > >> I have a Dialer that uses ESL from C, but I make an inbound connection to >> Freeswitch. It is a socket which I keep open permanently. >> I recommend reading chapter 4 of the FreeSwitch Cookbook. >> >> Guillermo >> >> >> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: >> >>> Hello, >>> >>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>> inbound calls is easy (using esl outbound socket), however i have some >>> problems with outbound calls. >>> My application is written in C. I create a handle to esl (esl_connect), >>> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >>> order to handle this connection just like inbound call. So far so good. >>> Here come the problems: >>> >>> 1. esl creates new thread for my call - thats ok, it was expected - but >>> I do not receive any events on this new handle... >>> >>> 2. esl_events fails - here is some strace output: >>> >>> [pid 31588] send(5, "filter unique-id >>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter >>> added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = >>> 110 >>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> [pid 31588] send(5, "linger", 6, 0) = 6 >>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> >>> 3. how can i distinguish, whether this call is successfull or not? >>> >>> What am I doing wrong? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/f08b23c8/attachment-0001.html From ms4esl at gmail.com Tue Oct 8 22:28:41 2013 From: ms4esl at gmail.com (Marcin S) Date: Tue, 8 Oct 2013 20:28:41 +0200 Subject: [Freeswitch-users] esl socket inbound + playback In-Reply-To: References: Message-ID: Yes, this is exactly this problem. For "socket outbound" was not neccessary, but for "socket inbound" it is. Thank you! 2013/10/8 Callum Guy > You are using inbound sockets so you need to tell the system the UUID of > the call you wish to execute the application on. > > See: > > [pid 32635] recv(6, "Content-Type: command/reply\nReply-Text: -ERR invalid > session id []\n\n", 65535, 0) = 68 > > You are missing a parameter. > > Message should be: > SendMsg 2b06b28a-3018-11e3-959e-8b8d1e220731 > call-command: execute > execute-app-name: playback > execute-app-arg: e.wav > event-lock: true > > Does that help? > > > ______________________________ > > Callum Guy > Developer > > X-on > Framlingham Technology Centre > Station Road, Framlingham, > Suffolk, IP13 9EZ > > T 0333 332 0116 > E callum.guy at x-on.co.uk > > > X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales > Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD > Company Registration No. 2578478 > > This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message > is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from > your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of > the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have > been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on > are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. > Please consider the environment before printing this email. > > > > On 8 October 2013 14:06, Marcin S wrote: > >> Some more info, output from strace: >> >> create_uid/filter/myevents/event - all OK >> >> [pid 32639] recv(6, "Content-Type: auth/request\n\n", 65535, 0) = 28 >> [pid 32639] send(6, "auth ClueCon\n\n", 14, 0) = 14 >> [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK >> accepted\n\n", 65535, 0) = 54 >> [pid 32639] send(6, "api create_uuid", 15, 0) = 15 >> [pid 32639] send(6, "\n\n", 2, 0) = 2 >> [pid 32639] recv(6, "Content-Type: api/response\nContent-Length: 36\n\n", >> 65535, 0) = 47 >> [pid 32639] recv(6, "67b9c26c-3018-11e3-87c1-311d1641c437", 65535, 0) = 36 >> [pid 32639] send(6, "filter unique-id >> 67b9c26c-3018-11e3-87c1-311d1641c437\n\n", 55, 0) = 55 >> [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK filter >> added. [unique-id]=[67b9c26c-3018-11e3-87c1-311d1641c437]\n\n", 65535, 0) = >> 110 >> [pid 32639] send(6, "myevents 67b9c26c-3018-11e3-87c1-311d1641c437", 45, >> 0) = 45 >> [pid 32639] send(6, "\n\n", 2, 0) = 2 >> [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK Events >> Enabled\n\n", 65535, 0) = 60 >> [pid 32639] send(6, "event plain ALL\n\n", 17, 0) = 17 >> [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: +OK event >> listener enabled plain\n\n", 65535, 0) = 74 >> >> >> linger fails: >> [pid 32639] send(6, "linger", 6, 0) = 6 >> [pid 32639] send(6, "\n\n", 2, 0) = 2 >> [pid 32639] recv(6, "Content-Type: command/reply\nReply-Text: -ERR not >> controlling a session\n\n", 65535, 0) = 72 >> >> >> originate - OK >> [pid 32635] send(6, "api originate >> {origination_uuid=67b9c26c-3018-11e3-87c1-311d1641c437}sofia/gateway/aster/473 >> &park", 98, 0) = 98 >> [pid 32635] send(6, "\n\n", 2, 0) = 2 >> [pid 32635] recv(6, "Content-Type: api/response\nContent-Length: 41\n\n", >> 65535, 0) = 47 >> [pid 32635] recv(6, "+OK 67b9c26c-3018-11e3-87c1-311d1641c437\n", 65535, >> 0) = 41 >> >> >> After CHANNEL_ANSWER there is a call to "playback" >> >> [pid 32635] send(6, "sendmsg\ncall-command: execute\nexecute-app-name: >> playback\nexecute-app-arg: >> /usrCC/CC/scenariusze/ScenTest-test/play/POLSKI/EWA//menu.wav\n\n", 137, 0) >> = 137 >> [pid 32635] recv(6, "Content-Type: command/reply\nReply-Text: -ERR >> invalid session id []\n\n", 65535, 0) = 68 >> [pid 32639] recv(6, "Content-Length: 1925\nContent-Type: >> text/event-plain\n\n", 65535, 0) = 53 >> [pid 32639] recv(6, "Event-Name: CHANNEL_CALLSTATE\nCore-UUID: >> 00fda624-2a7d-11e3-b4b2-311d1641c437\nFreeSWITCH-Hostname: >> fs-devel.altar\nFreeSWITCH-Switchname: fs-devel.altar\nFreeSWITCH-IPv4: >> 192.168.2.44\nFreeSWITCH-IPv6: %3A%3A1\nEvent-Date-Local: >> 2013-10-08%2014%3A52%3A53\nEvent-Date-GMT: >> Tue,%2008%20Oct%202013%2012%3A52%3A53%20GMT\nEvent-Date-Timestamp: >> 1381236773305888\nEvent-Calling-File: >> switch_channel.c\nEvent-Calling-Function: >> switch_channel_perform_set_callstate\nEvent-Calling-Line-Number: >> 242\nEvent-Sequence: 274033\nOriginal-Channel-Call-State: >> ACTIVE\nChannel-Call-State-Number: 6\nChannel-State: >> CS_EXECUTE\nChannel-Call-State: HANGUP\nChannel-State-Number: >> 10\nChannel-Name: sofia/external/473\nUnique-ID: >> 67b9c26c-3018-11e3-87c1-311d1641c437\nCall-Direction: >> outbound\nPresence-Call-Direction: outbound\nChannel-HIT-Dialplan: >> true\nChannel-Call-UUID: >> 67b9c26c-3018-11e3-87c1-311d1641c437\nAnswer-State: >> hangup\nChannel-Read-Codec-Name: PCMA\nChannel-Read-Codec-Rate: >> 8000\nChannel-Read-Codec-Bit-Rate: 64000\nChannel-Write-Codec-Name: >> PCMA\nChannel-Write-Codec-Rate: 8000\nChannel-Write-Codec-Bit-Rate: >> 64000\nCaller-Direction: outbound\nCaller-Caller-ID-Name: >> Outbound%20Call\nCaller-Caller-ID-Number: 473\nCaller-Network-Addr: >> 192.168.2.97\nCaller-Destination-Number: 473\nCaller-Unique-ID: >> 67b9c26c-3018-11e3-87c1-311d1641c437\nCaller-Source: >> src/switch_ivr_originate.c\nCaller-Context: default\nCaller-Channel-Name: >> sofia/external/473\nCaller-Profile-Index: 1\nCaller-Profile-Created-Time: >> 1381236712904967\nCaller-Channel-Created-Time: >> 1381236712904967\nCaller-Channel-Answered-Time: >> 1381236723045094\nCaller-Channel-Progress-Time: >> 1381236712904967\nCaller-Channel-Progress-Media-Time: >> 0\nCaller-Channel-Hangup-Time: 0\nCaller-Channel-Transfer-Time: >> 0\nCaller-Channel-Resurrect-Time: 0\nCaller-Channel-Bridged-Time: >> 0\nCaller-Channel-Last-Hold: 0\nCaller-Channel-Hold-Accum: >> 0\nCaller-Screen-Bit: true\nCaller-Privacy-Hide-Name: >> false\nCaller-Privacy-Hide-Number: false\n\n", 65535, 0) = 1925 >> >> How can I control the session in the same way as with socket outbound? >> >> >> >> >> >> >> >> >> >> >> >> >> >> 2013/10/8 Marcin S >> >>> Hello, >>> >>> I wrote simple C application, wich opens connection to esl - freeswitch >>> and makes call (originate ... &park). So far so good. I get >>> ESL_EVENT_CHANNEL_ORIGINATE, ESL_EVENT_CHANNEL_ANSWER and >>> ESL_EVENT_CHANNEL_PARK. Then I wan't to play wav file - but playback >>> command returns "-ERR invalid session id []". What is wrong? >>> >>> Specifying "originate ... &playback" is out of question. >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/020121af/attachment.html From ms4esl at gmail.com Tue Oct 8 22:30:11 2013 From: ms4esl at gmail.com (Marcin S) Date: Tue, 8 Oct 2013 20:30:11 +0200 Subject: [Freeswitch-users] esl dialer In-Reply-To: References: Message-ID: I was considering this approach, however I don't know how to detect and answer a call in "socket inboud" esl mode. 2013/10/8 Guillermo Ruiz Camauer > I handle all ESL communications from just one thread which receives and > processes or generates all events and commands. I do have a timer for each > call which I use for certain timeouts (user inputs and ring time). > > Guillermo > > > On Tue, Oct 8, 2013 at 4:20 AM, Marcin S wrote: > >> Ok, I will try this approach. Am I right, that in order to handle oubound >> calls similiar to inbound, I will have to spawn new thread for each call? >> >> >> 2013/10/5 Guillermo Ruiz Camauer >> >>> I have a Dialer that uses ESL from C, but I make an inbound connection >>> to Freeswitch. It is a socket which I keep open permanently. >>> I recommend reading chapter 4 of the FreeSwitch Cookbook. >>> >>> Guillermo >>> >>> >>> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: >>> >>>> Hello, >>>> >>>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>>> inbound calls is easy (using esl outbound socket), however i have some >>>> problems with outbound calls. >>>> My application is written in C. I create a handle to esl (esl_connect), >>>> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >>>> order to handle this connection just like inbound call. So far so good. >>>> Here come the problems: >>>> >>>> 1. esl creates new thread for my call - thats ok, it was expected - but >>>> I do not receive any events on this new handle... >>>> >>>> 2. esl_events fails - here is some strace output: >>>> >>>> [pid 31588] send(5, "filter unique-id >>>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK >>>> filter added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", >>>> 65535, 0) = 110 >>>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>>> command not found\n\n", 65535, 0) = 64 >>>> [pid 31588] send(5, "linger", 6, 0) = 6 >>>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>>> command not found\n\n", 65535, 0) = 64 >>>> >>>> 3. how can i distinguish, whether this call is successfull or not? >>>> >>>> What am I doing wrong? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/5359094f/attachment-0001.html From rafaelstnoliveira at gmail.com Tue Oct 8 22:26:02 2013 From: rafaelstnoliveira at gmail.com (Rafael Santana) Date: Tue, 8 Oct 2013 15:26:02 -0300 Subject: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio In-Reply-To: References: Message-ID: Hi guys! Thanks for your help! I got it running right now. For future reference, I did the following steps to accomplish a two way audio while calling from Chrome to a PSTN phone. I've migrated my environment to a CentOS 5.9. I've installed FreeSwitch following the default recipe, but instead of ./configure -C I did ./configure --with-openssl # suggested by @Iwan To make available websocket connection on FS, I've just uncommented the following on sip_profiles/internal.xml > And that is it! :D To reach PSTN I used a pre-configured Asterisk instance as gateway. It is important to note that my client (Chrome), FreeSwitch and Asterisk were all at the same network while doing this tests. PS: I turned off my iptables to eliminate any possibility of port blocking while doing my tests. Thanks again! 2013/10/6 Steven Ayre > STUN allows a client behind NAT to find the IP:port its packets are > leaving externally on so that it knows the location to tell the server > (FreeSWITCH) to send audio back to. > > In short STUN is used at whichever end is using NAT (which could be none, > one or both). > > If FreeSWITCH is on a public IP but your PC running Chrome is on NAT > (extremely likely) then Chrome still needs to use STUN. > > > On 4 October 2013 21:21, James Mortensen wrote: > >> But Chrome isn't on the same network, right? Also, I'm not an expert on >> this, but from what I understand, STUN binding is something that occurs >> between Chrome and the media server, not a STUN server. See Example 17 >> here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17. >> The STUN binding occurs between the two user agents, where one is the SIP >> user and the other could be your media server. >> >> Chrome will complain about STUN binding errors or receiving unknown >> packets. If audio isn't flowing, all I'm trying to say is it might not be a >> FreeSWITCH issue and you should make sure Chrome isn't the culprit before >> changing too many things in FreeSWITCH. In one instance, my server's >> network was the problem, and setting up the same exact FreeSWITCH (and even >> Asterisk) configuration resulted in two way audio. :D) >> >> Not saying this is your problem, just that you should definitely be >> watching what's happening in the Chrome debug logs too. >> >> Hope this helps! >> >> >> James >> >> >> >> On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana < >> rafaelstnoliveira at gmail.com> wrote: >> >>> Thanks for the replies! I couldn't test or check anything today. As soon >>> as I do the tests I will inform here the new status. >>> >>> @James >>> My application server (nginx) is on the same network my FreeSwitch and >>> Asterisk are, so I'm not using a Stun server. >>> >>> []'s >>> >>> >>> 2013/10/4 James Mortensen >>> >>>> Hi Rafael, >>>> >>>> You didn't mention whether the server was in the cloud. If you're >>>> server is on Amazon EC2, make sure you're following the guide here: >>>> https://wiki.freeswitch.org/wiki/Amazon_EC2 >>>> >>>> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk >>>> server, do you see audio flowing? Also, in Chrome, startup chrome from the >>>> command line with the options to enable debug logging: >>>> >>>> chrome --enable-logging --v=11 >>>> >>>> Then look to see if there are STUN binding errors. Also, check >>>> chrome://webrtc-internals, which will also tell you if Chrome is trying to >>>> send audio. >>>> >>>> Is the server behind NAT or is it on the public Internet with it's own >>>> public IP bound to the eth0 interface? >>>> >>>> Hope this helps! >>>> >>>> >>>> >>>> James >>>> >>>> >>>> >>>> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana < >>>> rafaelstnoliveira at gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm new to telephony and FreeSwitch's world, so I apologize in advance >>>>> for any nonsense I speak here. >>>>> >>>>> I've been trying to setup an environment where It can be possible to >>>>> make a call through Google Chrome Browser using JsSIP to a standard phone >>>>> device on PSTN. >>>>> >>>>> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I >>>>> can't chance it today). To communicate with Chrome I have a FreeSwitch >>>>> 1.5.5 instance and to get access to PSTN via this instance I had to >>>>> register my Asterisk instance as a gateway on my Sofia's external profile. >>>>> This part of my scenario works fine. I'm able to make calls using a >>>>> softphone registered on FreeSwitch to standard phones on PSTN with no >>>>> problems. What I wasn't able to do until now was the JsSIP + FreeSwitch >>>>> integration. >>>>> >>>>> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was >>>>> uncomment the line below on sip_profiles/internal.xml. >>>>> >>>>> >>>>> >>>>> I really don't know if just this is sufficient. Am I missing something >>>>> important? >>>>> >>>>> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit >>>>> JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a >>>>> call to a PSTN phone. The connection is established but I don't get any >>>>> audio in both endpoints. The same happens when I try to call the 5000 ivr >>>>> extension or an user on a softphone at the same network from my Chrome >>>>> browser. >>>>> >>>>> Assuming that all the services I've mentioned here are running on the >>>>> same network, do you have any idea why I can't get audio in both endpoints >>>>> of my experiment? >>>>> >>>>> Additional information: >>>>> Ubuntu 12.04 64 bits >>>>> FreeSwitch version 1.5.5 default install configuration >>>>> Tryit JsSIP Demo with jssip-0.3.0.js >>>>> >>>>> Thanks in advance, >>>>> Rafael. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Rafael Santana Oliveira >>> Mestre em Ci?ncia da Computa??o >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Rafael Santana Oliveira Mestre em Ci?ncia da Computa??o -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/cc4b6d4e/attachment.html From nickolayr at gmail.com Wed Oct 9 01:27:04 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Tue, 8 Oct 2013 17:27:04 -0400 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: <52542FED.1050506@gmail.com> References: <525378A5.40706@gmail.com> <52542FED.1050506@gmail.com> Message-ID: In second question is impossible to chose "other". -- Rogoshchenkov Nikolay On Tue, Oct 8, 2013 at 12:16 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Bump! Not sure if my message reached the list yesterday (given all the > "tests" today) :-) > > http://www.surveymonkey.com/s/J8SL9TS > > On 13-10-07 11:14 PM, Victor Chukalovskiy wrote: > > Greetings! > > > > Everyone who runs some sort of billing for freeswitch, please provide > > your feedback here: > > > > http://www.surveymonkey.com/s/J8SL9TS > > > > Just 4 question. Less then a minute to complete! > > > > Survey runs for 2-3 days. I'll post final results in this thread for > > your benefit. > > > > Cheers, > > -Victor > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/ef925e7f/attachment-0001.html From grcamauer at gmail.com Wed Oct 9 03:14:03 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 8 Oct 2013 20:14:03 -0300 Subject: [Freeswitch-users] esl dialer In-Reply-To: References: Message-ID: Are you writing a Dialer or an IVR? A Dialer doesn't really need to answer calls. You can always have one program (or thread) with an inbound socket and another with an outbound socket. Guillermo On Tue, Oct 8, 2013 at 3:30 PM, Marcin S wrote: > I was considering this approach, however I don't know how to detect and > answer a call in "socket inboud" esl mode. > > > 2013/10/8 Guillermo Ruiz Camauer > >> I handle all ESL communications from just one thread which receives and >> processes or generates all events and commands. I do have a timer for each >> call which I use for certain timeouts (user inputs and ring time). >> >> Guillermo >> >> >> On Tue, Oct 8, 2013 at 4:20 AM, Marcin S wrote: >> >>> Ok, I will try this approach. Am I right, that in order to handle >>> oubound calls similiar to inbound, I will have to spawn new thread for each >>> call? >>> >>> >>> 2013/10/5 Guillermo Ruiz Camauer >>> >>>> I have a Dialer that uses ESL from C, but I make an inbound connection >>>> to Freeswitch. It is a socket which I keep open permanently. >>>> I recommend reading chapter 4 of the FreeSwitch Cookbook. >>>> >>>> Guillermo >>>> >>>> >>>> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: >>>> >>>>> Hello, >>>>> >>>>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>>>> inbound calls is easy (using esl outbound socket), however i have some >>>>> problems with outbound calls. >>>>> My application is written in C. I create a handle to esl >>>>> (esl_connect), then issue "bgapi originate ... &socket(127.0.0.1:8084async full)" in order to handle this connection just like inbound call. So >>>>> far so good. >>>>> Here come the problems: >>>>> >>>>> 1. esl creates new thread for my call - thats ok, it was expected - >>>>> but I do not receive any events on this new handle... >>>>> >>>>> 2. esl_events fails - here is some strace output: >>>>> >>>>> [pid 31588] send(5, "filter unique-id >>>>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK >>>>> filter added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", >>>>> 65535, 0) = 110 >>>>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>>>> command not found\n\n", 65535, 0) = 64 >>>>> [pid 31588] send(5, "linger", 6, 0) = 6 >>>>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>>>> command not found\n\n", 65535, 0) = 64 >>>>> >>>>> 3. how can i distinguish, whether this call is successfull or not? >>>>> >>>>> What am I doing wrong? >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Guillermo Ruiz Camauer >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/8b42cd9f/attachment.html From jkr888 at gmail.com Wed Oct 9 05:37:44 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Tue, 8 Oct 2013 21:37:44 -0400 Subject: [Freeswitch-users] Checking call is connected while bridged in lua In-Reply-To: References: <3BC7920D-E937-4D9A-80A3-9642FEFAB2BE@freeswitch.org> <9D075F7A-1FB4-4FF6-8045-BA16F30ACA84@freeswitch.org> <67C28849-43B7-4CA3-B7BB-C0168A352A88@freeswitch.org> Message-ID: How about spawn a new lua thread (luarun) before bridge, passing uuid, etc to new script. then on that thread, you check db flagged and/or control call with uuid. Never done this, but i think should works. On Tue, Sep 24, 2013 at 6:33 PM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > I don't think this is something FS can do. Let me explain: > > We have FS-A and FS-B, between which are multiple SIP providers. We are > checking that the providers between A and B don't divert calls to > non-approved vendors. So what I'm doing is: > > When relaying the call from FS-A to the provider, I'm adding a SIP header > with a unique ID, which is inserted into a DB table. > When FS-B receives the call, i retrieve the header ID and look it up in > the table and mark it as answered. > On FS-A, when the call is answered, it looks up the ID and makes sure it > has been marked as "answered" by FS-B. > If it is NOT marked, FS-A MUST disconnect the call. > > I hope the explanation was clear, and if anyone has a better idea of how > to do this, I'm all ears :) > > Thanks > > David > > > > On Tue, Sep 24, 2013 at 5:12 PM, Brian West wrote: > >> Maybe if we understood more we could probably guide the end user in a >> better way to accomplish their goals! >> >> /b >> >> On Sep 24, 2013, at 9:46 AM, Michael Collins wrote: >> >> > I'm with Brian on this one. The bridge app is epic. So the question is >> this: what are you trying to accomplish with all this Lua stuff that you >> can't do with a simple bridge? >> > >> > -MC >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/0bfeaab6/attachment-0001.html From red.rain.seven at gmail.com Wed Oct 9 05:59:27 2013 From: red.rain.seven at gmail.com (Henry Huang) Date: Tue, 8 Oct 2013 18:59:27 -0700 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: References: <525378A5.40706@gmail.com> <52542FED.1050506@gmail.com> Message-ID: Will you share the results? On Tue, Oct 8, 2013 at 2:27 PM, Nikolay Rogoshchenkov wrote: > In second question is impossible to chose "other". > > > -- > Rogoshchenkov Nikolay > > > On Tue, Oct 8, 2013 at 12:16 PM, Victor Chukalovskiy < > victor.chukalovskiy at gmail.com> wrote: > >> Bump! Not sure if my message reached the list yesterday (given all the >> "tests" today) :-) >> >> http://www.surveymonkey.com/s/J8SL9TS >> >> On 13-10-07 11:14 PM, Victor Chukalovskiy wrote: >> > Greetings! >> > >> > Everyone who runs some sort of billing for freeswitch, please provide >> > your feedback here: >> > >> > http://www.surveymonkey.com/s/J8SL9TS >> > >> > Just 4 question. Less then a minute to complete! >> > >> > Survey runs for 2-3 days. I'll post final results in this thread for >> > your benefit. >> > >> > Cheers, >> > -Victor >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/b6795304/attachment.html From victor.chukalovskiy at gmail.com Wed Oct 9 07:02:37 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 08 Oct 2013 23:02:37 -0400 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: References: <525378A5.40706@gmail.com> <52542FED.1050506@gmail.com> Message-ID: <5254C74D.9000002@gmail.com> Thanks, fixed now! On 13-10-08 05:27 PM, Nikolay Rogoshchenkov wrote: > In second question is impossible to chose "other". > > > -- > Rogoshchenkov Nikolay > > > On Tue, Oct 8, 2013 at 12:16 PM, Victor Chukalovskiy > > > wrote: > > Bump! Not sure if my message reached the list yesterday (given all the > "tests" today) :-) > > http://www.surveymonkey.com/s/J8SL9TS > > On 13-10-07 11 :14 PM, Victor Chukalovskiy wrote: > > Greetings! > > > > Everyone who runs some sort of billing for freeswitch, please > provide > > your feedback here: > > > > http://www.surveymonkey.com/s/J8SL9TS > > > > Just 4 question. Less then a minute to complete! > > > > Survey runs for 2-3 days. I'll post final results in this thread for > > your benefit. > > > > Cheers, > > -Victor > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/b57c8edc/attachment.html From victor.chukalovskiy at gmail.com Wed Oct 9 07:03:14 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 08 Oct 2013 23:03:14 -0400 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: References: <525378A5.40706@gmail.com> <52542FED.1050506@gmail.com> Message-ID: <5254C772.2020205@gmail.com> Yes, I'll share the results once survey is complete. On 13-10-08 09:59 PM, Henry Huang wrote: > Will you share the results? > > > > On Tue, Oct 8, 2013 at 2:27 PM, Nikolay Rogoshchenkov > > wrote: > > In second question is impossible to chose "other". > > > -- > Rogoshchenkov Nikolay > > > On Tue, Oct 8, 2013 at 12:16 PM, Victor Chukalovskiy > > wrote: > > Bump! Not sure if my message reached the list yesterday (given > all the > "tests" today) :-) > > http://www.surveymonkey.com/s/J8SL9TS > > On 13-10-07 11 :14 PM, Victor Chukalovskiy > wrote: > > Greetings! > > > > Everyone who runs some sort of billing for freeswitch, > please provide > > your feedback here: > > > > http://www.surveymonkey.com/s/J8SL9TS > > > > Just 4 question. Less then a minute to complete! > > > > Survey runs for 2-3 days. I'll post final results in this > thread for > > your benefit. > > > > Cheers, > > -Victor > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131008/92da437b/attachment-0001.html From gmangudai at gmail.com Wed Oct 9 07:06:28 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Wed, 9 Oct 2013 11:06:28 +0800 Subject: [Freeswitch-users] Checking call is connected while bridged in lua In-Reply-To: References: <3BC7920D-E937-4D9A-80A3-9642FEFAB2BE@freeswitch.org> <9D075F7A-1FB4-4FF6-8045-BA16F30ACA84@freeswitch.org> <67C28849-43B7-4CA3-B7BB-C0168A352A88@freeswitch.org> Message-ID: hi David, maybe i got the issue that is very similar to yours, the scenario is A calls B and B is bridged in lua script (create a new session for leg b and after B answered the call, bridge leg a and b), but when A hangs up before B could answer the call, i found it is not possilbe to have lua get noticed about the hangup. as i understand, the bridge is made in a synchronous manner, thus you can do nothing before the bridge is finished (successfully or failed), one thing that may help is by using ignore_early_media while originating the b leg call, but this causes other problems for my issue, such as unable to check the result of no answer, so im still seeking a better solution for this. 2013/10/9 Johny Kadarisman Kwan > How about spawn a new lua thread (luarun) before bridge, passing uuid, etc > to new script. then on that thread, you check db flagged and/or control > call with uuid. > > Never done this, but i think should works. > > > On Tue, Sep 24, 2013 at 6:33 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello, >> >> I don't think this is something FS can do. Let me explain: >> >> We have FS-A and FS-B, between which are multiple SIP providers. We are >> checking that the providers between A and B don't divert calls to >> non-approved vendors. So what I'm doing is: >> >> When relaying the call from FS-A to the provider, I'm adding a SIP header >> with a unique ID, which is inserted into a DB table. >> When FS-B receives the call, i retrieve the header ID and look it up in >> the table and mark it as answered. >> On FS-A, when the call is answered, it looks up the ID and makes sure it >> has been marked as "answered" by FS-B. >> If it is NOT marked, FS-A MUST disconnect the call. >> >> I hope the explanation was clear, and if anyone has a better idea of how >> to do this, I'm all ears :) >> >> Thanks >> >> David >> >> >> >> On Tue, Sep 24, 2013 at 5:12 PM, Brian West wrote: >> >>> Maybe if we understood more we could probably guide the end user in a >>> better way to accomplish their goals! >>> >>> /b >>> >>> On Sep 24, 2013, at 9:46 AM, Michael Collins wrote: >>> >>> > I'm with Brian on this one. The bridge app is epic. So the question is >>> this: what are you trying to accomplish with all this Lua stuff that you >>> can't do with a simple bridge? >>> > >>> > -MC >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/721e8621/attachment.html From GB at cm.nl Wed Oct 9 12:29:22 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Wed, 9 Oct 2013 10:29:22 +0200 Subject: [Freeswitch-users] Multiple contexts per sip profile In-Reply-To: References: Message-ID: Isn't it possible to load more than one context? When entering the initial inbound context. It should be possible to transfer to another, shouldn't it? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Tuesday, October 8, 2013 5:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple contexts per sip profile Please have a look into here http://wiki.freeswitch.org/wiki/Multi-tenant Lloyd On Tue, Oct 8, 2013 at 10:51 AM, Grant Bagdasarian > wrote: Hello, I'm trying to configure FreeSWITCH to have multiple contexts per sip profile like below. But this does not seem to work. The reason I want this, is to have each incoming called number to have its own context in which extensions can be created. The problem with using extensions and only one context is I can't defined more than one extension with the same name. FS seems to overwrite or ignore ones that do. Is this even possible? If so, what do I need to change? Is there perhaps a better way for doing this? sip_profile/inbound.xml: dialplan/inbound.xml: dialplan/inbound/1000.xml: dialplan/inbound/1001.xml: _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/012d6996/attachment-0001.html From jleung at v10networks.ca Wed Oct 9 12:45:34 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 09 Oct 2013 16:45:34 +0800 Subject: [Freeswitch-users] esl dialer Message-ID: <001801cec4cb$f05fe7eb$7c07000a@smb.curriegrad2004.ca> Guillermo Ruiz Camauer wrote: I handle all ESL communications from just one thread which receives and processes or generates all events and commands. I do have a timer for each call which I use for certain timeouts (user inputs and ring time). Guillermo On Tue, Oct 8, 2013 at 4:20 AM, Marcin S wrote: > Ok, I will try this approach. Am I right, that in order to handle oubound > calls similiar to inbound, I will have to spawn new thread for each call? > > > 2013/10/5 Guillermo Ruiz Camauer > >> I have a Dialer that uses ESL from C, but I make an inbound connection to >> Freeswitch. It is a socket which I keep open permanently. >> I recommend reading chapter 4 of the FreeSwitch Cookbook. >> >> Guillermo >> >> >> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: >> >>> Hello, >>> >>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>> inbound calls is easy (using esl outbound socket), however i have some >>> problems with outbound calls. >>> My application is written in C. I create a handle to esl (esl_connect), >>> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >>> order to handle this connection just like inbound call. So far so good. >>> Here come the problems: >>> >>> 1. esl creates new thread for my call - thats ok, it was expected - but >>> I do not receive any events on this new handle... >>> >>> 2. esl_events fails - here is some strace output: >>> >>> [pid 31588] send(5, "filter unique-id >>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter >>> added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = >>> 110 >>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> [pid 31588] send(5, "linger", 6, 0) = 6 >>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> >>> 3. how can i distinguish, whether this call is successfull or not? >>> >>> What am I doing wrong? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer From jleung at v10networks.ca Wed Oct 9 12:56:02 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 09 Oct 2013 16:56:02 +0800 Subject: [Freeswitch-users] esl dialer Message-ID: <001901cec4cd$6530945e$7c07000a@smb.curriegrad2004.ca> Sorry for that empty reply, didn't lock the phone before i placed it in my pocket. Jeff Leung wrote: Guillermo Ruiz Camauer wrote: I handle all ESL communications from just one thread which receives and processes or generates all events and commands. I do have a timer for each call which I use for certain timeouts (user inputs and ring time). Guillermo On Tue, Oct 8, 2013 at 4:20 AM, Marcin S wrote: > Ok, I will try this approach. Am I right, that in order to handle oubound > calls similiar to inbound, I will have to spawn new thread for each call? > > > 2013/10/5 Guillermo Ruiz Camauer > >> I have a Dialer that uses ESL from C, but I make an inbound connection to >> Freeswitch. It is a socket which I keep open permanently. >> I recommend reading chapter 4 of the FreeSwitch Cookbook. >> >> Guillermo >> >> >> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: >> >>> Hello, >>> >>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>> inbound calls is easy (using esl outbound socket), however i have some >>> problems with outbound calls. >>> My application is written in C. I create a handle to esl (esl_connect), >>> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >>> order to handle this connection just like inbound call. So far so good. >>> Here come the problems: >>> >>> 1. esl creates new thread for my call - thats ok, it was expected - but >>> I do not receive any events on this new handle... >>> >>> 2. esl_events fails - here is some strace output: >>> >>> [pid 31588] send(5, "filter unique-id >>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter >>> added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = >>> 110 >>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> [pid 31588] send(5, "linger", 6, 0) = 6 >>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> >>> 3. how can i distinguish, whether this call is successfull or not? >>> >>> What am I doing wrong? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From andrew at cassidywebservices.co.uk Wed Oct 9 13:37:38 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 9 Oct 2013 10:37:38 +0100 Subject: [Freeswitch-users] Multiple contexts per sip profile In-Reply-To: References: Message-ID: Move the include outside of the context definition as shown below. You were creating contexts within a context which I'm not sure is possible. * * *dialplan/inbound.xml:* ** ** **** **** **** ** ** ** **** **** **** **** **** **** **** **** ** On 9 October 2013 09:29, Grant Bagdasarian wrote: > Isn?t it possible to load more than one context?**** > > ** ** > > When entering the initial inbound context. It should be possible to > transfer to another, shouldn?t it?**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd > Aloysius > *Sent:* Tuesday, October 8, 2013 5:00 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Multiple contexts per sip profile**** > > ** ** > > Please have a look into here**** > > ** ** > > http://wiki.freeswitch.org/wiki/Multi-tenant**** > > > **** > > Lloyd**** > > * ***** > > ** ** > > On Tue, Oct 8, 2013 at 10:51 AM, Grant Bagdasarian wrote:**** > > Hello,**** > > **** > > I?m trying to configure FreeSWITCH to have multiple contexts per sip > profile like below.**** > > **** > > But this does not seem to work.**** > > The reason I want this, is to have each incoming called number to have its > own context in which extensions can be created. **** > > The problem with using extensions and only one context is I can?t defined > more than one extension with the same name. FS seems to overwrite or ignore > ones that do.**** > > **** > > Is this even possible? If so, what do I need to change? Is there perhaps a > better way for doing this?**** > > **** > > *sip_profile/inbound.xml:***** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > *dialplan/inbound.xml:***** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > *dialplan/inbound/1000.xml:***** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > data="/usr/src/freeswitch/sounds/fs-hello-world-for-1000.wav"/>**** > > **** > > **** > > **** > > **** > > **** > > **** > > *dialplan/inbound/1001.xml:***** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > data="/usr/src/freeswitch/sounds/fs-hello-world-for-1001.wav"/>**** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/c49eab8c/attachment-0001.html From GB at cm.nl Wed Oct 9 13:52:19 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Wed, 9 Oct 2013 11:52:19 +0200 Subject: [Freeswitch-users] Multiple contexts per sip profile In-Reply-To: References: Message-ID: Ahhhh, right! Thank you, that worked! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy Sent: Wednesday, October 9, 2013 11:38 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple contexts per sip profile Move the include outside of the context definition as shown below. You were creating contexts within a context which I'm not sure is possible. dialplan/inbound.xml: On 9 October 2013 09:29, Grant Bagdasarian > wrote: Isn't it possible to load more than one context? When entering the initial inbound context. It should be possible to transfer to another, shouldn't it? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Tuesday, October 8, 2013 5:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple contexts per sip profile Please have a look into here http://wiki.freeswitch.org/wiki/Multi-tenant Lloyd On Tue, Oct 8, 2013 at 10:51 AM, Grant Bagdasarian > wrote: Hello, I'm trying to configure FreeSWITCH to have multiple contexts per sip profile like below. But this does not seem to work. The reason I want this, is to have each incoming called number to have its own context in which extensions can be created. The problem with using extensions and only one context is I can't defined more than one extension with the same name. FS seems to overwrite or ignore ones that do. Is this even possible? If so, what do I need to change? Is there perhaps a better way for doing this? sip_profile/inbound.xml: dialplan/inbound.xml: dialplan/inbound/1000.xml: dialplan/inbound/1001.xml: _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [cid:~WRD000.jpg] T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/d91a6848/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/d91a6848/attachment-0001.jpg From noc at sonerep.com Wed Oct 9 13:04:07 2013 From: noc at sonerep.com (Dzexolokpli AMOUZOU) Date: Wed, 09 Oct 2013 10:04:07 +0100 Subject: [Freeswitch-users] [freeswitch-codec] Message-ID: <52551C07.306@sonerep.com> Hello, I have installed freeswitch and kamailio on debian 7.0 wheezy referencing to http://nil.uniza.sk/sip/kamailio-33-and-freeswitch-122-interconnection-voicemail-and-conference-services-debian-squeeze-60-64bit-tutorial. I notice that the quality of the voice using kamailio is better than it using freeswitch. searching the problem source, I find that codec G722 is not include in freeswich. How can I include this codec in freeswwitch modules. thanks you in advance. From ms4esl at gmail.com Wed Oct 9 14:06:12 2013 From: ms4esl at gmail.com (Marcin S) Date: Wed, 9 Oct 2013 12:06:12 +0200 Subject: [Freeswitch-users] esl dialer In-Reply-To: References: Message-ID: I am writing call center app, dialer and ivr are part of it. Nevertheless, I still have no idea how to control inbound as well as outbound calls with one "socket inbound" esl handle... 2013/10/9 Guillermo Ruiz Camauer > Are you writing a Dialer or an IVR? A Dialer doesn't really need to > answer calls. You can always have one program (or thread) with an inbound > socket and another with an outbound socket. > > Guillermo > > > On Tue, Oct 8, 2013 at 3:30 PM, Marcin S wrote: > >> I was considering this approach, however I don't know how to detect and >> answer a call in "socket inboud" esl mode. >> >> >> 2013/10/8 Guillermo Ruiz Camauer >> >>> I handle all ESL communications from just one thread which receives and >>> processes or generates all events and commands. I do have a timer for each >>> call which I use for certain timeouts (user inputs and ring time). >>> >>> Guillermo >>> >>> >>> On Tue, Oct 8, 2013 at 4:20 AM, Marcin S wrote: >>> >>>> Ok, I will try this approach. Am I right, that in order to handle >>>> oubound calls similiar to inbound, I will have to spawn new thread for each >>>> call? >>>> >>>> >>>> 2013/10/5 Guillermo Ruiz Camauer >>>> >>>>> I have a Dialer that uses ESL from C, but I make an inbound connection >>>>> to Freeswitch. It is a socket which I keep open permanently. >>>>> I recommend reading chapter 4 of the FreeSwitch Cookbook. >>>>> >>>>> Guillermo >>>>> >>>>> >>>>> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing >>>>>> with inbound calls is easy (using esl outbound socket), however i have some >>>>>> problems with outbound calls. >>>>>> My application is written in C. I create a handle to esl >>>>>> (esl_connect), then issue "bgapi originate ... &socket(127.0.0.1:8084async full)" in order to handle this connection just like inbound call. So >>>>>> far so good. >>>>>> Here come the problems: >>>>>> >>>>>> 1. esl creates new thread for my call - thats ok, it was expected - >>>>>> but I do not receive any events on this new handle... >>>>>> >>>>>> 2. esl_events fails - here is some strace output: >>>>>> >>>>>> [pid 31588] send(5, "filter unique-id >>>>>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>>>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK >>>>>> filter added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", >>>>>> 65535, 0) = 110 >>>>>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>>>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>>>>> command not found\n\n", 65535, 0) = 64 >>>>>> [pid 31588] send(5, "linger", 6, 0) = 6 >>>>>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>>>>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>>>>> command not found\n\n", 65535, 0) = 64 >>>>>> >>>>>> 3. how can i distinguish, whether this call is successfull or not? >>>>>> >>>>>> What am I doing wrong? >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/cd3c5d3a/attachment.html From GB at cm.nl Wed Oct 9 14:58:04 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Wed, 9 Oct 2013 12:58:04 +0200 Subject: [Freeswitch-users] Multiple contexts per sip profile In-Reply-To: References: Message-ID: When setting the transfer_on_fail variable, does this only apply when bridging the call? The call hangs up immediately when transferring to another context fails. Is it possible to recover from a failed transfer? 2013-10-09 12:47:45.543748 [NOTICE] switch_ivr.c:1831 Transfer sofia/inbound/31700000000 at 192.168.1.10:5060 to XML[start at 1002] 2013-10-09 12:47:45.543748 [DEBUG] switch_core_state_machine.c:480 (sofia/inbound/31700000000 at 192.168.1.10:5060) State EXECUTE going to sleep 2013-10-09 12:47:45.543748 [DEBUG] switch_core_state_machine.c:418 (sofia/inbound/31700000000 at 192.168.1.10:5060) Running State Change CS_ROUTING 2013-10-09 12:47:45.543748 [DEBUG] switch_core_state_machine.c:473 (sofia/inbound/31700000000 at 192.168.1.10:5060) State ROUTING 2013-10-09 12:47:45.543748 [DEBUG] mod_sofia.c:137 sofia/inbound/31700000000 at 192.168.1.10:5060 SOFIA ROUTING 2013-10-09 12:47:45.543748 [DEBUG] switch_core_state_machine.c:117 sofia/inbound/31700000000 at 192.168.1.10:5060 Standard ROUTING 2013-10-09 12:47:45.543748 [INFO] mod_dialplan_xml.c:558 Processing 31700000000 <31700000000>->start in context 1002 2013-10-09 12:47:45.543748 [WARNING] mod_dialplan_xml.c:588 Context 1002 not found 2013-10-09 12:47:45.543748 [INFO] switch_core_state_machine.c:192 No Route, Aborting 2013-10-09 12:47:45.543748 [NOTICE] switch_core_state_machine.c:193 Hangup sofia/inbound/31700000000 at 192.168.1.10:5060 [CS_ROUTING] [NO_ROUTE_DESTINATION] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: Wednesday, October 9, 2013 11:52 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple contexts per sip profile Ahhhh, right! Thank you, that worked! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy Sent: Wednesday, October 9, 2013 11:38 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple contexts per sip profile Move the include outside of the context definition as shown below. You were creating contexts within a context which I'm not sure is possible. dialplan/inbound.xml: On 9 October 2013 09:29, Grant Bagdasarian > wrote: Isn't it possible to load more than one context? When entering the initial inbound context. It should be possible to transfer to another, shouldn't it? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Tuesday, October 8, 2013 5:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Multiple contexts per sip profile Please have a look into here http://wiki.freeswitch.org/wiki/Multi-tenant Lloyd On Tue, Oct 8, 2013 at 10:51 AM, Grant Bagdasarian > wrote: Hello, I'm trying to configure FreeSWITCH to have multiple contexts per sip profile like below. But this does not seem to work. The reason I want this, is to have each incoming called number to have its own context in which extensions can be created. The problem with using extensions and only one context is I can't defined more than one extension with the same name. FS seems to overwrite or ignore ones that do. Is this even possible? If so, what do I need to change? Is there perhaps a better way for doing this? sip_profile/inbound.xml: dialplan/inbound.xml: dialplan/inbound/1000.xml: dialplan/inbound/1001.xml: _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [cid:image001.jpg at 01CEC4EF.312306D0] T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image001.jpg Type: image/jpeg Size: 823 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/210c3578/attachment-0001.jpg From coslovich.innovats at yahoo.it Wed Oct 9 11:59:30 2013 From: coslovich.innovats at yahoo.it (Andrea Coslovich) Date: Wed, 9 Oct 2013 08:59:30 +0100 (BST) Subject: [Freeswitch-users] FreeTDM - Using ftdm_channel_wait with FTDM_EVENTS In-Reply-To: <1381242960.47963.YahooMailNeo@web172105.mail.ir2.yahoo.com> References: <1381135949.34409.YahooMailNeo@web172103.mail.ir2.yahoo.com> <1381235486.41730.YahooMailNeo@web172102.mail.ir2.yahoo.com> <1381242960.47963.YahooMailNeo@web172105.mail.ir2.yahoo.com> Message-ID: <1381305570.72484.YahooMailNeo@web172103.mail.ir2.yahoo.com> Dear guys, I'm new to the list so first of all let me thank the FreeSWITCH developers for their great work. But now I must come to the point. I have an A104 Sangoma board and I'm developing an application based on FreeTDM library. I've started by using and modifying the examples "freeswitch/libs/freetdm/src/test*.c" and now I'm able to retreive and send the audio data from/to the board. I'm using the "ftdm_channel_wait" to know when I can read/write from the freetdm's queues and all seems to work well. But I have a big issue when I try to use "ftdm_channel_wait" with the flag "FTDM_EVENTS" (flags=(ftdm_wait_flags_t)FTDM_EVENTS). Basically, what happen is that the "ftdm_channel_wait(ftdmchan,&flags,-1)" never exits even if the channel receives DTMF or is been hung-up. I've also tried to use "ftdm_channel_command(ftdmchan,FTDM_COMMAND_ENABLE_DTMF_DETECT,NULL)" before the "ftdm_channel_wait" but nothing changed. Now, my questions are the following: 1. What kind of events can cause the exit of "ftdm_channel_wait" function? I've seen the "ftdm_event_type_t" that defines "FTDM_EVENT_DTMF" and "FTDM_EVENT_OOB". Is it the answer to my question? 2. Is there something that I need to do before call "ftdm_channel_wait" to enable the "events generation"? I hope to be in the right place for these questions. Otherwise, please let me know where I can find someone that can help me. I haven't put a lot in this mail because I wasn't really sure if this is the right mail-list but, of course, if you need more information and details to help me don't esitate to ask. Excuse me for my bad english. Have a nice day! Andrea -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/c0db61f6/attachment.html From adahary at gmail.com Wed Oct 9 17:38:31 2013 From: adahary at gmail.com (adahary) Date: Wed, 9 Oct 2013 06:38:31 -0700 (PDT) Subject: [Freeswitch-users] Dialplan 'set' variable not found in CDR Message-ID: <1381325911740-7595574.post@n2.nabble.com> Hi, I'm trying to pass some custom variables from dialplan to cdr with no success. here is an example of my dialplan with the 'set' of some custom variable 'x_dialplan_mode'. why I cannot see this variable 'x_dialplan_mode' in the CDR XML ?
regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialplan-set-variable-not-found-in-CDR-tp7595574.html Sent from the freeswitch-users mailing list archive at Nabble.com. From callum.guy at x-on.co.uk Wed Oct 9 18:05:45 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 9 Oct 2013 15:05:45 +0100 Subject: [Freeswitch-users] Prevent A leg from hangup after bridge with inbound ESL socket In-Reply-To: <1FFF97C269757C458224B7C895F35F15153DF1@cantor.std.visionutv.se> References: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin> <569384504C492C4580E88B5D54DFEAEA30CAFD3530@jjex01.jajah.dublin> <569384504C492C4580E88B5D54DFEAEA30CB11E128@jjex01.jajah.dublin> <1FFF97C269757C458224B7C895F35F15153DF1@cantor.std.visionutv.se> Message-ID: Hi All, Using version 1.2.9 Just wondering if this ever got resolved as i am having a similar issue. I am also using ESL inbound and my event driven script will answer the call, set the ringback to a UK tone, play 1 second of silence, then issue the bridge as follows: {ignore_early_media=true,continue_on_fail=true,park_after_bridge=true,hangup_after_bridge=false}sofia/external/ 200010006 at sipserv.net Once this has all gone through everything works correctly when the user is available. When busy however the calling channel is ending the call rather than returning to park. I had anticipated collecting the CHANNEL_EXECUTE_COMPLETE event for the bridge command, reading the variable_DIALSTATUS for "BUSY" and then playing busy tone but it is just hanging up. Is there a clean way to resolve this? Even if its just setting the "busytone" equivalent of "ringback" that would probably do? Any help would be appreciated, Thanks, Callum ______________________________ Callum Guy Developer X-on Framlingham Technology Centre Station Road, Framlingham, Suffolk, IP13 9EZ T 0333 332 0116 E callum.guy at x-on.co.uk X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD Company Registration No. 2578478 This email has been sent from X-on.The contents and attachments are confidential to the sender and the intended addressees.If the message is received by anyone other than the addressee please return the message to the sender by replying to it and then delete the message from your computer without copying or disclosing the contents to anyone.Opinions, conclusions and statements of intent in this email are those of the sender and do not bind X-on unless confirmed by authorised representatives independently of this message.While best endeavours have been taken to avoid transmission of viruses, it is the responsibility of the recipient to scan for these.Please note emails sent to and from X-on are routinely monitored for record keeping and quality control, to ensure regulatory compliance and prevent unauthorised use of our systems. Please consider the environment before printing this email. On 3 September 2012 10:28, Peter Olsson wrote: > try park_after_bridge=true, it should park the call again after the > bridge, and keep it alive. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Alex Massover [ > alex at jajah.com] > Skickat: den 3 september 2012 09:00 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Prevent A leg from hangup after bridge with > inbound ESL socket > > Hi, > > We tried that, it doesn't do a trick, the bridge app still assumes > success, even if 180 is not forwarded to A leg. > But I think the problem is related to ESL socket, especially to inbound > socket. I'm pretty sure that all these things work with dailplan, and as > far as I remember even with outbound socket it's much easier. But looks > like outbound socket bypass some of these flags, as channel is control by > XML dialplan, but then is bridged by ESL API and not sure what exactly > happens. I understand that hangup_after_bridge=false may not work, as it's > not clear what should happen with the channel in case of inbound socket (as > dialplan is ended), but no reason for park_after_bridge=true not to park a > channel. > > Maybe it worth to fill a bug in JIRA about park_after_bridge=true and ESL > inbound socket. I'll do some more clear tests and will fill one. > > BR, Alex. > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: Thursday, August 30, 2012 7:17 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Prevent A leg from hangup after bridge > with inbound ESL socket > > If that's the case then you also need ignore_early_media=true: > data="{ignore_early_media=true}sofia/internal/1001@$${domain}"/> > > If you don't ignore early media then the bridge app assumes that when it > receives media from the far end that the bridge is "successful" even if you > don't actually get a 200OK. The caveat is that since you're ignoring early > media (i.e. ringing) from the B leg that you will need to supply some sort > of ringing indicator to the A leg. The good news is that you can do > whatever you want; just use the ring_back chan var. > > -MC > On Thu, Aug 30, 2012 at 12:07 AM, Alex Massover alex at jajah.com>> wrote: > Hi Michael, > > Thanks, that works in scenario when B legs response with 100 and then > let's say 486. But if B legs do ringing, i.e. 100, 180/183, 486 it doesn't > work. > > I found this in wiki "By the way, you'll be unable to rewrite the hangup > cause for a bridge that gets a 180 or 183 packet from the gateway before > getting a 4xx, 5xx or 6xx packet (because those bridges don't 'fail')." > > I understand that continue_on_fail won't help with this scenario. I see > that's a popular topic in the list, but nobody got a solution. > > BR, Alex. > > From: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [mailto: > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael > Collins > Sent: Wednesday, August 29, 2012 6:21 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Prevent A leg from hangup after bridge > with inbound ESL socket > > Try this: > http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail > > -MC > On Wed, Aug 29, 2012 at 6:57 AM, Alex Massover alex at jajah.com>> wrote: > Hi, > > I have a very simple dialplan that just do park for incoming calls. All > rest of leg management is done via ESL inbound socket. > > I'm trying to do the same behavior like in this dialplan example, but from > ESL inbound socket: > > > > > The problem is with bridge API, if B leg doesn't answer (e.g. 404, or > busy), A leg disconnects. But I'm trying to prevent A leg from > disconnecting in order to do bridge to other place. > > Looks like hangup_after_bridge=false, park_after_bridge=true, > transfer_after_bridge etc don't have any effect when bridge done from > inbound socket. A leg disconnects always. > > Is there any way to keep A leg after bridge with inbound socket? I'm aware > of originate, but prefer to user bridge. > > > > > -- > Best Regards, > Alex Massover > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > !DSPAM:504453d832762136219851! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/4142b84d/attachment-0001.html From steveu at coppice.org Wed Oct 9 18:50:06 2013 From: steveu at coppice.org (Steve Underwood) Date: Wed, 09 Oct 2013 22:50:06 +0800 Subject: [Freeswitch-users] [freeswitch-codec] In-Reply-To: <52551C07.306@sonerep.com> References: <52551C07.306@sonerep.com> Message-ID: <52556D1E.4020600@coppice.org> On 10/09/2013 05:04 PM, Dzexolokpli AMOUZOU wrote: > Hello, > > > I have installed freeswitch and kamailio on debian 7.0 wheezy > referencing to > > http://nil.uniza.sk/sip/kamailio-33-and-freeswitch-122-interconnection-voicemail-and-conference-services-debian-squeeze-60-64bit-tutorial. > > I notice that the quality of the voice using kamailio is better than it > using freeswitch. > searching the problem source, I find that codec G722 is not include in > freeswich. > > How can I include this codec in freeswwitch modules. > > thanks you in advance. > G.722 has always been included in FreeSwitch. Perhaps your configuration files are not offering this codec when negotiating the start of a call. Steve From krice at freeswitch.org Wed Oct 9 18:58:43 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 09 Oct 2013 09:58:43 -0500 Subject: [Freeswitch-users] Weekly News Notes Etc Message-ID: Hey Guys! Don?t forget to join us today for ClueCon weekly on 888 at 1P Eastern or 10A Pacific! If you haven?t joined the Groups on Google+, Follow FreeSWITCH here https://plus.google.com/communities/110013892794022037793 Follow ClueCon here https://plus.google.com/communities/108955134544785139154 And don?t forget to follow us on Twitter @FreeSWITCH_wire -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch @Ken4VoIP via Twitter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/8e62f9a1/attachment.html From peter at hartmanncomputer.com Wed Oct 9 21:13:50 2013 From: peter at hartmanncomputer.com (Peter Hartmann) Date: Wed, 9 Oct 2013 13:13:50 -0400 Subject: [Freeswitch-users] test Message-ID: Heloooooo! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com From rafal.gwizdala at gmail.com Wed Oct 9 21:43:16 2013 From: rafal.gwizdala at gmail.com (Rafal Gwizdala) Date: Wed, 9 Oct 2013 19:43:16 +0200 Subject: [Freeswitch-users] Weekly News Notes Etc In-Reply-To: References: Message-ID: Guys, Does anyone record these conferences? I'm trying to join and listen but the conf starts at 7PM for me (in Poland) and you know, it's quite difficult to concentrate and listen with a family around. It would be better to listen to a recorded presentation later... Best regards RG On Wed, Oct 9, 2013 at 4:58 PM, Ken Rice wrote: > Hey Guys! Don?t forget to join us today for ClueCon weekly on 888 at 1P > Eastern or 10A Pacific! > > If you haven?t joined the Groups on Google+, Follow FreeSWITCH here > https://plus.google.com/communities/110013892794022037793 > Follow ClueCon here > https://plus.google.com/communities/108955134544785139154 > > And don?t forget to follow us on Twitter @FreeSWITCH_wire > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > @Ken4VoIP via Twitter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/2aac27e0/attachment.html From jnvines at gmail.com Wed Oct 9 22:06:16 2013 From: jnvines at gmail.com (Nick Vines) Date: Wed, 9 Oct 2013 11:06:16 -0700 Subject: [Freeswitch-users] Weekly News Notes Etc In-Reply-To: References: Message-ID: Most of them are here http://torrents.freeswitch.org/ On Wed, Oct 9, 2013 at 10:43 AM, Rafal Gwizdala wrote: > Guys, > Does anyone record these conferences? I'm trying to join and listen but > the conf starts at 7PM for me (in Poland) and you know, it's quite > difficult to concentrate and listen with a family around. It would be > better to listen to a recorded presentation later... > Best regards > RG > > > > > > > On Wed, Oct 9, 2013 at 4:58 PM, Ken Rice wrote: > >> Hey Guys! Don?t forget to join us today for ClueCon weekly on 888 at 1P >> Eastern or 10A Pacific! >> >> If you haven?t joined the Groups on Google+, Follow FreeSWITCH here >> https://plus.google.com/communities/110013892794022037793 >> Follow ClueCon here >> https://plus.google.com/communities/108955134544785139154 >> >> And don?t forget to follow us on Twitter @FreeSWITCH_wire >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> @Ken4VoIP via Twitter >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/89b46845/attachment.html From acrow at integrafin.co.uk Wed Oct 9 22:49:04 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 09 Oct 2013 19:49:04 +0100 Subject: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions In-Reply-To: <52541FB2.9090602@integrafin.co.uk> References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch> <9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch> <52541FB2.9090602@integrafin.co.uk> Message-ID: <5255A520.8040909@integrafin.co.uk> Hi, JIRA issue posted, FS-5684. Can the original reporters please update/add details? Thanks Alex ----Original Message---- *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions *Date:* Tue, 08 Oct 2013 16:07:30 +0100 *From:* Alex Crow *To:* freeswitch-users at lists.freeswitch.org > Hi, > > I can confirm that the issue affects me in version: > > FreeSWITCH Version > 1.2.11-n20130817T172515Z-1~wheezy+1+git~20130816T225403Z~8566ffa82a > (-n20130817T172515Z-1~wheezy+1git 8566ffa 2013-08-16 22:54:03Z) > > but not in: > > FreeSWITCH Version 1.2.8+git~20130405T052920Z~57fb368b32 (git 57fb368 > 2013-04-05 05:29:20Z) > > if that helps anyone. > > These are production systems though so I've had to disable the > relevant codecs. > > Cheers > > Alex > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in > all recent FreeSWITCH versions > *Date:* Sun, 4 Aug 2013 02:43:27 +0000 > *From:* Moishe Grunstein > *To:* FreeSWITCH Users Help > > > >> I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related. >> >> >> Thanks, >> >> Moishe Grunstein >> Tornado Computer Systems, Inc. >> 212.400.7650 888.IPPBX.US >> Service Request Email:support at nysolutions.com >> Polycom Certified VAR >> Microsoft Small Business Specialist, Cisco SMB Select Certified >> >> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS >> >> -----Original Message----- >> From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah >> Sent: Saturday, August 03, 2013 10:34 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions >> >> I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. >> Thanks and I'll make sure I get it through the process. >> >> Cheers >> On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson wrote: >> >>> On Sat, Aug 3, 2013 at 3:31 PM, Emrah wrote: >>>> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy. >>>> >>>> The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end? >>> This is what you need to know: >>> https://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >>> >>> Best, >>> Gabe >>> >>> ______________________________________________________________________ >>> ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> rs >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > "Transact" is operated by Integrated Financial Arrangements plc. 29 > Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 > 5300. (Registered office: as above; Registered in England and Wales > under number: 3727592). Authorised and regulated by the Financial > Conduct Authority (entered on the Financial Services Register; no. 190856). -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc. 29 Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 5300. (Registered office: as above; Registered in England and Wales under number: 3727592). Authorised and regulated by the Financial Conduct Authority (entered on the Financial Services Register; no. 190856). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/0ba4684c/attachment-0001.html From tru083 at yahoo.com Thu Oct 10 02:55:10 2013 From: tru083 at yahoo.com (D D) Date: Wed, 9 Oct 2013 15:55:10 -0700 (PDT) Subject: [Freeswitch-users] Where are the weekly conference calls archived now? Message-ID: <1381359310.2679.YahooMailNeo@web120704.mail.ne1.yahoo.com> Hi, Where are the weekly conference calls archived now? Thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/66f8eded/attachment.html From krice at freeswitch.org Thu Oct 10 03:50:49 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 9 Oct 2013 18:50:49 -0500 Subject: [Freeswitch-users] Where are the weekly conference calls archived now? In-Reply-To: <1381359310.2679.YahooMailNeo@web120704.mail.ne1.yahoo.com> References: <1381359310.2679.YahooMailNeo@web120704.mail.ne1.yahoo.com> Message-ID: same place they just need to getupdated on the wiki, i'll do that tonight Ken Sent from my iPad On Oct 9, 2013, at 17:55, D D wrote: > Hi, > > Where are the weekly conference calls archived now? > > Thanks! > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131009/d4f41b48/attachment.html From nandy1925 at gmail.com Thu Oct 10 05:18:54 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 10 Oct 2013 09:18:54 +0800 Subject: [Freeswitch-users] Playing announcement file before MOH Message-ID: Hello everyone, I like to play an announcement file before playing music-on-hold. Is this possible? Any hint? Tks, Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/996aa654/attachment.html From ascensiontech at gmail.com Thu Oct 10 02:14:39 2013 From: ascensiontech at gmail.com (Peter Hartmann) Date: Wed, 9 Oct 2013 18:14:39 -0400 Subject: [Freeswitch-users] old calls hanging around Message-ID: Hi, First, thanks for Freeswitch! I'm experiencing an issue where 'show calls' returns several calls that aren't actually happening both inbound and outbound. Has anyone seen this before? Rebooting the handset (Polycom IP 550) associated with that extension has no effect so it seems in FS. freeswitch at internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,unknown ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 9 total. Running: FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 2013-10-02 21:30:46Z) Thanks much! From cal.leeming at simplicitymedialtd.co.uk Thu Oct 10 07:46:24 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 10 Oct 2013 04:46:24 +0100 Subject: [Freeswitch-users] Playing announcement file before MOH In-Reply-To: References: Message-ID: Have a look into some of these variables/options below, use google+wiki; uuid_displace - allows you to inject audio into a channel temp_hold_music - allows you specify music during transfer hold campon - places call into parking when on hold (not 100% on this, needs clarification) You could also modify the MOH file to play the message you require at the desired intervals. Realistically I think this would be the best way, as this will give you much more control over the quality delivered (fading, volume, etc). Hope this helps Cal On Thu, Oct 10, 2013 at 2:18 AM, Nandy Dagondon wrote: > Hello everyone, > > I like to play an announcement file before playing music-on-hold. Is this > possible? Any hint? > > Tks, > Nandy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/2f71ca5d/attachment.html From yehavi.bourvine at gmail.com Thu Oct 10 08:37:21 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 10 Oct 2013 07:37:21 +0300 Subject: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions In-Reply-To: <5255A520.8040909@integrafin.co.uk> References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch> <9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch> <52541FB2.9090602@integrafin.co.uk> <5255A520.8040909@integrafin.co.uk> Message-ID: It is FS-5864... __Yehavi: 2013/10/9 Alex Crow > Hi, > > JIRA issue posted, FS-5684. Can the original reporters please update/add > details? > > Thanks > > > Alex > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in all > recent > FreeSWITCH versions > *Date:* Tue, 08 Oct 2013 16:07:30 +0100 > *From:* Alex Crow > *To:* freeswitch-users at lists.freeswitch.org > > > > Hi, > > I can confirm that the issue affects me in version: > > FreeSWITCH Version > 1.2.11-n20130817T172515Z-1~wheezy+1+git~20130816T225403Z~8566ffa82a > (-n20130817T172515Z-1~wheezy+1git 8566ffa 2013-08-16 22:54:03Z) > > but not in: > > FreeSWITCH Version 1.2.8+git~20130405T052920Z~57fb368b32 (git 57fb368 > 2013-04-05 05:29:20Z) > > if that helps anyone. > > These are production systems though so I've had to disable the relevant > codecs. > > Cheers > > Alex > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in > all recent FreeSWITCH versions > *Date:* Sun, 4 Aug 2013 02:43:27 +0000 > *From:* Moishe Grunstein > *To:* FreeSWITCH Users Help > > > > I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related. > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Emrah > Sent: Saturday, August 03, 2013 10:34 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions > > I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. > Thanks and I'll make sure I get it through the process. > > Cheers > On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson wrote: > > > On Sat, Aug 3, 2013 at 3:31 PM, Emrah wrote: > > I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy. > > The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end? > > This is what you need to know:https://wiki.freeswitch.org/wiki/Reporting_Bugs > > > Best, > Gabe > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rshttp://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > "Transact" is operated by Integrated Financial Arrangements plc. 29 > Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 > 5300. (Registered office: as above; Registered in England and Wales > under number: 3727592). Authorised and regulated by the Financial > Conduct Authority (entered on the Financial Services Register; no. 190856). > > > > > > -- > > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > "Transact" is operated by Integrated Financial Arrangements plc. 29 > Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 > 5300. (Registered office: as above; Registered in England and Wales > under number: 3727592). Authorised and regulated by the Financial > Conduct Authority (entered on the Financial Services Register; no. 190856). > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/3fc1bcf6/attachment-0001.html From nandy1925 at gmail.com Thu Oct 10 08:54:09 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 10 Oct 2013 12:54:09 +0800 Subject: [Freeswitch-users] Playing announcement file before MOH In-Reply-To: References: Message-ID: Hi Cal, The "please_hold" vanilla dialplan in features.xml is what I initially. The tips you shared - someday I can use them. Thanks a lot! /Nandy On Thu, Oct 10, 2013 at 11:46 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Have a look into some of these variables/options below, use google+wiki; > > uuid_displace - allows you to inject audio into a channel > temp_hold_music - allows you specify music during transfer hold > campon - places call into parking when on hold (not 100% on this, needs > clarification) > > You could also modify the MOH file to play the message you require at the > desired intervals. Realistically I think this would be the best way, as > this will give you much more control over the quality delivered (fading, > volume, etc). > > Hope this helps > > Cal > > > > On Thu, Oct 10, 2013 at 2:18 AM, Nandy Dagondon wrote: > >> Hello everyone, >> >> I like to play an announcement file before playing music-on-hold. Is this >> possible? Any hint? >> >> Tks, >> Nandy >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/1238e023/attachment.html From rafal.gwizdala at gmail.com Thu Oct 10 09:11:05 2013 From: rafal.gwizdala at gmail.com (Rafal Gwizdala) Date: Thu, 10 Oct 2013 07:11:05 +0200 Subject: [Freeswitch-users] Weekly News Notes Etc In-Reply-To: References: Message-ID: Thanks, R On Wed, Oct 9, 2013 at 8:06 PM, Nick Vines wrote: > Most of them are here http://torrents.freeswitch.org/ > > > > > On Wed, Oct 9, 2013 at 10:43 AM, Rafal Gwizdala wrote: > >> Guys, >> Does anyone record these conferences? I'm trying to join and listen but >> the conf starts at 7PM for me (in Poland) and you know, it's quite >> difficult to concentrate and listen with a family around. It would be >> better to listen to a recorded presentation later... >> Best regards >> RG >> >> >> >> >> >> >> On Wed, Oct 9, 2013 at 4:58 PM, Ken Rice wrote: >> >>> Hey Guys! Don?t forget to join us today for ClueCon weekly on 888 at >>> 1P Eastern or 10A Pacific! >>> >>> If you haven?t joined the Groups on Google+, Follow FreeSWITCH here >>> https://plus.google.com/communities/110013892794022037793 >>> Follow ClueCon here >>> https://plus.google.com/communities/108955134544785139154 >>> >>> And don?t forget to follow us on Twitter @FreeSWITCH_wire >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> @Ken4VoIP via Twitter >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/99188bed/attachment.html From miha at softnet.si Thu Oct 10 10:00:08 2013 From: miha at softnet.si (Miha) Date: Thu, 10 Oct 2013 08:00:08 +0200 Subject: [Freeswitch-users] Diversion header In-Reply-To: <52EA0C6C-84A2-4812-ADAE-8F70D1358A14@freeswitch.org> References: <5252A838.3050408@softnet.si> <5253A6B6.7030307@softnet.si> <52EA0C6C-84A2-4812-ADAE-8F70D1358A14@freeswitch.org> Message-ID: <52564268.2080406@softnet.si> Brian tnx. I was trying to export it but probem is that I can not find it in varibales that are loged with action application "info". br, miha Dne 10/8/2013 5:39 PM, pis(e Brian West: > You have to remember this one very key fact... we aren't a proxy, if you want the diversion header on the b-leg you MUST export it/set it on the b-leg. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > On Oct 8, 2013, at 1:31 AM, Miha wrote: > >> Hi, >> >> I am sending invite throught FS with diversion header. Is it possible to >> set that FS will not remove diversion header and it will send it further? >> >> tnx! >> >> br, >> miha >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/194b63e9/attachment.html From vishal.kakkar at gmail.com Thu Oct 10 10:07:27 2013 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Thu, 10 Oct 2013 11:37:27 +0530 Subject: [Freeswitch-users] Channels Getting Stuck Message-ID: I have setup with 2 PRIs(terminating on sangoma cards) connected to FS using wanpipe. During campaign my channels are getting stuck.. show cahnnels shows 52 channels but actual calls are only few.. Here is output of show channels- uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num *18a7fa23-3f47-4a31-8c0d-c37e6b12b668,outbound,2013-10-10 11:16:49,1381384009,FreeTDM/1:9/09830701954,CS_CONSUME_MEDIA,,01203896280,,09830701954,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09830701954,,18a7fa23-3f47-4a31-8c0d-c37e6b12b668,,* 44de04f1-8b22-44d5-9665-c5771d1d66b4,outbound,2013-10-10 11:21:41,1381384301,FreeTDM/1:3/09327241861,CS_EXECUTE,Outbound Call,09327241861,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,44de04f1-8b22-44d5-9665-c5771d1d66b4,, e564edcf-d590-4c45-ab0e-3f19e08a872c,outbound,2013-10-10 11:22:02,1381384322,FreeTDM/1:7/07830354545,CS_EXECUTE,Outbound Call,07830354545,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,e564edcf-d590-4c45-ab0e-3f19e08a872c,, b5a83d5a-b705-41ad-8abb-d9c319d2e594,outbound,2013-10-10 11:22:32,1381384352,FreeTDM/1:18/07566326690,CS_EXECUTE,Outbound Call,07566326690,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,b5a83d5a-b705-41ad-8abb-d9c319d2e594,, 53df0a17-a450-44b8-ab70-915f8f662b47,outbound,2013-10-10 11:22:45,1381384365,FreeTDM/1:16/09937432216,CS_EXECUTE,Outbound Call,09937432216,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,53df0a17-a450-44b8-ab70-915f8f662b47,, 510a75eb-2340-4a50-9284-6247b52f2946,outbound,2013-10-10 11:22:46,1381384366,FreeTDM/1:22/09438189454,CS_EXECUTE,Outbound Call,09438189454,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,510a75eb-2340-4a50-9284-6247b52f2946,, ae801140-b8eb-4824-b3bf-49cefae8eef0,outbound,2013-10-10 11:22:52,1381384372,FreeTDM/1:19/09767443205,CS_EXECUTE,Outbound Call,09767443205,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,ae801140-b8eb-4824-b3bf-49cefae8eef0,, 0fd82b66-34cb-4c72-8a0f-3e7781c4b4f3,outbound,2013-10-10 11:23:02,1381384382,FreeTDM/1:2/09777825850,CS_EXECUTE,Outbound Call,09777825850,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,0fd82b66-34cb-4c72-8a0f-3e7781c4b4f3,, 3ba5d15b-5233-441c-9954-a8668a4f4a39,outbound,2013-10-10 11:23:02,1381384382,FreeTDM/1:12/09765341919,CS_EXECUTE,Outbound Call,09765341919,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,3ba5d15b-5233-441c-9954-a8668a4f4a39,, bb4be954-f5a4-49ad-a02f-4d9903466d69,outbound,2013-10-10 11:23:18,1381384398,FreeTDM/1:3/09329366443,CS_CONSUME_MEDIA,,01203896280,,09329366443,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09329366443,,bb4be954-f5a4-49ad-a02f-4d9903466d69,, d6d66ff1-13cb-4c22-a66c-246baaed03c7,outbound,2013-10-10 11:23:18,1381384398,FreeTDM/1:8/09690699408,CS_CONSUME_MEDIA,,01203896280,,09690699408,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09690699408,,d6d66ff1-13cb-4c22-a66c-246baaed03c7,, *a9adc429-6f37-4f8f-a395-c6fab5d15ca2,outbound,2013-10-10 11:23:38,1381384418,FreeTDM/1:9/09623862679,CS_EXECUTE,Outbound Call,09623862679,,obdExtension,hangup,,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,a9adc429-6f37-4f8f-a395-c6fab5d15ca2,,* 0f12198e-87fc-4836-9255-47a03644a151,outbound,2013-10-10 11:23:52,1381384432,FreeTDM/1:6/09812005112,CS_EXECUTE,Outbound Call,09812005112,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,0f12198e-87fc-4836-9255-47a03644a151,, 1660165c-51b3-450a-a4a4-23f616dc04ab,outbound,2013-10-10 11:24:03,1381384443,FreeTDM/1:3/09754094192,CS_EXECUTE,Outbound Call,09754094192,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,1660165c-51b3-450a-a4a4-23f616dc04ab,, 9e773534-1d16-4ff6-8760-f743c02a3973,outbound,2013-10-10 11:24:08,1381384448,FreeTDM/1:4/09420423005,CS_EXECUTE,Outbound Call,09420423005,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,9e773534-1d16-4ff6-8760-f743c02a3973,, 5fc752f4-e34a-462d-b21d-80da1af55b73,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:6/09439082697,CS_CONSUME_MEDIA,,01203896280,,09439082697,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09439082697,,5fc752f4-e34a-462d-b21d-80da1af55b73,, 4aaf9bc3-da10-422e-aa3d-3120a9751671,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:11/09300153456,CS_EXECUTE,Outbound Call,09300153456,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,4aaf9bc3-da10-422e-aa3d-3120a9751671,, d96829c1-cba2-43ad-8ff3-6a6976bcf78d,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:13/09439082697,CS_EXECUTE,Outbound Call,09439082697,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,d96829c1-cba2-43ad-8ff3-6a6976bcf78d,, 46d90236-18df-4f3d-b5ae-11aac8e9baf3,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:14/09977252323,CS_EXECUTE,Outbound Call,09977252323,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,46d90236-18df-4f3d-b5ae-11aac8e9baf3,, 95cea2f1-4c3b-4a15-9961-5706855b427b,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:17/09420423005,CS_CONSUME_MEDIA,,01203896280,,09420423005,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09420423005,,95cea2f1-4c3b-4a15-9961-5706855b427b,, 2f87fdb2-ea5c-4157-a0c8-d1746cf8ad55,outbound,2013-10-10 11:24:30,1381384470,FreeTDM/1:18/09810666158,CS_CONSUME_MEDIA,,01203896280,,09810666158,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09810666158,,2f87fdb2-ea5c-4157-a0c8-d1746cf8ad55,, c710f549-ca34-40ea-9a75-e0b675ea1b5c,outbound,2013-10-10 11:24:30,1381384470,FreeTDM/1:19/09810666158,CS_EXECUTE,Outbound Call,09810666158,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,c710f549-ca34-40ea-9a75-e0b675ea1b5c,, bde4aea8-e9d3-4209-b186-63fe3cacf67d,outbound,2013-10-10 11:24:46,1381384486,FreeTDM/1:12/09766434011,CS_EXECUTE,Outbound Call,09766434011,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,bde4aea8-e9d3-4209-b186-63fe3cacf67d,, *c9c88437-64e1-49a4-a180-985777f265d4,outbound,2013-10-10 11:24:51,1381384491,FreeTDM/1:9/07587465873,CS_EXECUTE,Outbound Call,07587465873,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,c9c88437-64e1-49a4-a180-985777f265d4,,* 5f89a635-ba5c-42c1-a98f-573eda504372,outbound,2013-10-10 11:24:56,1381384496,FreeTDM/1:5/09921748436,CS_EXECUTE,Outbound Call,09921748436,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,5f89a635-ba5c-42c1-a98f-573eda504372,, 0cb88568-46e7-4d4c-8180-e5cf489947ba,outbound,2013-10-10 11:24:56,1381384496,FreeTDM/1:7/09920817954,CS_CONSUME_MEDIA,,01203896280,,09920817954,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09920817954,,0cb88568-46e7-4d4c-8180-e5cf489947ba,, 56de78a3-62b7-4710-ae96-43f35c00335c,outbound,2013-10-10 11:25:06,1381384506,FreeTDM/1:15/09617761948,CS_EXECUTE,Outbound Call,09617761948,,obdExtension,hangup,,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,56de78a3-62b7-4710-ae96-43f35c00335c,, 8bd2bddc-72c2-460f-a731-158f86abe268,outbound,2013-10-10 11:25:18,1381384518,FreeTDM/1:4/09890804877,CS_CONSUME_MEDIA,,01203896280,,09890804877,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09890804877,,8bd2bddc-72c2-460f-a731-158f86abe268,, 7b016887-890a-4c66-aebe-62895b6f79bd,outbound,2013-10-10 11:25:22,1381384522,FreeTDM/1:2/09536983588,CS_EXECUTE,Outbound Call,09536983588,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,7b016887-890a-4c66-aebe-62895b6f79bd,, 5fc9f447-2f2b-4e81-9d36-f76c9c3ffb53,outbound,2013-10-10 11:25:22,1381384522,FreeTDM/1:6/09763675941,CS_EXECUTE,Outbound Call,09763675941,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,5fc9f447-2f2b-4e81-9d36-f76c9c3ffb53,, 86983e1e-7632-47eb-a5aa-e88aff98feae,outbound,2013-10-10 11:25:22,1381384522,FreeTDM/1:8/09823961213,CS_EXECUTE,Outbound Call,09823961213,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,86983e1e-7632-47eb-a5aa-e88aff98feae,, f838d469-bc5b-4485-9960-5760d1aa92df,outbound,2013-10-10 11:25:31,1381384531,FreeTDM/1:1/09831622477,CS_EXECUTE,Outbound Call,09831622477,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,f838d469-bc5b-4485-9960-5760d1aa92df,, 1509d219-d910-4aa1-80db-15ccaad98d27,outbound,2013-10-10 11:25:32,1381384532,FreeTDM/1:10/09827184894,CS_EXECUTE,Outbound Call,09827184894,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,1509d219-d910-4aa1-80db-15ccaad98d27,, 95ea1b21-146b-4e5c-a39d-a11447429eda,outbound,2013-10-10 11:25:38,1381384538,FreeTDM/1:11/09301583677,CS_EXECUTE,Outbound Call,09301583677,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,95ea1b21-146b-4e5c-a39d-a11447429eda,, f189518b-4f59-4660-840d-4ad87e243ec5,outbound,2013-10-10 11:25:48,1381384548,FreeTDM/1:3/09423175934,CS_CONSUME_MEDIA,,01203896280,,09423175934,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09423175934,,f189518b-4f59-4660-840d-4ad87e243ec5,, 76103c25-1509-4469-8061-181e2d9c3179,outbound,2013-10-10 11:25:48,1381384548,FreeTDM/1:13/09423175934,CS_EXECUTE,Outbound Call,09423175934,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,76103c25-1509-4469-8061-181e2d9c3179,, fae8ffb9-03a4-436b-b194-c5bed8dd4f55,outbound,2013-10-10 11:25:51,1381384551,FreeTDM/1:4/09891012004,CS_EXECUTE,Outbound Call,09891012004,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,fae8ffb9-03a4-436b-b194-c5bed8dd4f55,, d43115c8-88a4-4853-a2d7-6a2355f250e9,outbound,2013-10-10 11:25:52,1381384552,FreeTDM/1:14/09433214771,CS_EXECUTE,Outbound Call,09433214771,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,d43115c8-88a4-4853-a2d7-6a2355f250e9,, ad016f35-3186-40cb-b1db-1d0b961f9945,outbound,2013-10-10 11:25:52,1381384552,FreeTDM/1:16/09883025715,CS_EXECUTE,Outbound Call,09883025715,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,ad016f35-3186-40cb-b1db-1d0b961f9945,, 4d8fca62-d5a4-4cdb-b70d-7193a2ff2307,outbound,2013-10-10 11:26:07,1381384567,FreeTDM/1:7/09583755624,CS_EXECUTE,Outbound Call,09583755624,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,4d8fca62-d5a4-4cdb-b70d-7193a2ff2307,, *931fb548-3ed0-4085-9c65-2196b76999a4,outbound,2013-10-10 11:26:09,1381384569,FreeTDM/1:9/09001192140,CS_EXECUTE,Outbound Call,09001192140,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,*,,ACTIVE,,,RECV,931fb548-3ed0-4085-9c65-2196b76999a4,, c1d192aa-9589-422d-b2b6-385a837ad8fc,outbound,2013-10-10 11:26:09,1381384569,FreeTDM/1:10/09583755624,CS_CONSUME_MEDIA,,01203896280,,09583755624,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09583755624,,c1d192aa-9589-422d-b2b6-385a837ad8fc,, 31955dd1-d701-4cb0-b76b-cc33581e1ae2,outbound,2013-10-10 11:26:09,1381384569,FreeTDM/1:12/09223317664,CS_EXECUTE,Outbound Call,09223317664,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,ACTIVE,,,RECV,31955dd1-d701-4cb0-b76b-cc33581e1ae2,, 87570d9f-621c-4ee9-ba2c-4cacb5375adf,outbound,2013-10-10 11:26:10,1381384570,FreeTDM/1:17/09583755624,CS_CONSUME_MEDIA,,01203896280,,09583755624,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09583755624,,87570d9f-621c-4ee9-ba2c-4cacb5375adf,, 2d254646-62bf-4505-870e-e7c5eabd7832,outbound,2013-10-10 11:26:37,1381384597,FreeTDM/1:1/09469549674,CS_CONSUME_MEDIA,,01203896280,,09469549674,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09469549674,,2d254646-62bf-4505-870e-e7c5eabd7832,, fcbcf8ad-2660-4b6a-8310-b837305b72db,outbound,2013-10-10 11:26:38,1381384598,FreeTDM/1:3/09453287889,CS_CONSUME_MEDIA,,01203896280,,09453287889,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09453287889,,fcbcf8ad-2660-4b6a-8310-b837305b72db,, 56a96b36-e840-4af9-9aef-bf9d44ba93eb,outbound,2013-10-10 11:26:40,1381384600,FreeTDM/1:5/09223317664,CS_CONSUME_MEDIA,,01203896280,,09223317664,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,09223317664,,56a96b36-e840-4af9-9aef-bf9d44ba93eb,, 921446b6-944b-4e5f-999a-69091cf6c5ca,outbound,2013-10-10 11:26:43,1381384603,FreeTDM/1:10/08018158992,CS_CONSUME_MEDIA,,01203896280,,08018158992,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,08018158992,,921446b6-944b-4e5f-999a-69091cf6c5ca,, 71c7a9ea-c897-4722-bb66-7a8d92801bef,outbound,2013-10-10 11:26:43,1381384603,FreeTDM/1:15/08018158992,CS_CONSUME_MEDIA,,01203896280,,08018158992,,,,default,PCMA,8000,64000,PCMA,8000,64000,, xx.yyy.com,,,EARLY,Outbound Call,08018158992,,71c7a9ea-c897-4722-bb66-7a8d92801bef,, 49 channels Same channel is apearing multiple times as i have bolded *channel 1:9 in above* output. Please help what should i do to overcome it.. My 3 hour campaign is now taking 18 hours becasue of this issue. Thanks, -Vishal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/a70dab96/attachment-0001.html From anton.vazir at gmail.com Thu Oct 10 10:09:01 2013 From: anton.vazir at gmail.com (Anton VG) Date: Thu, 10 Oct 2013 10:09:01 +0400 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: References: Message-ID: I clean such calls with external script, using uuid_kill - but yes, the calls are there. 2013/10/10 Peter Hartmann > Hi, > First, thanks for Freeswitch! I'm experiencing an issue where 'show > calls' returns several calls that aren't actually happening both > inbound and outbound. Has anyone seen this before? > > Rebooting the handset (Polycom IP 550) associated with that extension > has no effect so it seems in FS. > > freeswitch at internal> show calls > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch > d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 > 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com > ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 > 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 > 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com > ,CS_EXECUTE,unknown > ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 > 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 > 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension > 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound > Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension > 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, > bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 > 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 > 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 > 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com > ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 > 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com > ,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > > 9 total. > > > Running: > FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 > 2013-10-02 21:30:46Z) > > > Thanks much! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/adf0e5f4/attachment.html From karl at xtronics.com Thu Oct 10 10:29:58 2013 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 10 Oct 2013 01:29:58 -0500 Subject: [Freeswitch-users] Garbled audio solved on polycom phone Message-ID: <52564966.3040101@xtronics.com> Play back of audio with the 'playback' command produced garbled audio. First, I found a message complaining that the sample rate was 8K instead of 16K - re-sampling brought no joy. (This begs the questions - what is the best audio format for playback of messages? File type - sample rate? etc - If I record via the 'record' application, the file is : Sample Rate : 8000 Precision : 16-bit Bit Rate : 128k Sample Encoding: 16-bit Signed Integer PCM But is seems to expect 16000 sample rates to play back? What is the internal audio traffic in freeswitch based on? ) Then I noticed that the garbling occurred on a polycom IP560 phone as the zrtp utils messages were getting dumped in the log. Doing a global_setvar zrtp_secure_media=false Fixes the problem. Seems like this should fail more gracefully? I have the latest Debian - 1.2.12~2-1~wheezy+1 ( Placing in the directory entry also fixes the problem ) EXECUTE sofia/internal/11 at hostname playback(mysounds/answer.wav) 2013-10-10 00:53:44.989857 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16 at 16000hz 1 channels 20ms 2013-10-10 00:53:45.010017 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/11 at hostname [BREAK] 2013-10-10 00:53:45.010017 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/11 at hostname [BREAK] 2013-10-10 00:53:45.010017 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/11 at hostname [BREAK] 2013-10-10 00:53:45.029906 [DEBUG] sofia.c:5720 Channel sofia/internal/11 at hostname entering state [ready][200] 2013-10-10 00:53:45.049900 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36849 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:45.129740 [DEBUG] switch_rtp.c:3706 Correct ip/port confirmed. 2013-10-10 00:53:45.149778 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36850 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:45.349846 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36851 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:45.549840 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36852 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:45.749841 [DEBUG] switch_rtp.c:928 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=39 2013-10-10 00:53:45.749841 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36853 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:45.949840 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36854 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:46.149819 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36855 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:46.349793 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36856 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:46.549841 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36857 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:46.749792 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36858 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:46.969846 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36859 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:47.169786 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36860 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:47.369841 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36861 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:47.569842 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36862 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:47.769850 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36863 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:47.969844 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36864 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:48.169838 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36865 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:48.369839 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36866 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:48.569849 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=307819312 seq=36867 size=144. Stream 39:CLEAR:START 2013-10-10 00:53:48.769843 [DEBUG] switch_rtp.c:928 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=39 2013-10-10 00:53:48.769843 [DEBUG] switch_rtp.c:928 [ zrtp]: Stream ID=39 CLEAR switching ---> . 2013-10-10 00:53:51.569844 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/11 at hostname [BREAK] -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Borrow money from pessimists - they don't expect it back. -------------------------------------------------------------------------------- From sirimmfs at gmail.com Thu Oct 10 10:35:29 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 10 Oct 2013 17:35:29 +1100 Subject: [Freeswitch-users] Root tag missing Message-ID: Hi, I am looking at a system from a customer having an old version of Freeswitch. For some reason, freeswitch doesn't seem to start - logs are something like: [ERR] switch_xml.c:1467 Short write! [ERR] switch_xml.c:1467 Short write! [ERR] switch_xml.c:1467 Short write! [ERR] switch_xml.c:1467 Short write! . . . Cannot Initialize [[error near line 1]: root tag missing! log/freeswitch.xml.fsxml is empty, and customer is not using xml_curl. I ran xmllint, but couldn't really find any obvious problems. freeswitch.xml and vars.xml seem fine.. anything else that I could check? May be some xml is missing, but I am not sure which one! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/546c3879/attachment.html From noc at sonerep.com Thu Oct 10 12:16:45 2013 From: noc at sonerep.com (Labolinux) Date: Thu, 10 Oct 2013 10:16:45 +0200 Subject: [Freeswitch-users] [Freeswitch-video calls] Message-ID: <5256626D.3090206@sonerep.com> Hello, I have installed Freeswitch with kamailio on Debian 7.0.0 wheezy. Calls, instant messaginng, video calls on Kamailio work well. Confrence calls on freeswitch work well. Now I want to know if it is possible to configure video conference. There is a module for this? How can resolve this problem? thanks From matt at inveroak.com Thu Oct 10 15:06:31 2013 From: matt at inveroak.com (Matt Broad) Date: Thu, 10 Oct 2013 12:06:31 +0100 Subject: [Freeswitch-users] originate not creating xml_cdr record In-Reply-To: <5231D135.3000207@inveroak.com> References: <52307F11.9020704@inveroak.com> <5230B5EB.9050909@inveroak.com> <5230BCA0.9090102@inveroak.com> <5231C8A1.7030407@inveroak.com> <5231D135.3000207@inveroak.com> Message-ID: <52568A37.9000502@inveroak.com> Hi just as an update to this thread, sorry for the delay :) I have resolved the issue. Freeswitch was behaving impeccably, my XML parsing code on the other hand was not :) Error handling when a node is not present is always a good thing to have! thanks Matt On 12/09/2013 15:35, Matt Broad wrote: > Hi Cal, > > have just seen the reply from Lloyd (my email's threading isn't too > good), yes the bleg logging is set to true, and is working fine on > calls instigated via an incoming call. > > I'll take a look at the bug reporting pages and try and give as much > info as possible. > > > > thanks > Matt > > > On 12/09/2013 15:24, Cal Leeming [Simplicity Media Ltd] wrote: >> Just to be sure, can you double check that b_leg is set to true in >> your xml_cdr.conf (as per Lloyds' suggestion). >> >> If it is set to true, then next step would be to raise a bug report >> in JIRA with as much info as possible about the steps you have taken >> to reproduce this bug. >> >> If you are unfamiliar with this process, please see; >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Where_And_How_To_File_Bug_Reports >> >> Sadly I won't have any time to look into this until late next week, >> but perhaps another volunteer will jump on and take a look before I do. >> >> Cal >> >> >> >> On Thu, Sep 12, 2013 at 2:58 PM, Matt Broad > > wrote: >> >> Hi Cal, >> >> thanks for the suggestions. >> I have tested adding an err log dir and also enabled the >> log-http-and-disk option. >> >> These work fine when I make a normal call (I have forced a http >> error on the webserver to test the error logging), but when doing >> an originate from the CLI, there are no logs at all. >> >> I don't see any messages about a failed log ("mod_xml_cdr.c:365 >> Got error [0] posting to web server http://******" & >> "mod_xml_cdr.c:372 Retry will be with url http://****") when the >> call ends. >> >> thanks >> Matt >> >> >> >> On 11/09/2013 20:29, Cal Leeming [Simplicity Media Ltd] wrote: >>> Oh damn, sorry I was thinking of mod_xml_curl! >>> >>> In that case, try this instead; >>> http://wiki.freeswitch.org/wiki/Mod_xml_cdr >>> >>> >>> Enable that param, and check to see if the request ends up in >>> that directory. >>> >>> You can also enable 'log-http-and-disk' which will ensure the >>> request is always written to disk. >>> >>> You can also enable access logs on your web server to check if >>> the request is received, or even just run "nc -l 80" on the web >>> server and wait for the connection to be received. >>> >>> The main point of the above is to pinpoint where the fault is >>> happening (if the entry is pushed to disk, then it means the >>> problem is with your web server.. if the entry is not pushed to >>> disk, then there's a problem somewhere in FS, whether it be in >>> config or a bug). >>> >>> Sorry for the confusion >>> >>> Cal >>> >>> >>> >>> On Wed, Sep 11, 2013 at 7:55 PM, Matt Broad >> > wrote: >>> >>> Hi Cal, >>> >>> I have tried the method that you described but I get the >>> error "-ERR xml_cdr debug_on Command not found!" >>> could this be a typo? I have tried googling but with no >>> success. >>> >>> thanks >>> Matt >>> >>> >>> On 11/09/2013 19:26, Matt Broad wrote: >>>> Thanks for the quick reply Cal. >>>> >>>> I will give that a try and see what I can see. I will >>>> report back with my findings :) >>>> >>>> thanks >>>> Matt >>>> >>>> >>>> On 11/09/2013 15:55, Cal Leeming [Simplicity Media Ltd] wrote: >>>>> Matt, >>>>> >>>>> As far as I know, the API originate method should also >>>>> trigger xml_cdr to post back to your server. >>>>> >>>>> Can you please run "console loglevel debug" and "xml_cdr >>>>> debug_on", attempt to use api originate, then see if >>>>> anything shows up in console? (that command will enable >>>>> xml_cdr debugging). >>>>> >>>>> If this still does not work, please capture all the logs >>>>> and explain the test procedure you went through and send >>>>> it all in a JIRA ticket. >>>>> >>>>> Hope this helps >>>>> >>>>> Cal >>>>> >>>>> >>>>> On Wed, Sep 11, 2013 at 3:32 PM, Matt Broad >>>>> > wrote: >>>>> >>>>> Hi, >>>>> >>>>> I have setup xml_cdr and it is working great, both >>>>> aleg and bleg calls >>>>> are logged. >>>>> I am using the api originate method, but this does not >>>>> create an xml_cdr >>>>> record (though a record is logged in >>>>> /log/cdr-csv/master/csv). Is this >>>>> correct? >>>>> >>>>> >>>>> >>>>> thanks >>>>> Matt >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>> Server >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/05367793/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Thu Oct 10 15:22:53 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 10 Oct 2013 13:22:53 +0200 Subject: [Freeswitch-users] Garbled audio solved on polycom phone In-Reply-To: <52564966.3040101@xtronics.com> References: <52564966.3040101@xtronics.com> Message-ID: <52568E0D.7040302@puzzled.xs4all.nl> On 10/10/2013 08:29 AM, Karl Schmidt wrote: [snip] > Then I noticed that the garbling occurred on a polycom IP560 phone as the zrtp utils messages were > getting dumped in the log. > > Doing a > > global_setvar zrtp_secure_media=false > > Fixes the problem. Seems like this should fail more gracefully? > > I have the latest Debian - 1.2.12~2-1~wheezy+1 > > ( Placing in the directory entry also > fixes the problem ) If you think it's a bug then please do file a bug in Jira. Issues reported on the mailing list tend to get lost in the noise. So if you want something fixed, Jira is the only way. Regards, Patrick From steveayre at gmail.com Thu Oct 10 16:15:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 10 Oct 2013 13:15:06 +0100 Subject: [Freeswitch-users] Dialplan 'set' variable not found in CDR In-Reply-To: <1381325911740-7595574.post@n2.nabble.com> References: <1381325911740-7595574.post@n2.nabble.com> Message-ID: Is the CDR for the aleg or bleg of the bridge? They'll only be set on the aleg (incoming) leg. On 9 October 2013 14:38, adahary wrote: > Hi, > > I'm trying to pass some custom variables from dialplan to cdr with no > success. > here is an example of my dialplan with the 'set' of some custom variable > 'x_dialplan_mode'. > > why I cannot see this variable 'x_dialplan_mode' in the CDR XML ? > > > >
description='CALL Routing request'> > > > field='destination_number' expression='^sony27i'> > data='call_timeout=45'/> > > > > > > > > > > > >
> > > regards > Assaf > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Dialplan-set-variable-not-found-in-CDR-tp7595574.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/2bcfcd72/attachment.html From mike at jerris.com Thu Oct 10 16:42:10 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Oct 2013 08:42:10 -0400 Subject: [Freeswitch-users] Root tag missing In-Reply-To: References: Message-ID: Is the disk out of space or no permissions to create a file in the log directory? On Oct 10, 2013, at 2:35 AM, Siri MM wrote: > Hi, > > I am looking at a system from a customer having an old version of Freeswitch. For some reason, freeswitch doesn't seem to start - logs are something like: > > [ERR] switch_xml.c:1467 Short write! > [ERR] switch_xml.c:1467 Short write! > [ERR] switch_xml.c:1467 Short write! > [ERR] switch_xml.c:1467 Short write! > . > . > . > Cannot Initialize [[error near line 1]: root tag missing! > > log/freeswitch.xml.fsxml is empty, and customer is not using xml_curl. > > I ran xmllint, but couldn't really find any obvious problems. freeswitch.xml and vars.xml seem fine.. anything else that I could check? May be some xml is missing, but I am not sure which one! From acrow at integrafin.co.uk Thu Oct 10 16:45:55 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 10 Oct 2013 13:45:55 +0100 Subject: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions In-Reply-To: References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch> <9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch> <52541FB2.9090602@integrafin.co.uk> <5255A520.8040909@integrafin.co.uk> Message-ID: <5256A183.5020806@integrafin.co.uk> Oops - sorry - had the kids around when I was typing that! Cheers Alex ----Original Message---- *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions *Date:* Thu, 10 Oct 2013 07:37:21 +0300 *From:* Yehavi Bourvine *To:* FreeSWITCH Users Help > It is FS-5864... > __Yehavi: > > > 2013/10/9 Alex Crow > > > Hi, > > JIRA issue posted, FS-5684. Can the original reporters please > update/add details? > > Thanks > > > Alex > > ----Original Message---- > *Subject:* Re: [Freeswitch-users] Possible Polycom bridging issues > in all recent > FreeSWITCH versions > *Date:* Tue, 08 Oct 2013 16:07:30 +0100 > *From:* Alex Crow > > *To:* freeswitch-users at lists.freeswitch.org > > > > >> Hi, >> >> I can confirm that the issue affects me in version: >> >> FreeSWITCH Version >> 1.2.11-n20130817T172515Z-1~wheezy+1+git~20130816T225403Z~8566ffa82a >> (-n20130817T172515Z-1~wheezy+1git 8566ffa 2013-08-16 22:54:03Z) >> >> but not in: >> >> FreeSWITCH Version 1.2.8+git~20130405T052920Z~57fb368b32 (git >> 57fb368 2013-04-05 05:29:20Z) >> >> if that helps anyone. >> >> These are production systems though so I've had to disable the >> relevant codecs. >> >> Cheers >> >> Alex >> >> ----Original Message---- >> *Subject:* Re: [Freeswitch-users] Possible Polycom bridging >> issues in >> all recent FreeSWITCH versions >> *Date:* Sun, 4 Aug 2013 02:43:27 +0000 >> *From:* Moishe Grunstein >> >> *To:* FreeSWITCH Users Help >> >> >> >> >> >>> I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related. >>> >>> >>> Thanks, >>> >>> Moishe Grunstein >>> Tornado Computer Systems, Inc. >>> 212.400.7650888.IPPBX.US >>> Service Request Email:support at nysolutions.com >>> Polycom Certified VAR >>> Microsoft Small Business Specialist, Cisco SMB Select Certified >>> >>> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS >>> >>> -----Original Message----- >>> From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah >>> Sent: Saturday, August 03, 2013 10:34 PM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions >>> >>> I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. >>> Thanks and I'll make sure I get it through the process. >>> >>> Cheers >>> On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson wrote: >>> >>>> On Sat, Aug 3, 2013 at 3:31 PM, Emrah wrote: >>>>> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy. >>>>> >>>>> The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end? >>>> This is what you need to know: >>>> https://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> >>>> Best, >>>> Gabe >>>> >>>> ______________________________________________________________________ >>>> ___ Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>>> rs >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> This message is intended only for the addressee and may contain >> confidential information. Unless you are that person, you may not >> disclose its contents or use it in any way and are requested to delete >> the message along with any attachments and notify us immediately. >> "Transact" is operated by Integrated Financial Arrangements plc. 29 >> Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 >> 5300. (Registered office: as above; Registered in England and Wales >> under number: 3727592). Authorised and regulated by the Financial >> Conduct Authority (entered on the Financial Services Register; no. 190856). > > > -- > > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > "Transact" is operated by Integrated Financial Arrangements plc. 29 > Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 > 5300. (Registered office: as above; Registered in England and Wales > under number: 3727592). Authorised and regulated by the Financial > Conduct Authority (entered on the Financial Services Register; no. 190856). > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc. 29 Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 5300. (Registered office: as above; Registered in England and Wales under number: 3727592). Authorised and regulated by the Financial Conduct Authority (entered on the Financial Services Register; no. 190856). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/ecb12c68/attachment-0001.html From vladget at gmail.com Thu Oct 10 17:35:23 2013 From: vladget at gmail.com (Vladimir Getmanshchuk) Date: Thu, 10 Oct 2013 16:35:23 +0300 Subject: [Freeswitch-users] Dialplan 'set' variable not found in CDR In-Reply-To: <1381325911740-7595574.post@n2.nabble.com> References: <1381325911740-7595574.post@n2.nabble.com> Message-ID: Hi There are at least 3 ways: Collect CDR for both (a and b) legs; Export variable from a-leg to be leg; Or set you variable before bridge application, use {variable=value}/sofia/gw/... for global, and [variable=value]/sofia/gw/... for b-leg only. On Wed, Oct 9, 2013 at 4:38 PM, adahary wrote: > Hi, > > I'm trying to pass some custom variables from dialplan to cdr with no > success. > here is an example of my dialplan with the 'set' of some custom variable > 'x_dialplan_mode'. > > why I cannot see this variable 'x_dialplan_mode' in the CDR XML ? > > > >
description='CALL Routing request'> > > > field='destination_number' expression='^sony27i'> > data='call_timeout=45'/> > > > > > > > > > > > >
> > > regards > Assaf > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Dialplan-set-variable-not-found-in-CDR-tp7595574.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Yours sincerely, Vladimir Getmanshchuk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/06bb2f45/attachment.html From adahary at gmail.com Thu Oct 10 18:17:31 2013 From: adahary at gmail.com (adahary) Date: Thu, 10 Oct 2013 07:17:31 -0700 (PDT) Subject: [Freeswitch-users] Dialplan 'set' variable not found in CDR In-Reply-To: <1381325911740-7595574.post@n2.nabble.com> References: <1381325911740-7595574.post@n2.nabble.com> Message-ID: <1381414651596-7595599.post@n2.nabble.com> It works now - I can see all custom variables in CDR. thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialplan-set-variable-not-found-in-CDR-tp7595574p7595599.html Sent from the freeswitch-users mailing list archive at Nabble.com. From pranav.lal at gmail.com Thu Oct 10 20:10:41 2013 From: pranav.lal at gmail.com (Pranav Lal) Date: Thu, 10 Oct 2013 21:40:41 +0530 Subject: [Freeswitch-users] Creating a rss reading solution using mod rss and espeak? Message-ID: <000001cec5d3$4738f710$d5aae530$@gmail.com> Hi all, Warning, this is not a telephony question per se but yes, it does involve the use of freeswitch. I have an octogenarian grandmother. She has a hard time entertaining herself so I want to build a newspaper reading solution using freeswitch. The advantage of using freeswitch is that she is familiar with a phone so can dial in and work the phone keypad. I plan to give her a tablet with a voice over IP application so that she can hit the phone keys and use it independently. Yes, I know this is iffy but it is the best I can do right now. Most of our local newspapers do have rss feeds so I am planning to use mod rss. The wiki suggests I use mod_cepstral. My grandmother finds it easier to communicate in Hindi. I did not see a cepstral voice for Hindi hence I want to use espeak. My questions are as follows. 1. Can I use espeak with mod rss? If so what are the steps for doing so? 2. I see the java script at http://wiki.freeswitch.org/wiki/Mod_rss which pulls the rss feed. Does anyone have an example where multiple feeds are rendered through freeswitch? Note: I already have freeswitch compiled and running on my raspberry pi. Suggestions are welcome. Pranav From vishal.kakkar at gmail.com Thu Oct 10 20:12:58 2013 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Thu, 10 Oct 2013 21:42:58 +0530 Subject: [Freeswitch-users] Error While upgrading from 1.05 to latest Git head Message-ID: Hi All, Today i tried to upgrade FS from (Version 1.5.2b git a433b97 2013-06-07 17:29:42Z) to latest one using make current. Following is the error i am getting- /bin/sh ../libtool --tag=CC --mode=link gcc -g -O2 -fvisibility=hidden -o libspeexdsp.la -rpath /usr/local/freeswitch/lib -no-undefined -version-info 6:0:5 preprocess.lo jitter.lo mdf.lo fftwrap.lo filterbank.lo resample.lo buffer.lo scal.lo smallft.lo -lm ar cru .libs/libspeexdsp.a preprocess.o jitter.o mdf.o fftwrap.o filterbank.o resample.o buffer.o scal.o smallft.o ranlib .libs/libspeexdsp.a creating libspeexdsp.la (cd .libs && rm -f libspeexdsp.la && ln -s ../libspeexdsp.la libspeexdsp.la) make[4]: Leaving directory `/home/installers/freeswitch/libs/speex/libspeex' Making all in include make[4]: Entering directory `/home/installers/freeswitch/libs/speex/include' Making all in speex make[5]: Entering directory `/home/installers/freeswitch/libs/speex/include/speex' make[5]: Nothing to be done for `all'. make[5]: Leaving directory `/home/installers/freeswitch/libs/speex/include/speex' make[5]: Entering directory `/home/installers/freeswitch/libs/speex/include' make[5]: Nothing to be done for `all-am'. make[5]: Leaving directory `/home/installers/freeswitch/libs/speex/include' make[4]: Leaving directory `/home/installers/freeswitch/libs/speex/include' make[4]: Entering directory `/home/installers/freeswitch/libs/speex' make[4]: Leaving directory `/home/installers/freeswitch/libs/speex' make[3]: Leaving directory `/home/installers/freeswitch/libs/speex' make[2]: Leaving directory `/home/installers/freeswitch/libs/speex' cat /home/installers/freeswitch/src/include/switch_cpp.h | perl /home/installers/freeswitch/build/strip.pl > /home/installers/freeswitch/src/include/switch_swigable_cpp.h make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive make[2]: Entering directory `/home/installers/freeswitch' In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:52, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/logging.h:85: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?span_log_test? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:56, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/queue.h:67: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?queue_empty? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:77, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/crc.h:65: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?crc_itu32_check? /home/installers/freeswitch/libs/spandsp/src/spandsp/crc.h:98: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?crc_itu16_check? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:78, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/async.h:175: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/async.h:209: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:79, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:88, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:93, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v8.h:138: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v8.h:150: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:94, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v42.h:89: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v42.h:90: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:96, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:97, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:98, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:99, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:103, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:104, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:105, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:109, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/v18.h:121: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:112, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:365: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:117, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t85.h:67: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t85_analyse_header? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:118, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t42.h:79: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t42_analyse_header? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:122, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t30.h:193: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:123, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:232: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:373: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:436: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:443: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:126, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t35.h:88: error: expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t35_decode? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:127, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119: error: expected specifier-qualifier-list before ?bool? cc1: warnings being treated as errors /home/installers/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:132: warning: struct has no members In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:130, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:131, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:132, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66: error: expected specifier-qualifier-list before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:133, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125: error: expected declaration specifiers or ?...? before ?bool? In file included from /home/installers/freeswitch/libs/spandsp/src/spandsp.h:134, from ./src/include/private/switch_core_pvt.h:35, from src/switch.c:53: /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:96: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:103: error: expected declaration specifiers or ?...? before ?bool? /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:111: *error: expected declaration specifiers or ?...? before ?bool?* /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:118: *error: expected declaration specifiers or ?...? before ?bool?* make[2]: *** [freeswitch-switch.o] Error 1 make[2]: Leaving directory `/home/installers/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/installers/freeswitch' make: *** [current] Error 2 [xx at yy freeswitch]$ Please help what am i doing wrong.. Earlier it used to upgrade with no issues. My OS : Linux host.abc.com 2.6.18-194.el5 #1 SMP Tue Mar 16 21:52:39 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux Thanks, -Vishal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/692f95f7/attachment-0001.html From lloyd.aloysius at gmail.com Thu Oct 10 20:24:04 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 10 Oct 2013 12:24:04 -0400 Subject: [Freeswitch-users] Error While upgrading from 1.05 to latest Git head Message-ID: Visal Did you install all the dependencies for the 1.5 ? http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide Lloyd * * On Thu, Oct 10, 2013 at 12:12 PM, Vishal Kakkar wrote: > Hi All, > > Today i tried to upgrade FS from (Version 1.5.2b git a433b97 2013-06-07 > 17:29:42Z) to latest one using make current. > > Following is the error i am getting- > > > /bin/sh ../libtool --tag=CC --mode=link gcc -g -O2 -fvisibility=hidden > -o libspeexdsp.la -rpath /usr/local/freeswitch/lib -no-undefined > -version-info 6:0:5 preprocess.lo jitter.lo mdf.lo fftwrap.lo filterbank.lo > resample.lo buffer.lo scal.lo smallft.lo -lm > ar cru .libs/libspeexdsp.a preprocess.o jitter.o mdf.o fftwrap.o > filterbank.o resample.o buffer.o scal.o smallft.o > ranlib .libs/libspeexdsp.a > creating libspeexdsp.la > (cd .libs && rm -f libspeexdsp.la && ln -s ../libspeexdsp.la > libspeexdsp.la) > make[4]: Leaving directory > `/home/installers/freeswitch/libs/speex/libspeex' > Making all in include > make[4]: Entering directory > `/home/installers/freeswitch/libs/speex/include' > Making all in speex > make[5]: Entering directory > `/home/installers/freeswitch/libs/speex/include/speex' > make[5]: Nothing to be done for `all'. > make[5]: Leaving directory > `/home/installers/freeswitch/libs/speex/include/speex' > make[5]: Entering directory > `/home/installers/freeswitch/libs/speex/include' > make[5]: Nothing to be done for `all-am'. > make[5]: Leaving directory `/home/installers/freeswitch/libs/speex/include' > make[4]: Leaving directory `/home/installers/freeswitch/libs/speex/include' > make[4]: Entering directory `/home/installers/freeswitch/libs/speex' > make[4]: Leaving directory `/home/installers/freeswitch/libs/speex' > make[3]: Leaving directory `/home/installers/freeswitch/libs/speex' > make[2]: Leaving directory `/home/installers/freeswitch/libs/speex' > cat /home/installers/freeswitch/src/include/switch_cpp.h | perl > /home/installers/freeswitch/build/strip.pl > > /home/installers/freeswitch/src/include/switch_swigable_cpp.h > make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" > /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq > )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; > done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then > tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e > "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in > $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" > "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" > /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq > )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install > ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then > tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e > "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in > $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" > "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" > /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" > | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; > echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" > /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" > | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; > echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" > /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" > | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; > echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" > /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" > | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; > done)"; echo $mods )" `test -n "" || echo -s` all-recursive > make[2]: Entering directory `/home/installers/freeswitch' > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:52, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/logging.h:85: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?span_log_test? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:56, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/queue.h:67: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?queue_empty? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:77, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/crc.h:65: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?crc_itu32_check? > /home/installers/freeswitch/libs/spandsp/src/spandsp/crc.h:98: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?crc_itu16_check? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:78, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/async.h:175: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/async.h:209: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:79, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:88, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:93, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v8.h:138: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v8.h:150: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:94, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v42.h:89: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v42.h:90: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:96, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:97, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:98, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:99, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:103, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:104, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:105, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:109, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/v18.h:121: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:112, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:365: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:117, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t85.h:67: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t85_analyse_header? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:118, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t42.h:79: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t42_analyse_header? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:122, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t30.h:193: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:123, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:232: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:373: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:436: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:443: error: > expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:126, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t35.h:88: error: > expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t35_decode? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:127, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119: > error: expected specifier-qualifier-list before ?bool? > cc1: warnings being treated as errors > /home/installers/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:132: > warning: struct has no members > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:130, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:131, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:132, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66: > error: expected specifier-qualifier-list before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:133, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115: > error: expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125: > error: expected declaration specifiers or ?...? before ?bool? > In file included from > /home/installers/freeswitch/libs/spandsp/src/spandsp.h:134, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch.c:53: > /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:96: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:103: error: > expected declaration specifiers or ?...? before ?bool? > /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:111: *error: > expected declaration specifiers or ?...? before ?bool?* > /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:118: *error: > expected declaration specifiers or ?...? before ?bool?* > make[2]: *** [freeswitch-switch.o] Error 1 > make[2]: Leaving directory `/home/installers/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/home/installers/freeswitch' > make: *** [current] Error 2 > [xx at yy freeswitch]$ > > > Please help what am i doing wrong.. Earlier it used to upgrade with no > issues. > > My OS : Linux host.abc.com 2.6.18-194.el5 #1 SMP Tue Mar 16 21:52:39 EDT > 2010 x86_64 x86_64 x86_64 GNU/Linux > > Thanks, > -Vishal. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/c6b04778/attachment-0001.html From krice at freeswitch.org Thu Oct 10 20:29:01 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 10 Oct 2013 11:29:01 -0500 Subject: [Freeswitch-users] Error While upgrading from 1.05 to latest Git head In-Reply-To: Message-ID: Make spandsp-reconf then make current again On 10/10/13 11:24 AM, "Lloyd Aloysius" wrote: > Visal > > Did you install all the dependencies for the 1.5 ? > > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide > > Lloyd > > > ? > > > On Thu, Oct 10, 2013 at 12:12 PM, Vishal Kakkar > wrote: >> Hi All, >> >> Today i tried to upgrade FS from (Version 1.5.2b git a433b97 2013-06-07 >> 17:29:42Z) to latest one using make current. >> >> Following is the error i am getting- >> >> >> /bin/sh ../libtool --tag=CC --mode=link gcc? -g -O2 -fvisibility=hidden?? -o >> libspeexdsp.la -rpath /usr/local/freeswitch/lib >> -no-undefined -version-info 6:0:5 preprocess.lo jitter.lo mdf.lo fftwrap.lo >> filterbank.lo resample.lo buffer.lo scal.lo smallft.lo? -lm >> ar cru .libs/libspeexdsp.a? preprocess.o jitter.o mdf.o fftwrap.o >> filterbank.o resample.o buffer.o scal.o smallft.o >> ranlib .libs/libspeexdsp.a >> creating libspeexdsp.la >> (cd .libs && rm -f libspeexdsp.la && ln -s >> ../libspeexdsp.la libspeexdsp.la >> ) >> make[4]: Leaving directory `/home/installers/freeswitch/libs/speex/libspeex' >> Making all in include >> make[4]: Entering directory `/home/installers/freeswitch/libs/speex/include' >> Making all in speex >> make[5]: Entering directory >> `/home/installers/freeswitch/libs/speex/include/speex' >> make[5]: Nothing to be done for `all'. >> make[5]: Leaving directory >> `/home/installers/freeswitch/libs/speex/include/speex' >> make[5]: Entering directory `/home/installers/freeswitch/libs/speex/include' >> make[5]: Nothing to be done for `all-am'. >> make[5]: Leaving directory `/home/installers/freeswitch/libs/speex/include' >> make[4]: Leaving directory `/home/installers/freeswitch/libs/speex/include' >> make[4]: Entering directory `/home/installers/freeswitch/libs/speex' >> make[4]: Leaving directory `/home/installers/freeswitch/libs/speex' >> make[3]: Leaving directory `/home/installers/freeswitch/libs/speex' >> make[2]: Leaving directory `/home/installers/freeswitch/libs/speex' >> cat /home/installers/freeswitch/src/include/switch_cpp.h | perl >> /home/installers/freeswitch/build/strip.pl > >> /home/installers/freeswitch/src/include/switch_swigable_cpp.h >> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" >> /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq >> )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; >> done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then >> tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e >> "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in >> $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" >> "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" >> /home/installers/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq >> )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; >> done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then >> tmp_mods="$(grep -v "#" /home/installers/freeswitch/modules.conf | sed -e >> "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in >> $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" >> "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" >> /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | >> sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo >> $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" >> /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | >> sort | uniq )";? mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; >> echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" >> /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | >> sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; >> echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" >> /home/installers/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | >> sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; >> echo $mods )" `test -n "" || echo -s` all-recursive >> make[2]: Entering directory `/home/installers/freeswitch' >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:52, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/logging.h:85: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?span_log_test? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:56, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/queue.h:67: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?queue_empty? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:77, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/crc.h:65: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?crc_itu32_check? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/crc.h:98: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?crc_itu16_check? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:78, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/async.h:175: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/async.h:209: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:79, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:88, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:93, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v8.h:138: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v8.h:150: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:94, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v42.h:89: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v42.h:90: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:96, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:97, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:98, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:99, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:103, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:104, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:105, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:109, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/v18.h:121: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:112, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:365: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:117, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t85.h:67: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t85_analyse_header? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:118, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t42.h:79: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t42_analyse_header? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:122, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t30.h:193: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:123, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:232: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:373: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:436: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t30_api.h:443: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:126, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t35.h:88: error: >> expected ?=?, ?,?, ?;?, ?asm? or ?__attribute__? before ?t35_decode? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:127, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119: >> error: expected specifier-qualifier-list before ?bool? >> cc1: warnings being treated as errors >> /home/installers/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:132: >> warning: struct has no members >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:130, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375: error: >> expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:131, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99: >> error: expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:132, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66: error: >> expected specifier-qualifier-list before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177: >> error: expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:133, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115: >> error: expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125: >> error: expected declaration specifiers or ?...? before ?bool? >> In file included from >> /home/installers/freeswitch/libs/spandsp/src/spandsp.h:134, >> ???????????????? from ./src/include/private/switch_core_pvt.h:35, >> ???????????????? from src/switch.c:53: >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:96: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:103: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:111: error: >> expected declaration specifiers or ?...? before ?bool? >> /home/installers/freeswitch/libs/spandsp/src/spandsp/t31.h:118: error: >> expected declaration specifiers or ?...? before ?bool? >> make[2]: *** [freeswitch-switch.o] Error 1 >> make[2]: Leaving directory `/home/installers/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/home/installers/freeswitch' >> make: *** [current] Error 2 >> [xx at yy freeswitch]$ >> >> >> Please help what am i doing wrong.. Earlier it used to upgrade with no >> issues. >> >> My OS : Linux host.abc.com 2.6.18-194.el5 #1 SMP Tue >> Mar 16 21:52:39 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux >> >> Thanks, >> -Vishal. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/a511a783/attachment-0001.html From mike at jerris.com Thu Oct 10 20:41:10 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Oct 2013 12:41:10 -0400 Subject: [Freeswitch-users] Creating a rss reading solution using mod rss and espeak? In-Reply-To: <000001cec5d3$4738f710$d5aae530$@gmail.com> References: <000001cec5d3$4738f710$d5aae530$@gmail.com> Message-ID: <6D7213E5-938B-4435-9C76-E5C9FF742A7D@jerris.com> You should be able to use any engine that support MRCP https://wiki.freeswitch.org/wiki/Mrcp_client On Oct 10, 2013, at 12:10 PM, Pranav Lal wrote: > Hi all, > > Warning, this is not a telephony question per se but yes, it does involve > the use of freeswitch. > I have an octogenarian grandmother. She has a hard time entertaining herself > so I want to build a newspaper reading solution using freeswitch. The > advantage of using freeswitch is that she is familiar with a phone so can > dial in and work the phone keypad. I plan to give her a tablet with a voice > over IP application so that she can hit the phone keys and use it > independently. Yes, I know this is iffy but it is the best I can do right > now. > Most of our local newspapers do have rss feeds so I am planning to use mod > rss. The wiki suggests I use mod_cepstral. My grandmother finds it easier to > communicate in Hindi. I did not see a cepstral voice for Hindi hence I want > to use espeak. My questions are as follows. > 1. Can I use espeak with mod rss? If so what are the steps for doing so? > 2. I see the java script at http://wiki.freeswitch.org/wiki/Mod_rss which > pulls the rss feed. Does anyone have an example where multiple feeds are > rendered through freeswitch? > > Note: > I already have freeswitch compiled and running on my raspberry pi. > > Suggestions are welcome. > Pranav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/22336cda/attachment.html From ascensiontech at gmail.com Thu Oct 10 21:25:26 2013 From: ascensiontech at gmail.com (Peter Hartmann) Date: Thu, 10 Oct 2013 13:25:26 -0400 Subject: [Freeswitch-users] troubleshooting packet loss Message-ID: Hi, I just want to make sure I'm understanding something about troubleshooting RTP packet loss. I've done a capture between a handset and the Freeswitch server on a local network and during the call there were obvious audio drop outs from the incoming side. The analysis in Wireshark showed there were zero packets lost in either direction. My question: is this what 'outside' problems look like? Thanks, Peter From ascensiontech at gmail.com Thu Oct 10 22:04:56 2013 From: ascensiontech at gmail.com (Peter Hartmann) Date: Thu, 10 Oct 2013 14:04:56 -0400 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: References: Message-ID: Thanks Anton. Do you have some way of differentiating these dead calls from actual calls in progress? I'd rather not kill someones active call. Cheers, Peter On Thu, Oct 10, 2013 at 2:09 AM, Anton VG wrote: > I clean such calls with external script, using uuid_kill - but yes, the > calls are there. > > > 2013/10/10 Peter Hartmann >> >> Hi, >> First, thanks for Freeswitch! I'm experiencing an issue where 'show >> calls' returns several calls that aren't actually happening both >> inbound and outbound. Has anyone seen this before? >> >> Rebooting the handset (Polycom IP 550) associated with that extension >> has no effect so it seems in FS. >> >> freeswitch at internal> show calls >> >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch >> d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 >> >> 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 >> >> 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 >> >> 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,unknown >> ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 >> >> 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 >> 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension >> 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound >> Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension >> 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, >> bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 >> >> 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 >> >> 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 >> >> 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 >> >> 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> >> 9 total. >> >> >> Running: >> FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 >> 2013-10-02 21:30:46Z) >> >> >> Thanks much! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From victor.chukalovskiy at gmail.com Fri Oct 11 00:07:01 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 10 Oct 2013 16:07:01 -0400 Subject: [Freeswitch-users] mod_translate question Message-ID: <525708E5.4070706@gmail.com> Hi All, How do I specify translation profile when using this line in SIP profile? What I'm trying to achieve is to make sure call only goes through one translations profile. I don't want to rely on the order of translations profiles in translation.conf. So, I want to do something like this: in translate.conf.xml I have: Thank you, Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/fc2598d7/attachment.html From ssinyagin at yahoo.com Fri Oct 11 00:50:05 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 10 Oct 2013 13:50:05 -0700 (PDT) Subject: [Freeswitch-users] Intel Atom performance Message-ID: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> hi, can anyone share some useful experience with Intel Atom CPU? If used with FreeSWITCH, -- what's the maximum number of concurrent transcoding calls without service degradation? -- maximum CPS without service degradation? -- maximum size of a conference call (with or without transcoding) There's a choice of fanless appliances with 1.6 GHz Atom processors, and I wonder how well they can perform as PBX. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/e3a3b4ef/attachment.html From nickolayr at gmail.com Fri Oct 11 00:58:33 2013 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Thu, 10 Oct 2013 16:58:33 -0400 Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: https://wiki.freeswitch.org/wiki/Real-world_results -- Rogoshchenkov Nikolay On Thu, Oct 10, 2013 at 4:50 PM, Stanislav Sinyagin wrote: > hi, > > can anyone share some useful experience with Intel Atom CPU? If used with > FreeSWITCH, > > -- what's the maximum number of concurrent transcoding calls without > service degradation? > -- maximum CPS without service degradation? > -- maximum size of a conference call (with or without transcoding) > > There's a choice of fanless appliances with 1.6 GHz Atom processors, and I > wonder how well they can perform as PBX. > > thanks > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/184cddd2/attachment-0001.html From grcamauer at gmail.com Fri Oct 11 01:12:20 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 10 Oct 2013 18:12:20 -0300 Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: You might want to specify the codecs for transcoding since they are not all the same and also the ATOM processor type, as there are many generations now. Guillermo On Thu, Oct 10, 2013 at 5:50 PM, Stanislav Sinyagin wrote: > hi, > > can anyone share some useful experience with Intel Atom CPU? If used with > FreeSWITCH, > > -- what's the maximum number of concurrent transcoding calls without > service degradation? > -- maximum CPS without service degradation? > -- maximum size of a conference call (with or without transcoding) > > There's a choice of fanless appliances with 1.6 GHz Atom processors, and I > wonder how well they can perform as PBX. > > thanks > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/68890112/attachment.html From ssinyagin at yahoo.com Fri Oct 11 01:14:35 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 10 Oct 2013 14:14:35 -0700 (PDT) Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: <1381439675.23786.YahooMailNeo@web126204.mail.ne1.yahoo.com> I forgot to mention that I saw that page, and it doesn't give much. On Thursday, October 10, 2013 11:01 PM, Nikolay Rogoshchenkov wrote: https://wiki.freeswitch.org/wiki/Real-world_results -- Rogoshchenkov Nikolay On Thu, Oct 10, 2013 at 4:50 PM, Stanislav Sinyagin wrote: hi, > >can anyone share some useful experience with Intel Atom CPU? If used with FreeSWITCH, > >-- what's the maximum number of concurrent transcoding calls without service degradation? >-- maximum CPS without service degradation? >-- maximum size of a conference call (with or without transcoding) > >There's a choice of fanless appliances with 1.6 GHz Atom processors, and I wonder how well they can perform as PBX. > >thanks > > > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/b20068f1/attachment.html From ssinyagin at yahoo.com Fri Oct 11 01:59:33 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 10 Oct 2013 14:59:33 -0700 (PDT) Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: <1381439675.23786.YahooMailNeo@web126204.mail.ne1.yahoo.com> References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> <1381439675.23786.YahooMailNeo@web126204.mail.ne1.yahoo.com> Message-ID: <1381442373.81067.YahooMailNeo@web126201.mail.ne1.yahoo.com> any available information would be welcome as a reference :) otherwise, I will do the testing myself with available Atom processors, and then I will specify the exact parameters :) On Thursday, October 10, 2013 11:16 PM, Stanislav Sinyagin wrote: I forgot to mention that I saw that page, and it doesn't give much. On Thursday, October 10, 2013 11:01 PM, Nikolay Rogoshchenkov wrote: https://wiki.freeswitch.org/wiki/Real-world_results -- Rogoshchenkov Nikolay On Thu, Oct 10, 2013 at 4:50 PM, Stanislav Sinyagin wrote: hi, > >can anyone share some useful experience with Intel Atom CPU? If used with FreeSWITCH, > >-- what's the maximum number of concurrent transcoding calls without service degradation? >-- maximum CPS without service degradation? >-- maximum size of a conference call (with or without transcoding) > >There's a choice of fanless appliances with 1.6 GHz Atom processors, and I wonder how well they can perform as PBX. > >thanks > > > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/17f5ec3c/attachment-0001.html From john_platts at hotmail.com Fri Oct 11 02:54:40 2013 From: john_platts at hotmail.com (John Platts) Date: Thu, 10 Oct 2013 17:54:40 -0500 Subject: [Freeswitch-users] Fax calls drop in FreeSWITCH 1.2.12 Message-ID: We have had fax calls drop in FreeSWITCH 1.2.12, but we have been able to get the problem solved by enabling proxy media for fax calls. The problem occurs when neither proxy media nor bypass media are enabled. We need to get fax to email and T.38 support working. We need to get these features working correctly. We also need to have faxes working consistently. We need a solution that will allow faxes to work consistently when neither proxy media nor bypass media are enabled. From lloyd.aloysius at gmail.com Fri Oct 11 04:26:44 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 10 Oct 2013 20:26:44 -0400 Subject: [Freeswitch-users] c function - find the user id from number alias Message-ID: Hi All, Can someone tell me which c function that i can use to get user id from number-alias. I have the user directory like this I know the number-alias and want to find the user-id. I could not find any in the source. Thank you Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/1ef4c875/attachment.html From nneul at mst.edu Fri Oct 11 05:49:10 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 10 Oct 2013 20:49:10 -0500 Subject: [Freeswitch-users] c function - find the user id from number alias In-Reply-To: References: Message-ID: <52575916.9010908@mst.edu> I don't know the function, but I seem to remember seeing code that was something of the nature of "find xml by userid" and it would search for either the userid or the alias. You could use that, and then grab the 'id' attribute from the resulting object. -- Nathan On 10/10/2013 07:26 PM, Lloyd Aloysius wrote: > Hi All, > > Can someone tell me which c function that i can use to get user id from number-alias. > > I have the user directory like this > > I know the number-alias and want to find the user-id. I could not find any in the source. > > > Thank you > Lloyd > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From karl at xtronics.com Fri Oct 11 06:19:10 2013 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 10 Oct 2013 21:19:10 -0500 Subject: [Freeswitch-users] recommendations for Wifi SIP phones? In-Reply-To: <5244B020.1080504@xtronics.com> References: <523B8595.4000502@mst.edu> <523B92F1.2070407@xtronics.com> <5244B020.1080504@xtronics.com> Message-ID: <5257601E.2010002@xtronics.com> A few more notes: It appears that Skype and Google voice work well enough over wifi - but not SIP. So I'm starting to think that the SIP protocol stack is the problem? There are other standards - Most are proprietary, but it looks like jingle is on a BSD licenses. I can't seem to find anything that mentions the license of IAX2 ?? There are some softphones that support IAX2. Mod Opal lets freeswitch talk via IAX2 - I don't know if it is working? If anyone has tested a IAX2/freeswitch/wifi combination - I would be quite interested in the results. Squinting into the future - the question is if the new wifi standard 802.11ac or a move beyond SIP will be the solution to VoIP/wifi It does look like 802.11ac has addressed some latency issues with MU-MIMO, but I've been unable to find any real-world numbers. This is a complex new standard that will take a while to get debugged and bench marked. Could be that any wifi that has more than one connection will glitch audio - and MU-MIMO could fix that. Right now it still looks like a SIP world that we live in.. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Prosperity is the best protector of principle. -- Mark Twain -------------------------------------------------------------------------------- From fs at voice2net.ca Fri Oct 11 06:56:46 2013 From: fs at voice2net.ca (fs) Date: Thu, 10 Oct 2013 22:56:46 -0400 Subject: [Freeswitch-users] call park blf References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch><9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch><52541FB2.9090602@integrafin.co.uk> <5255A520.8040909@integrafin.co.uk> Message-ID: <003601cec62d$871c4d50$4dd1a8c0@DARCY> I use park+{slot} to park calls using keys on snom and grandstream phones. The busy lamp field lights but goes out shortly after, about 5 seconds. However, the call is still on hold and can be picked up by pressing the illuminated light. When I do a sip trace, there is a notify message saying the park+{slot} has been terminated. Any ideals. Incidentally, all other BLF's work prefectly. Following is the settings in the dial plan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/0b7a3ac3/attachment.html From fs at voice2net.ca Fri Oct 11 06:57:57 2013 From: fs at voice2net.ca (fs) Date: Thu, 10 Oct 2013 22:57:57 -0400 Subject: [Freeswitch-users] call park blf Message-ID: <003f01cec62d$b121bf90$4dd1a8c0@DARCY> sorry, I forgot to paste in the settings in the dial plan I use park+{slot} to park calls using keys on snom and grandstream phones. The busy lamp field lights but goes out shortly after, about 5 seconds. However, the call is still on hold and can be picked up by pressing the illuminated light. When I do a sip trace, there is a notify message saying the park+{slot} has been terminated. Any ideals. Incidentally, all other BLF's work prefectly. Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/b66e09b5/attachment.html From moises.silva at gmail.com Fri Oct 11 07:21:10 2013 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 10 Oct 2013 23:21:10 -0400 Subject: [Freeswitch-users] FreeTDM - Using ftdm_channel_wait with FTDM_EVENTS In-Reply-To: <1381305570.72484.YahooMailNeo@web172103.mail.ir2.yahoo.com> References: <1381135949.34409.YahooMailNeo@web172103.mail.ir2.yahoo.com> <1381235486.41730.YahooMailNeo@web172102.mail.ir2.yahoo.com> <1381242960.47963.YahooMailNeo@web172105.mail.ir2.yahoo.com> <1381305570.72484.YahooMailNeo@web172103.mail.ir2.yahoo.com> Message-ID: Hello, I think your question would be better suited to the freeswitch-dev mailing list given that is about C development with FreeSWITCH APIs, but I'll answer here anyways. 1. What kind of events can cause the exit of "ftdm_channel_wait" function? > I've seen the "ftdm_event_type_t" that defines "FTDM_EVENT_DTMF" and > "FTDM_EVENT_OOB". Is it the answer to my question? > Events such as CAS/RBS events and DTMF events if you have hardware DTMF enabled (and your hardware supports it). Given that you are using Sangoma hardware, it should work if you have a hardware DTMF enabled in the wanpipe configuration files (ie /etc/wanpipe/wanpipe1.conf) > 2. Is there something that I need to do before call "ftdm_channel_wait" to > enable the "events generation"? > Did you call ftdm_span_start() to start the signaling thread on the span? Cheers, Moy *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131010/a0ed7abf/attachment-0001.html From max at nysolutions.com Fri Oct 11 07:22:32 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 11 Oct 2013 03:22:32 +0000 Subject: [Freeswitch-users] call park blf In-Reply-To: <003f01cec62d$b121bf90$4dd1a8c0@DARCY> References: <003f01cec62d$b121bf90$4dd1a8c0@DARCY> Message-ID: What firmware version are you using on Snom? I am using it on Polycom without any issues, however in the Fusion PBX IRC channel someone tried every Snom firmware version and it only work on 1 version. This is the dialplan FusionPBX uses. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of fs Sent: Thursday, October 10, 2013 10:58 PM To: fs; FreeSWITCH Users Help Subject: Re: [Freeswitch-users] call park blf sorry, I forgot to paste in the settings in the dial plan I use park+{slot} to park calls using keys on snom and grandstream phones. The busy lamp field lights but goes out shortly after, about 5 seconds. However, the call is still on hold and can be picked up by pressing the illuminated light. When I do a sip trace, there is a notify message saying the park+{slot} has been terminated. Any ideals. Incidentally, all other BLF's work prefectly. Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/94af06f7/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/94af06f7/attachment.jpg From coslovich.innovats at yahoo.it Fri Oct 11 11:23:53 2013 From: coslovich.innovats at yahoo.it (Andrea Coslovich) Date: Fri, 11 Oct 2013 08:23:53 +0100 (BST) Subject: [Freeswitch-users] FreeTDM - Using ftdm_channel_wait with FTDM_EVENTS In-Reply-To: References: <1381135949.34409.YahooMailNeo@web172103.mail.ir2.yahoo.com> <1381235486.41730.YahooMailNeo@web172102.mail.ir2.yahoo.com> <1381242960.47963.YahooMailNeo@web172105.mail.ir2.yahoo.com> <1381305570.72484.YahooMailNeo@web172103.mail.ir2.yahoo.com> Message-ID: <1381476233.376.YahooMailNeo@web172102.mail.ir2.yahoo.com> Hello everybody and many thanks to Moy for its useful reply! >>1. What kind of events can cause the exit of "ftdm_channel_wait" function? >>I've seen the "ftdm_event_type_t" that defines "FTDM_EVENT_DTMF" and "FTDM_EVENT_OOB". Is it the answer to my question? >Events such as CAS/RBS events and DTMF events if you have hardware DTMF enabled (and your hardware supports it). >Given that you are using Sangoma hardware, it should work if you have a hardware DTMF enabled in the wanpipe >configuration files (ie /etc/wanpipe/wanpipe1.conf) I'm using an E1 PRI so, if I've correctly understood, CAS/RBS events don't exist. Regarding DTMF I've found some TDMV_HW_DTMF=NO in my wanpipe*.conf, now I will correct them to TDMV_HW_DTMF=YES and try again. ? >>2. Is there something that I need to do before call "ftdm_channel_wait" to enable the "events generation"? >Did you call ftdm_span_start() to start the signaling thread on the span? Yes, of course! :) Thanks again. Best regards! Andrea -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/0e808ae1/attachment.html From vipkilla at gmail.com Fri Oct 11 16:03:42 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 11 Oct 2013 08:03:42 -0400 Subject: [Freeswitch-users] mod_snmp and freeswitch MIB Message-ID: FreeSWITCH MIB is now included in Solarwind's Orion MIB I only had to submit the request to mibs at solarwinds.com :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/cb0c463d/attachment.html From fs at voice2net.ca Fri Oct 11 16:13:49 2013 From: fs at voice2net.ca (fs) Date: Fri, 11 Oct 2013 08:13:49 -0400 Subject: [Freeswitch-users] call park blf References: <003f01cec62d$b121bf90$4dd1a8c0@DARCY> Message-ID: <002a01cec67b$58876b70$4dd1a8c0@DARCY> We use snom version 8.7.3.10. The dial plan for snom is immediately below. You need one dial plan to park and one dial plan to unpark. The strange thing, this worked perfectly for about 1 month then it started dropping the blf for parked calls, not just on snom phones but also on grandstream. Might be tied to an upgrade we did on the freeswitch. we use FreeSWITCH Version 1.2.12+git~20130909T220327Z~07f2d94d9c (git 07f2d94 2013-09-09 22:03:27Z) Darcy Primrose ----- Original Message ----- From: Moishe Grunstein To: FreeSWITCH Users Help Sent: Thursday, October 10, 2013 11:22 PM Subject: Re: [Freeswitch-users] call park blf What firmware version are you using on Snom? I am using it on Polycom without any issues, however in the Fusion PBX IRC channel someone tried every Snom firmware version and it only work on 1 version. This is the dialplan FusionPBX uses. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of fs Sent: Thursday, October 10, 2013 10:58 PM To: fs; FreeSWITCH Users Help Subject: Re: [Freeswitch-users] call park blf sorry, I forgot to paste in the settings in the dial plan I use park+{slot} to park calls using keys on snom and grandstream phones. The busy lamp field lights but goes out shortly after, about 5 seconds. However, the call is still on hold and can be picked up by pressing the illuminated light. When I do a sip trace, there is a notify message saying the park+{slot} has been terminated. Any ideals. Incidentally, all other BLF's work prefectly. Darcy Primrose ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4158 / Virus Database: 3609/6738 - Release Date: 10/10/13 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/a666dc65/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/a666dc65/attachment-0001.jpe From krice at freeswitch.org Fri Oct 11 17:00:35 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 11 Oct 2013 08:00:35 -0500 Subject: [Freeswitch-users] mod_snmp and freeswitch MIB In-Reply-To: Message-ID: Cool! On 10/11/13 7:03 AM, "Vik Killa" wrote: > FreeSWITCH MIB is now included in Solarwind's Orion MIB > I only had to submit the request to?mibs at solarwinds.com > :-) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/0170daad/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:31:06 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:31:06 +0100 Subject: [Freeswitch-users] originate not creating xml_cdr record In-Reply-To: <52568A37.9000502@inveroak.com> References: <52307F11.9020704@inveroak.com> <5230B5EB.9050909@inveroak.com> <5230BCA0.9090102@inveroak.com> <5231C8A1.7030407@inveroak.com> <5231D135.3000207@inveroak.com> <52568A37.9000502@inveroak.com> Message-ID: Glad to hear you were able to get it sorted, thanks for letting us know! Cal On Thu, Oct 10, 2013 at 12:06 PM, Matt Broad wrote: > Hi > > just as an update to this thread, sorry for the delay :) > > I have resolved the issue. Freeswitch was behaving impeccably, my XML > parsing code on the other hand was not :) > > Error handling when a node is not present is always a good thing to have! > > > thanks > Matt > > > > On 12/09/2013 15:35, Matt Broad wrote: > > Hi Cal, > > have just seen the reply from Lloyd (my email's threading isn't too good), > yes the bleg logging is set to true, and is working fine on calls > instigated via an incoming call. > > I'll take a look at the bug reporting pages and try and give as much info > as possible. > > > > thanks > Matt > > > On 12/09/2013 15:24, Cal Leeming [Simplicity Media Ltd] wrote: > > Just to be sure, can you double check that b_leg is set to true in your > xml_cdr.conf (as per Lloyds' suggestion). > > If it is set to true, then next step would be to raise a bug report in > JIRA with as much info as possible about the steps you have taken to > reproduce this bug. > > If you are unfamiliar with this process, please see; > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Where_And_How_To_File_Bug_Reports > > Sadly I won't have any time to look into this until late next week, but > perhaps another volunteer will jump on and take a look before I do. > > Cal > > > > On Thu, Sep 12, 2013 at 2:58 PM, Matt Broad wrote: > >> Hi Cal, >> >> thanks for the suggestions. >> I have tested adding an err log dir and also enabled the >> log-http-and-disk option. >> >> These work fine when I make a normal call (I have forced a http error on >> the webserver to test the error logging), but when doing an originate from >> the CLI, there are no logs at all. >> >> I don't see any messages about a failed log ("mod_xml_cdr.c:365 Got error >> [0] posting to web server http://******" & "mod_xml_cdr.c:372 Retry will >> be with url http://****") when the call ends. >> >> thanks >> Matt >> >> >> >> On 11/09/2013 20:29, Cal Leeming [Simplicity Media Ltd] wrote: >> >> Oh damn, sorry I was thinking of mod_xml_curl! >> >> In that case, try this instead; >> http://wiki.freeswitch.org/wiki/Mod_xml_cdr >> >> >> Enable that param, and check to see if the request ends up in that >> directory. >> >> You can also enable 'log-http-and-disk' which will ensure the request >> is always written to disk. >> >> You can also enable access logs on your web server to check if the >> request is received, or even just run "nc -l 80" on the web server and wait >> for the connection to be received. >> >> The main point of the above is to pinpoint where the fault is happening >> (if the entry is pushed to disk, then it means the problem is with your web >> server.. if the entry is not pushed to disk, then there's a problem >> somewhere in FS, whether it be in config or a bug). >> >> Sorry for the confusion >> >> Cal >> >> >> >> On Wed, Sep 11, 2013 at 7:55 PM, Matt Broad wrote: >> >>> Hi Cal, >>> >>> I have tried the method that you described but I get the error "-ERR >>> xml_cdr debug_on Command not found!" >>> could this be a typo? I have tried googling but with no success. >>> >>> thanks >>> Matt >>> >>> >>> On 11/09/2013 19:26, Matt Broad wrote: >>> >>> Thanks for the quick reply Cal. >>> >>> I will give that a try and see what I can see. I will report back with >>> my findings :) >>> >>> thanks >>> Matt >>> >>> >>> On 11/09/2013 15:55, Cal Leeming [Simplicity Media Ltd] wrote: >>> >>> Matt, >>> >>> As far as I know, the API originate method should also trigger xml_cdr >>> to post back to your server. >>> >>> Can you please run "console loglevel debug" and "xml_cdr debug_on", >>> attempt to use api originate, then see if anything shows up in console? >>> (that command will enable xml_cdr debugging). >>> >>> If this still does not work, please capture all the logs and explain >>> the test procedure you went through and send it all in a JIRA ticket. >>> >>> Hope this helps >>> >>> Cal >>> >>> >>> On Wed, Sep 11, 2013 at 3:32 PM, Matt Broad wrote: >>> >>>> Hi, >>>> >>>> I have setup xml_cdr and it is working great, both aleg and bleg calls >>>> are logged. >>>> I am using the api originate method, but this does not create an xml_cdr >>>> record (though a record is logged in /log/cdr-csv/master/csv). Is this >>>> correct? >>>> >>>> >>>> >>>> thanks >>>> Matt >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/6e74b82d/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:32:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:32:15 +0100 Subject: [Freeswitch-users] Playing announcement file before MOH In-Reply-To: References: Message-ID: No problem, glad we could help. Once you have managed to achieve what you are trying to do, perhaps you could update this thread and share your approach as it will help others in the future. Cal On Thu, Oct 10, 2013 at 5:54 AM, Nandy Dagondon wrote: > Hi Cal, The "please_hold" vanilla dialplan in features.xml is what I > initially. The tips you shared - someday I can use them. Thanks a lot! > /Nandy > > > On Thu, Oct 10, 2013 at 11:46 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Have a look into some of these variables/options below, use google+wiki; >> >> uuid_displace - allows you to inject audio into a channel >> temp_hold_music - allows you specify music during transfer hold >> campon - places call into parking when on hold (not 100% on this, needs >> clarification) >> >> You could also modify the MOH file to play the message you require at the >> desired intervals. Realistically I think this would be the best way, as >> this will give you much more control over the quality delivered (fading, >> volume, etc). >> >> Hope this helps >> >> Cal >> >> >> >> On Thu, Oct 10, 2013 at 2:18 AM, Nandy Dagondon wrote: >> >>> Hello everyone, >>> >>> I like to play an announcement file before playing music-on-hold. Is >>> this possible? Any hint? >>> >>> Tks, >>> Nandy >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/6447819d/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:34:42 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:34:42 +0100 Subject: [Freeswitch-users] ANI missing in cli originate In-Reply-To: <1380624829.9197.75.camel@luna.madrid.commsmundi.com> References: <1380624829.9197.75.camel@luna.madrid.commsmundi.com> Message-ID: Hi Francois, I don't quite understand what this patch would be needed for, and I suspect others had similar thoughts based on the lack of response on this thread. Could you possibly raise a JIRA ticket explaining exactly what you intend for the patch to do, and why it is needed? Try and include as much detail as possible. Thanks Cal On Tue, Oct 1, 2013 at 11:53 AM, Fran?ois wrote: > Hello, > > I've seen some FS users using ANI for billing purposes, and I'm not sure > if it is normal that the ANI variable is missing on calls originated > from CLI (or other originate with no session associated). > > Would a small patch copying the specified Caller ID Number to ANI in > those cases be an acceptable solution? > > Thanks, > Fran?ois. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/283185e1/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:36:04 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:36:04 +0100 Subject: [Freeswitch-users] Integrate openfire to freeswitch In-Reply-To: <524C3BBF.3010306@sonerep.com> References: <524C3BBF.3010306@sonerep.com> Message-ID: Have a read of this; http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration_.28Openfire.29 Let me know if this is not what you are trying to achieve. Cal On Wed, Oct 2, 2013 at 4:29 PM, Groupe SOGO wrote: > Dear Friends, > > I installed freeswitch latest version on CentOS 6.4 and everything is > working well. > The default sip internal accounts can make and receive calls. > > I also installed openfire 3.8.1 on the same machine and it is working well. > I can create users with spark client 2.6.3 and they can chat and create > conference. > > So everything works very well. > > Now I want to integrate openfire to freeswitch. > Has anyone tried it? What did he do to achieve that? > What can i do to achieve the integration? > > thanks in advance for your help. > > Labolinux > K. Amouzou > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/979da8da/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:37:26 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:37:26 +0100 Subject: [Freeswitch-users] Automate wancfg In-Reply-To: References: Message-ID: Hi DN, Your question was very generic, and difficult for people to reply to. Can you please give us some more detail about what you have tried so far, and what exact problems you have encountered? You'll usually get a better response from the list if you give specific questions that people can answer. Cal On Thu, Oct 3, 2013 at 8:21 AM, DN Singh wrote: > Hello list, > > Has anyone tried to automate the wancfg configuration with freeswitch? I > have tried, but gets automatically configured with T1, even if I give it > parameters for E1. > I have done this by using wancfg_zaptel with some arguments. > > Please let me know if more details would be required. > > Cheers!! > DN > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/338963cc/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:43:16 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:43:16 +0100 Subject: [Freeswitch-users] park_after_bridge not working In-Reply-To: <0aaa01cec072$5538beb0$ffaa3c10$@verizon.net> References: <0aaa01cec072$5538beb0$ffaa3c10$@verizon.net> Message-ID: Andre, As far as I know, park_after_bridge cannot be used because using bypass_media would take FS out of the media path. Have a read of this, slightly related; http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/037592.html Can you please explain what you mean by "doesn't work". Please include as much debugging info as possible using; sofia global siptrace on sofia loglevel all 9 sofia tracelevel alert console loglevel debug fsctl debug_level 10 Thanks Cal On Thu, Oct 3, 2013 at 8:54 PM, Andre wrote: > Hi,**** > > ** ** > > My Park_after_bridge doesn?t work if I add bypass_media=true. How can I > get Park to work when I?m bypass_media mode?**** > > Thanks**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/7f7997ed/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 11 17:44:47 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 11 Oct 2013 14:44:47 +0100 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: References: Message-ID: Hi Peter, Can you please raise a JIRA for this, and also include a full SIP trace/logging for one of these calls, showing the answer/hangup etc. Cal On Mon, Oct 7, 2013 at 9:55 PM, Peter Hartmann wrote: > Hi, > First, thanks for Freeswitch! I'm experiencing an issue where 'show > calls' returns several calls that aren't actually happening both > inbound and outbound. Has anyone seen this before? > > Rebooting the handset (Polycom IP 550) associated with that extension > has no effect so it seems in FS. > > freeswitch at internal> show calls > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch > d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 > 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com > ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 > 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 > 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com > ,CS_EXECUTE,unknown > ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 > 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 > 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension > 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound > Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension > 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, > bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 > 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 > 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 > 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com > ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 > 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com > ,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > > 9 total. > > > Running: > FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 > 2013-10-02 21:30:46Z) > > > Thanks much! > > > > > Peter Hartmann > Hartmann Computer Consulting > http://blog.hartmanncomputer.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/d1a2617e/attachment.html From mike at jerris.com Fri Oct 11 18:02:02 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Oct 2013 10:02:02 -0400 Subject: [Freeswitch-users] recommendations for Wifi SIP phones? In-Reply-To: <5257601E.2010002@xtronics.com> References: <523B8595.4000502@mst.edu> <523B92F1.2070407@xtronics.com> <5244B020.1080504@xtronics.com> <5257601E.2010002@xtronics.com> Message-ID: <3B6185DF-4C4D-45B6-88CB-429FBCBFB1FF@jerris.com> Audio issues on wifi have nothing to do with the signaling protocol, they have everything to do with the codecs and other media features. My biggest issue with wifi based sip phones is the battery life sucks. Don't bother w/ IAX2 its a solution looking for a problem. Mike On Oct 10, 2013, at 10:19 PM, Karl Schmidt wrote: > A few more notes: > > It appears that Skype and Google voice work well enough over wifi - but not SIP. So I'm starting to > think that the SIP protocol stack is the problem? > > There are other standards - Most are proprietary, but it looks like jingle is on a BSD licenses. > > I can't seem to find anything that mentions the license of IAX2 ?? There are some softphones that > support IAX2. Mod Opal lets freeswitch talk via IAX2 - I don't know if it is working? If anyone has > tested a IAX2/freeswitch/wifi combination - I would be quite interested in the results. > > Squinting into the future - the question is if the new wifi standard 802.11ac or a move beyond SIP > will be the solution to VoIP/wifi > > It does look like 802.11ac has addressed some latency issues with MU-MIMO, but I've been unable to > find any real-world numbers. This is a complex new standard that will take a while to get debugged > and bench marked. Could be that any wifi that has more than one connection will glitch audio - and > MU-MIMO could fix that. > > Right now it still looks like a SIP world that we live in.. > From veerabhadrarao.kankatala at panamaxil.com Fri Oct 11 14:47:43 2013 From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadra Rao) Date: Fri, 11 Oct 2013 16:17:43 +0530 Subject: [Freeswitch-users] play hold music Message-ID: <5257D74F.7070808@panamaxil.com> hello I am Using Freeswitch 1.2.12 version. I want to play some music-on-hold(moh) sound on B leg when A leg dial C leg for establish three way conference. please help me. Thank you. From andrew at cassidywebservices.co.uk Fri Oct 11 18:42:23 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 11 Oct 2013 15:42:23 +0100 Subject: [Freeswitch-users] mod_translate question In-Reply-To: <525708E5.4070706@gmail.com> References: <525708E5.4070706@gmail.com> Message-ID: At the moment the only way I can see is to set country in the sofia or gateway configuration to the profile name. On 10 October 2013 21:07, Victor Chukalovskiy wrote: > Hi All, > > How do I specify translation profile when using this line in SIP profile? > > > > What I'm trying to achieve is to make sure call only goes through one > translations profile. I don't want to rely on the order of translations > profiles in translation.conf. So, I want to do something like this: > > > > in translate.conf.xml I have: > > > > > > > > > > > > > > > > > > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/3e6ed6c4/attachment-0001.html From victor.chukalovskiy at gmail.com Fri Oct 11 19:39:15 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Fri, 11 Oct 2013 11:39:15 -0400 Subject: [Freeswitch-users] mod_translate question In-Reply-To: References: <525708E5.4070706@gmail.com> Message-ID: <52581BA3.3040903@gmail.com> Thank you for the response! Hmm, Is there a way to set channel variable per SIP profile??? I tried this: And this: But it never picks-up the param. Still default to US: 2013-10-11 11:33:26.251629 [DEBUG] mod_translate.c:291 using default_country variable [US] for translate profile I understand "param" is not the same as channel variable. Thank you, Victor On 13-10-11 10:42 AM, Andrew Cassidy wrote: > At the moment the only way I can see is to set country in the sofia or > gateway configuration to the profile name. > > > > On 10 October 2013 21:07, Victor Chukalovskiy > > > wrote: > > Hi All, > > How do I specify translation profile when using this line in SIP > profile? > > > > What I'm trying to achieve is to make sure call only goes through > one translations profile. I don't want to rely on the order of > translations profiles in translation.conf. So, I want to do > something like this: > > > > in translate.conf.xml I have: > > > > > > > > > > replace="+1$1"/> > > > > > > > > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/2bfe46a4/attachment.html From andrew at cassidywebservices.co.uk Fri Oct 11 20:08:56 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 11 Oct 2013 17:08:56 +0100 Subject: [Freeswitch-users] mod_translate question In-Reply-To: <52581BA3.3040903@gmail.com> References: <525708E5.4070706@gmail.com> <52581BA3.3040903@gmail.com> Message-ID: No idea, I only attempted the documentation by reading the code I'm afraid. I have default_country set in vars.xml On 11 October 2013 16:39, Victor Chukalovskiy wrote: > Thank you for the response! Hmm, Is there a way to set channel variable > per SIP profile??? > > I tried this: > > > > And this: > > > > But it never picks-up the param. Still default to US: > > 2013-10-11 11:33:26.251629 [DEBUG] mod_translate.c:291 using > default_country variable [US] for translate profile > > I understand "param" is not the same as channel variable. > > Thank you, > Victor > > On 13-10-11 10:42 AM, Andrew Cassidy wrote: > > At the moment the only way I can see is to set country in the sofia or > gateway configuration to the profile name. > > > > On 10 October 2013 21:07, Victor Chukalovskiy < > victor.chukalovskiy at gmail.com> wrote: > >> Hi All, >> >> How do I specify translation profile when using this line in SIP profile? >> >> >> >> What I'm trying to achieve is to make sure call only goes through one >> translations profile. I don't want to rely on the order of translations >> profiles in translation.conf. So, I want to do something like this: >> >> >> >> in translate.conf.xml I have: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/fdea09a2/attachment-0001.html From krice at freeswitch.org Fri Oct 11 20:21:20 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 11 Oct 2013 11:21:20 -0500 Subject: [Freeswitch-users] Friday Free For All Today, 2P Eastern until ??? Message-ID: FreeSWITCHers, Today, 2P Eastern, join us on 888 for the weekly FridayFreeForAll! Today?s Topic? Who knows! Its the FreeForAll! Have a question? Join us and ask it, want to hang out and chat with the FreeSWITCH Devs or other FreeSWITCHers? Call in! Call sip:888 at conference.freeswitch.org or visit http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions for more info on joining! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/4b3056e8/attachment.html From ira at connectmevoice.com Fri Oct 11 20:26:49 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Fri, 11 Oct 2013 12:26:49 -0400 Subject: [Freeswitch-users] lua session question Message-ID: I have a lua script that I am creating a new session: legA = freeswitch.Session(dialA); Where dial is: {dtmf_type=none,originate_timeout=60,instant_ringback=true,origination_caller_id_name=xxxxxxxxxx,origination_caller_id_number=xxxxxxxxxx,followme_callid=e9886a3f-58c5-456a-949a-b5bd89b11b38}sofia/gateway/sr/xxxxxxxxxx|sofia/gateway/voipinnovations.com/xxxxxxxxxx When the session gets created, it is in the "default" context. How can I set the context for the channel to be created in? Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/29beb496/attachment.html From jaykris at gmail.com Fri Oct 11 20:27:19 2013 From: jaykris at gmail.com (JP) Date: Fri, 11 Oct 2013 09:27:19 -0700 Subject: [Freeswitch-users] play hold music In-Reply-To: <5257D74F.7070808@panamaxil.com> References: <5257D74F.7070808@panamaxil.com> Message-ID: Maybe the "Example 22: Play MOH while doing a database lookup" in https://wiki.freeswitch.org/wiki/Dialplan_XML will help you. -JP On Fri, Oct 11, 2013 at 3:47 AM, Veerabhadra Rao < veerabhadrarao.kankatala at panamaxil.com> wrote: > hello > > I am Using Freeswitch 1.2.12 version. > I want to play some music-on-hold(moh) sound on B leg when > A leg dial C leg for establish three way conference. > > please help me. > > Thank you. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/169faf57/attachment.html From smontour at verizon.net Fri Oct 11 21:16:18 2013 From: smontour at verizon.net (Sam Montour) Date: Fri, 11 Oct 2013 12:16:18 -0500 Subject: [Freeswitch-users] Error when building ESL for PHP Message-ID: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> Hi all, I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. It's throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. Any idea on how to fix this error? Thanks root:/usr/local/src/freeswitch/libs/esl# make root:/usr/local/src/freeswitch/libs/esl# make phpmod-install make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp: In function 'void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)': esl_wrap.cpp:869:65: error: invalid conversion from 'const char*' to 'char*' [-fpermissive] esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] . . . esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/8eeb9921/attachment.html From rgenthner at symplicity.com Fri Oct 11 21:24:32 2013 From: rgenthner at symplicity.com (Richard Genthner) Date: Fri, 11 Oct 2013 13:24:32 -0400 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> Message-ID: <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> Sam, Thats a known bug. diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile index dce8fd4..97674f8 100644 --- a/libs/esl/php/Makefile +++ b/libs/esl/php/Makefile @@ -17,6 +17,7 @@ all: ESL.so esl_wrap.cpp: swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp esl_wrap.o: esl_wrap.cpp $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp index 8c91f25..0389cc2 100644 --- a/libs/esl/php/esl_wrap.cpp +++ b/libs/esl/php/esl_wrap.cpp @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { swig_object_wrapper *value; void *p; int type; - char *type_name; + const char *type_name; value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); if ( flags && SWIG_POINTER_DISOWN ) { Thats the Patch for it. -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com On Oct 11, 2013, at 1:16 PM, Sam Montour wrote: > Hi all, > I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. > > libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. > > It?s throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. > > I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. > > Any idea on how to fix this error? > > Thanks > > root:/usr/local/src/freeswitch/libs/esl# make > root:/usr/local/src/freeswitch/libs/esl# make phpmod-install > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php > make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' > g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp: In function ?void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)?: > esl_wrap.cpp:869:65: error: invalid conversion from ?const char*? to ?char*? [-fpermissive] > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)?: > esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] > > . > . > . > esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] > make[1]: *** [esl_wrap.o] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' > make: *** [phpmod] Error 2 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/6cbe96a7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 496 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/6cbe96a7/attachment-0001.bin From mike at jerris.com Fri Oct 11 21:35:20 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Oct 2013 13:35:20 -0400 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> Message-ID: This patch should already be in tree. Ken Rice- Is this in a release build yet? On Oct 11, 2013, at 1:24 PM, Richard Genthner wrote: > Sam, > Thats a known bug. > > diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile > index dce8fd4..97674f8 100644 > --- a/libs/esl/php/Makefile > +++ b/libs/esl/php/Makefile > @@ -17,6 +17,7 @@ all: ESL.so > > esl_wrap.cpp: > swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i > + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp > > esl_wrap.o: esl_wrap.cpp > $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o > diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp > index 8c91f25..0389cc2 100644 > --- a/libs/esl/php/esl_wrap.cpp > +++ b/libs/esl/php/esl_wrap.cpp > @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { > swig_object_wrapper *value; > void *p; > int type; > - char *type_name; > + const char *type_name; > > value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); > if ( flags && SWIG_POINTER_DISOWN ) { > > Thats the Patch for it. > -- > Thanks, > > Richard Genthner > System Administrator > Symplicity > tel 703.351.0200 x 8051 > web www.symplicity.com > > On Oct 11, 2013, at 1:16 PM, Sam Montour wrote: > >> Hi all, >> I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. >> >> libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. >> >> It?s throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. >> >> I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. >> >> Any idea on how to fix this error? >> >> Thanks >> >> root:/usr/local/src/freeswitch/libs/esl# make >> root:/usr/local/src/freeswitch/libs/esl# make phpmod-install >> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php >> make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' >> g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o >> esl_wrap.cpp: In function ?void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)?: >> esl_wrap.cpp:869:65: error: invalid conversion from ?const char*? to ?char*? [-fpermissive] >> esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)?: >> esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] >> >> . >> . >> . >> esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] >> make[1]: *** [esl_wrap.o] Error 1 >> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' >> make: *** [phpmod] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/cf248629/attachment.html From smontour at verizon.net Fri Oct 11 21:48:44 2013 From: smontour at verizon.net (Sam Montour) Date: Fri, 11 Oct 2013 12:48:44 -0500 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> Message-ID: <00c801cec6aa$21929e80$64b7db80$@verizon.net> I am running FS 1.2.12. Is the patch included in FS 1.2.13? I can upgrade then. If not, where can I download the patch? Thanks. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, October 11, 2013 12:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error when building ESL for PHP This patch should already be in tree. Ken Rice- Is this in a release build yet? On Oct 11, 2013, at 1:24 PM, Richard Genthner wrote: Sam, Thats a known bug. diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile index dce8fd4..97674f8 100644 --- a/libs/esl/php/Makefile +++ b/libs/esl/php/Makefile @@ -17,6 +17,7 @@ all: ESL.so esl_wrap.cpp: swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp esl_wrap.o: esl_wrap.cpp $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp index 8c91f25..0389cc2 100644 --- a/libs/esl/php/esl_wrap.cpp +++ b/libs/esl/php/esl_wrap.cpp @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { swig_object_wrapper *value; void *p; int type; - char *type_name; + const char *type_name; value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); if ( flags && SWIG_POINTER_DISOWN ) { Thats the Patch for it. -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com On Oct 11, 2013, at 1:16 PM, Sam Montour wrote: Hi all, I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. It's throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. Any idea on how to fix this error? Thanks root:/usr/local/src/freeswitch/libs/esl# make root:/usr/local/src/freeswitch/libs/esl# make phpmod-install make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp: In function 'void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)': esl_wrap.cpp:869:65: error: invalid conversion from 'const char*' to 'char*' [-fpermissive] esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] . . . esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/a36153f5/attachment-0001.html From steveayre at gmail.com Fri Oct 11 21:57:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 11 Oct 2013 18:57:23 +0100 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: References: Message-ID: If you use uuid_kill and the channel exists you'll see +OK, if it isn't found you'll see an error. Similar errors probably occur with Apis such as uuid_setvar... Perhaps that will give you a way to tell them apart. The reason for this is likely a database one. Those commands run queries against the calls and channels tables in the core DB. FreeSWITCH updates these when channels start, change state and end. It may be the transactions to remove the entries at the end if the call, either with a logged error or silently. That'd make the db entry persist even though the call no longer does. If that's the cause it's worth investigating why. It could be load, db errors, or transaction locking where another connection is already using the row. The ODBC options in use might be worth checking. Also if FS is batching up updates to the DB that might conceivably be affecting it. On Wednesday, October 9, 2013, Peter Hartmann wrote: > Hi, > First, thanks for Freeswitch! I'm experiencing an issue where 'show > calls' returns several calls that aren't actually happening both > inbound and outbound. Has anyone seen this before? > > Rebooting the handset (Polycom IP 550) associated with that extension > has no effect so it seems in FS. > > freeswitch at internal> show calls > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch > d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 > 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com > ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 > 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 > 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com > ,CS_EXECUTE,unknown > ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 > 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 > 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension > 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound > Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension > 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, > bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 > 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 > 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com > ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 > 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com > ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 > 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com > ,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound > Call,1000,,,,,,,,,,,,,,,,,,, > > 9 total. > > > Running: > FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 > 2013-10-02 21:30:46Z) > > > Thanks much! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/c0ba2c4a/attachment.html From mike at jerris.com Fri Oct 11 21:59:47 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Oct 2013 13:59:47 -0400 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: <00c801cec6aa$21929e80$64b7db80$@verizon.net> References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> <00c801cec6aa$21929e80$64b7db80$@verizon.net> Message-ID: <571C5BA6-A5A2-48E2-BF61-3652F2A8EE43@jerris.com> I'll wait for ken to chime in, it might not be a release yet, you can just hand patch libs/esl/php/esl_wrap.cpp real quick and it will fix the issue On Oct 11, 2013, at 1:48 PM, Sam Montour wrote: > I am running FS 1.2.12. Is the patch included in FS 1.2.13? I can upgrade then. If not, where can I download the patch? Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: Friday, October 11, 2013 12:35 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Error when building ESL for PHP > > This patch should already be in tree. Ken Rice- Is this in a release build yet? > > On Oct 11, 2013, at 1:24 PM, Richard Genthner wrote: > > > Sam, > Thats a known bug. > > diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile > index dce8fd4..97674f8 100644 > --- a/libs/esl/php/Makefile > +++ b/libs/esl/php/Makefile > @@ -17,6 +17,7 @@ all: ESL.so > > esl_wrap.cpp: > swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i > + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp > > esl_wrap.o: esl_wrap.cpp > $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o > diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp > index 8c91f25..0389cc2 100644 > --- a/libs/esl/php/esl_wrap.cpp > +++ b/libs/esl/php/esl_wrap.cpp > @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { > swig_object_wrapper *value; > void *p; > int type; > - char *type_name; > + const char *type_name; > > value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); > if ( flags && SWIG_POINTER_DISOWN ) { > > Thats the Patch for it. > -- > Thanks, > > Richard Genthner > System Administrator > Symplicity > tel 703.351.0200 x 8051 > web www.symplicity.com > > On Oct 11, 2013, at 1:16 PM, Sam Montour wrote: > > > Hi all, > I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. > > libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. > > It?s throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. > > I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. > > Any idea on how to fix this error? > > Thanks > > root:/usr/local/src/freeswitch/libs/esl# make > root:/usr/local/src/freeswitch/libs/esl# make phpmod-install > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php > make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' > g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp: In function ?void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)?: > esl_wrap.cpp:869:65: error: invalid conversion from ?const char*? to ?char*? [-fpermissive] > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)?: > esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] > > . > . > . > esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] > make[1]: *** [esl_wrap.o] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' > make: *** [phpmod] Error 2 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/98eb9210/attachment-0001.html From victor.chukalovskiy at gmail.com Fri Oct 11 22:04:48 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Fri, 11 Oct 2013 14:04:48 -0400 Subject: [Freeswitch-users] mod_translate question In-Reply-To: References: <525708E5.4070706@gmail.com> <52581BA3.3040903@gmail.com> Message-ID: <52583DC0.7030402@gmail.com> I see! Wonder if anyone else can comment. Any way to set channel variable on per sip profile basis? On 13-10-11 12:08 PM, Andrew Cassidy wrote: > No idea, I only attempted the documentation by reading the code I'm > afraid. > > I have default_country set in vars.xml > > > On 11 October 2013 16:39, Victor Chukalovskiy > > > wrote: > > Thank you for the response! Hmm, Is there a way to set channel > variable per SIP profile??? > > I tried this: > > > > And this: > > > > But it never picks-up the param. Still default to US: > > 2013-10-11 11:33:26.251629 [DEBUG] mod_translate.c:291 using > default_country variable [US] for translate profile > > I understand "param" is not the same as channel variable. > > Thank you, > Victor > > On 13-10-11 10:42 AM, Andrew Cassidy wrote: >> At the moment the only way I can see is to set country in the >> sofia or gateway configuration to the profile name. >> >> >> >> On 10 October 2013 21:07, Victor Chukalovskiy >> > > wrote: >> >> Hi All, >> >> How do I specify translation profile when using this line in >> SIP profile? >> >> >> >> What I'm trying to achieve is to make sure call only goes >> through one translations profile. I don't want to rely on >> the order of translations profiles in translation.conf. So, I >> want to do something like this: >> >> >> >> in translate.conf.xml I have: >> >> >> >> >> >> >> >> >> >> > replace="+1$1"/> >> >> >> >> >> >> >> >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 >> *F >> *03300 100 961 >> *E >> *andrew at cassidywebservices.co.uk >> >> *W >> *www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/c94a38d6/attachment.html From smontour at verizon.net Fri Oct 11 22:09:46 2013 From: smontour at verizon.net (Sam Montour) Date: Fri, 11 Oct 2013 13:09:46 -0500 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: <571C5BA6-A5A2-48E2-BF61-3652F2A8EE43@jerris.com> References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> <00c801cec6aa$21929e80$64b7db80$@verizon.net> <571C5BA6-A5A2-48E2-BF61-3652F2A8EE43@jerris.com> Message-ID: <00df01cec6ad$11dcadc0$35960940$@verizon.net> I'd like to test the patch for now. Where can I download the patch file? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, October 11, 2013 1:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error when building ESL for PHP I'll wait for ken to chime in, it might not be a release yet, you can just hand patch libs/esl/php/esl_wrap.cpp real quick and it will fix the issue On Oct 11, 2013, at 1:48 PM, Sam Montour wrote: I am running FS 1.2.12. Is the patch included in FS 1.2.13? I can upgrade then. If not, where can I download the patch? Thanks. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, October 11, 2013 12:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error when building ESL for PHP This patch should already be in tree. Ken Rice- Is this in a release build yet? On Oct 11, 2013, at 1:24 PM, Richard Genthner < rgenthner at symplicity.com> wrote: Sam, Thats a known bug. diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile index dce8fd4..97674f8 100644 --- a/libs/esl/php/Makefile +++ b/libs/esl/php/Makefile @@ -17,6 +17,7 @@ all: ESL.so esl_wrap.cpp: swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp esl_wrap.o: esl_wrap.cpp $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp index 8c91f25..0389cc2 100644 --- a/libs/esl/php/esl_wrap.cpp +++ b/libs/esl/php/esl_wrap.cpp @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { swig_object_wrapper *value; void *p; int type; - char *type_name; + const char *type_name; value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); if ( flags && SWIG_POINTER_DISOWN ) { Thats the Patch for it. -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com On Oct 11, 2013, at 1:16 PM, Sam Montour < smontour at verizon.net> wrote: Hi all, I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. It's throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. Any idea on how to fix this error? Thanks root:/usr/local/src/freeswitch/libs/esl# make root:/usr/local/src/freeswitch/libs/esl# make phpmod-install make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp: In function 'void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)': esl_wrap.cpp:869:65: error: invalid conversion from 'const char*' to 'char*' [-fpermissive] esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] . . . esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com <> Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http:// lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/91540bbe/attachment-0001.html From mike at jerris.com Fri Oct 11 22:31:25 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Oct 2013 14:31:25 -0400 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: <00df01cec6ad$11dcadc0$35960940$@verizon.net> References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> <00c801cec6aa$21929e80$64b7db80$@verizon.net> <571C5BA6-A5A2-48E2-BF61-3652F2A8EE43@jerris.com> <00df01cec6ad$11dcadc0$35960940$@verizon.net> Message-ID: You can just do a checkout of the master or 1.2 branch directly from git. Mike On Oct 11, 2013, at 2:09 PM, Sam Montour wrote: > I?d like to test the patch for now. Where can I download the patch file? > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: Friday, October 11, 2013 1:00 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Error when building ESL for PHP > > I'll wait for ken to chime in, it might not be a release yet, you can just hand patch libs/esl/php/esl_wrap.cpp real quick and it will fix the issue > > On Oct 11, 2013, at 1:48 PM, Sam Montour wrote: > > > I am running FS 1.2.12. Is the patch included in FS 1.2.13? I can upgrade then. If not, where can I download the patch? Thanks. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: Friday, October 11, 2013 12:35 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Error when building ESL for PHP > > This patch should already be in tree. Ken Rice- Is this in a release build yet? > > On Oct 11, 2013, at 1:24 PM, Richard Genthner wrote: > > > > Sam, > Thats a known bug. > > diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile > index dce8fd4..97674f8 100644 > --- a/libs/esl/php/Makefile > +++ b/libs/esl/php/Makefile > @@ -17,6 +17,7 @@ all: ESL.so > > esl_wrap.cpp: > swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i > + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp > > esl_wrap.o: esl_wrap.cpp > $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o > diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp > index 8c91f25..0389cc2 100644 > --- a/libs/esl/php/esl_wrap.cpp > +++ b/libs/esl/php/esl_wrap.cpp > @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { > swig_object_wrapper *value; > void *p; > int type; > - char *type_name; > + const char *type_name; > > value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); > if ( flags && SWIG_POINTER_DISOWN ) { > > Thats the Patch for it. > -- > Thanks, > > Richard Genthner > System Administrator > Symplicity > tel 703.351.0200 x 8051 > web www.symplicity.com > > On Oct 11, 2013, at 1:16 PM, Sam Montour wrote: > > > > Hi all, > I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. > > libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. > > It?s throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. > > I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. > > Any idea on how to fix this error? > > Thanks > > root:/usr/local/src/freeswitch/libs/esl# make > root:/usr/local/src/freeswitch/libs/esl# make phpmod-install > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php > make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' > g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp: In function ?void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)?: > esl_wrap.cpp:869:65: error: invalid conversion from ?const char*? to ?char*? [-fpermissive] > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)?: > esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] > > . > . > . > esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] > make[1]: *** [esl_wrap.o] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' > make: *** [phpmod] Error 2 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/738c6326/attachment.html From smontour at verizon.net Fri Oct 11 23:21:36 2013 From: smontour at verizon.net (Sam Montour) Date: Fri, 11 Oct 2013 14:21:36 -0500 Subject: [Freeswitch-users] Error when building ESL for PHP In-Reply-To: References: <00ae01cec6a5$99d0e230$cd72a690$@verizon.net> <183BB1C4-F62C-472E-A2F4-684863E7C7EC@symplicity.com> <00c801cec6aa$21929e80$64b7db80$@verizon.net> <571C5BA6-A5A2-48E2-BF61-3652F2A8EE43@jerris.com> <00df01cec6ad$11dcadc0$35960940$@verizon.net> Message-ID: <000701cec6b7$1a8f3230$4fad9690$@verizon.net> Thanks Mike. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, October 11, 2013 1:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error when building ESL for PHP You can just do a checkout of the master or 1.2 branch directly from git. Mike On Oct 11, 2013, at 2:09 PM, Sam Montour wrote: I'd like to test the patch for now. Where can I download the patch file? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, October 11, 2013 1:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error when building ESL for PHP I'll wait for ken to chime in, it might not be a release yet, you can just hand patch libs/esl/php/esl_wrap.cpp real quick and it will fix the issue On Oct 11, 2013, at 1:48 PM, Sam Montour < smontour at verizon.net> wrote: I am running FS 1.2.12. Is the patch included in FS 1.2.13? I can upgrade then. If not, where can I download the patch? Thanks. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, October 11, 2013 12:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Error when building ESL for PHP This patch should already be in tree. Ken Rice- Is this in a release build yet? On Oct 11, 2013, at 1:24 PM, Richard Genthner < rgenthner at symplicity.com> wrote: Sam, Thats a known bug. diff --git a/libs/esl/php/Makefile b/libs/esl/php/Makefile index dce8fd4..97674f8 100644 --- a/libs/esl/php/Makefile +++ b/libs/esl/php/Makefile @@ -17,6 +17,7 @@ all: ESL.so esl_wrap.cpp: swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp ../ESL.i + sed -e 's/ char \*type_name;/ const char \*type_name;/' -i esl_wrap.cpp esl_wrap.o: esl_wrap.cpp $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) $(WRAP_GCC_WARNING_SILENCE) -c esl_wrap.cpp -o esl_wrap.o diff --git a/libs/esl/php/esl_wrap.cpp b/libs/esl/php/esl_wrap.cpp index 8c91f25..0389cc2 100644 --- a/libs/esl/php/esl_wrap.cpp +++ b/libs/esl/php/esl_wrap.cpp @@ -857,7 +857,7 @@ SWIG_ZTS_ConvertResourcePtr(zval *z, swig_type_info *ty, int flags TSRMLS_DC) { swig_object_wrapper *value; void *p; int type; - char *type_name; + const char *type_name; value = (swig_object_wrapper *) zend_list_find(z->value.lval, &type); if ( flags && SWIG_POINTER_DISOWN ) { Thats the Patch for it. -- Thanks, Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com On Oct 11, 2013, at 1:16 PM, Sam Montour < smontour at verizon.net> wrote: Hi all, I am trying to build ESL library for PHP but received the error below. I checked all pre-requisite libraries according to ESL wiki page and seem to have them all. libxml2-dev libpcre3-dev libcurl4-openssl-dev libgmp3-dev libaspell-dev python-dev php5-dev libonig-dev libqdbm-dev libedit-dev libdb-dev. It's throwing an error for an invalid conversion from const char to char in esl_wrapp.cpp. I am running FreeSWITCH Version 1.2.12+git~20130816T225403Z~8566ffa82a on Linux Mint 14. Any idea on how to fix this error? Thanks root:/usr/local/src/freeswitch/libs/esl# make root:/usr/local/src/freeswitch/libs/esl# make phpmod-install make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp: In function 'void* SWIG_ZTS_ConvertResourcePtr(zval*, swig_type_info*, int)': esl_wrap.cpp:869:65: error: invalid conversion from 'const char*' to 'char*' [-fpermissive] esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] . . . esl_wrap.cpp:2757:46: warning: format not a string literal and no format arguments [-Wformat-security] make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/537140b9/attachment-0001.html From mario_fs at mgtech.com Fri Oct 11 23:55:20 2013 From: mario_fs at mgtech.com (Mario G) Date: Fri, 11 Oct 2013 12:55:20 -0700 Subject: [Freeswitch-users] Playing announcement file before MOH In-Reply-To: References: Message-ID: <9BC09009-A287-4554-87F7-FE6B39F52774@mgtech.com> I am doing something similar using local_stream to replace the ring the caller hears. In fact I gave Anthony some cash to add a timeout option which I documented in the wiki: http://wiki.freeswitch.org/wiki/Mod_local_stream#Execution_Parameters Here is what callers hear (Using timeout I can limit how long music plays between voice, I use LUA to inject the persons name into the message): 1. Hello Jim, please listen to some music while the phones ring. 2. Music. 3. Hold on Jim, while we try our cell phones. 4. Music. 5. Sorry Jim, please leave a message. Message is then emailed to cell phones. Hope this helps a little, Mario G On Oct 11, 2013, at 6:32 AM, Cal Leeming [Simplicity Media Ltd] wrote: > No problem, glad we could help. > > Once you have managed to achieve what you are trying to do, perhaps you could update this thread and share your approach as it will help others in the future. > > Cal > > > On Thu, Oct 10, 2013 at 5:54 AM, Nandy Dagondon wrote: > Hi Cal, The "please_hold" vanilla dialplan in features.xml is what I initially. The tips you shared - someday I can use them. Thanks a lot! /Nandy > > > On Thu, Oct 10, 2013 at 11:46 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Have a look into some of these variables/options below, use google+wiki; > > uuid_displace - allows you to inject audio into a channel > temp_hold_music - allows you specify music during transfer hold > campon - places call into parking when on hold (not 100% on this, needs clarification) > > You could also modify the MOH file to play the message you require at the desired intervals. Realistically I think this would be the best way, as this will give you much more control over the quality delivered (fading, volume, etc). > > Hope this helps > > Cal > > > > On Thu, Oct 10, 2013 at 2:18 AM, Nandy Dagondon wrote: > Hello everyone, > > I like to play an announcement file before playing music-on-hold. Is this possible? Any hint? > > Tks, > Nandy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/3577771a/attachment.html From mishehu at freeswitch.org Sat Oct 12 02:22:15 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Fri, 11 Oct 2013 17:22:15 -0500 Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: <52587A17.1050009@freeswitch.org> Also note that at least up until one generation ago on Atom - and possibly still true for the current generation - the processors are all in-order processing. Something like a Via Nano X2/X4 is going to be out-of-order processing, which *might* give you more bang for your buck. I have a Via Nano X2 here but all I've done with it so far was some testing of OpenSIPS on it, not Freeswitch yet. In general knowing the specs necessary for a given build-out depends highly on your usage scenario. There's no simple equation to plug in. -Yossi On 10/10/2013 04:12 PM, Guillermo Ruiz Camauer wrote: > You might want to specify the codecs for transcoding since they are > not all the same and also the ATOM processor type, as there are many > generations now. > > Guillermo > > > On Thu, Oct 10, 2013 at 5:50 PM, Stanislav Sinyagin > > wrote: > > hi, > > can anyone share some useful experience with Intel Atom CPU? If > used with FreeSWITCH, > > -- what's the maximum number of concurrent transcoding calls > without service degradation? > -- maximum CPS without service degradation? > -- maximum size of a conference call (with or without transcoding) > > There's a choice of fanless appliances with 1.6 GHz Atom > processors, and I wonder how well they can perform as PBX. > > thanks > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Guillermo Ruiz Camauer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/5d8ec0d7/attachment.html From karl at xtronics.com Sat Oct 12 02:39:00 2013 From: karl at xtronics.com (Karl Schmidt) Date: Fri, 11 Oct 2013 17:39:00 -0500 Subject: [Freeswitch-users] recommendations for Wifi SIP phones? In-Reply-To: <3B6185DF-4C4D-45B6-88CB-429FBCBFB1FF@jerris.com> References: <523B8595.4000502@mst.edu> <523B92F1.2070407@xtronics.com> <5244B020.1080504@xtronics.com> <5257601E.2010002@xtronics.com> <3B6185DF-4C4D-45B6-88CB-429FBCBFB1FF@jerris.com> Message-ID: <52587E04.2080906@xtronics.com> On 10/11/2013 09:02 AM, Michael Jerris wrote: > Audio issues on wifi have nothing to do with the signaling protocol, they have everything to do > with the codecs and other media features. Not sure what you mean - it appears to mostly have to do with the nature of wifi - today's wifi only serves one client at a time - thus there are gaps that best case could be covered with large buffers and delay. I did more tests just today with a Debian desk top and linphone as the client - works fine if the host is the only one connected. If you have a second host moving a data stream, it gets bad pretty fast. The closer the AP and host the better it works as there is more bandwidth - but I could make things fail even at close range. What I'm looking at is the ability for skype and Google voice to work where SIP does not. I have a lot more tests to run. I've bumped into this claim several times. ( I did find confounding problems if zrtp was turned on with a client without support - this is even on 1000BaseT not wifi related ) I'm also reading that CISCO has APs that make SIP work better - might just be marketing hype - but could be careful sharing of bandwidth and QOS. > My biggest issue with wifi based sip phones is the > battery life sucks. Even when the phone is used as a phone it is a problem - wifi tends to burn even more power. I've wanted to make it work with something in the 7" format that would have more battery. Newer chipsets can use a lot less power if the drivers are right. Don't bother w/ IAX2 its a solution looking for a problem. That is what I've heard - but I do see some products supporting it - makes me think there is a reason. ( my searches did not find happy freeswitch users using IAX2 ) -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 In a business deal, both parties enter for their own gain. It is only when dealing with the government, because of the threat of physical force, that we enter in deals that are not in our interest. kps -------------------------------------------------------------------------------- From grcamauer at gmail.com Sat Oct 12 03:15:39 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 11 Oct 2013 20:15:39 -0300 Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: <52587A17.1050009@freeswitch.org> References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52587A17.1050009@freeswitch.org> Message-ID: The newest ATOMs have out-of-order execution, and even come in 4 and 8 core versions. Different amounts of cache also. Wildly varying TDPs also, from 4W to 20W, On Fri, Oct 11, 2013 at 7:22 PM, I put the Who? in Mishehu < mishehu at freeswitch.org> wrote: > Also note that at least up until one generation ago on Atom - and > possibly still true for the current generation - the processors are all > in-order processing. Something like a Via Nano X2/X4 is going to be > out-of-order processing, which *might* give you more bang for your buck. I > have a Via Nano X2 here but all I've done with it so far was some testing > of OpenSIPS on it, not Freeswitch yet. > > In general knowing the specs necessary for a given build-out depends > highly on your usage scenario. There's no simple equation to plug in. > > -Yossi > > > On 10/10/2013 04:12 PM, Guillermo Ruiz Camauer wrote: > > You might want to specify the codecs for transcoding since they are not > all the same and also the ATOM processor type, as there are many > generations now. > > Guillermo > > > On Thu, Oct 10, 2013 at 5:50 PM, Stanislav Sinyagin wrote: > >> hi, >> >> can anyone share some useful experience with Intel Atom CPU? If used with >> FreeSWITCH, >> >> -- what's the maximum number of concurrent transcoding calls without >> service degradation? >> -- maximum CPS without service degradation? >> -- maximum size of a conference call (with or without transcoding) >> >> There's a choice of fanless appliances with 1.6 GHz Atom processors, and >> I wonder how well they can perform as PBX. >> >> thanks >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/07c8023a/attachment.html From kworm at sofnet.com Sat Oct 12 03:57:37 2013 From: kworm at sofnet.com (Kevin Wormington) Date: Fri, 11 Oct 2013 18:57:37 -0500 Subject: [Freeswitch-users] recommendations for Wifi SIP phones? In-Reply-To: <52587E04.2080906@xtronics.com> References: <523B8595.4000502@mst.edu> <523B92F1.2070407@xtronics.com> <5244B020.1080504@xtronics.com> <5257601E.2010002@xtronics.com> <3B6185DF-4C4D-45B6-88CB-429FBCBFB1FF@jerris.com> <52587E04.2080906@xtronics.com> Message-ID: <52589071.2000908@sofnet.com> Karl, I think what Michael means is that SIP is a signaling protocol and would having nothing to do with audio issues. If google voice or skype work on a wifi link and SIP does not then it is most likely due to the choice of audio codec. Google voice is probably using something like g.729 or speex/opus whereas if you haven't made changes to a default SIP install you may be using G.711 or G.722, etc. My experience has been that wifi on good APs with the appropriate codec is just barely ok for business users. I think most people that are serious about it in a business environment are using DECT. Kevin On 10/11/2013 05:39 PM, Karl Schmidt wrote: > On 10/11/2013 09:02 AM, Michael Jerris wrote: >> Audio issues on wifi have nothing to do with the signaling protocol, they have everything to do >> with the codecs and other media features. > > Not sure what you mean - it appears to mostly have to do with the nature of wifi - today's wifi only > serves one client at a time - thus there are gaps that best case could be covered with large buffers > and delay. I did more tests just today with a Debian desk top and linphone as the client - works > fine if the host is the only one connected. If you have a second host moving a data stream, it gets > bad pretty fast. The closer the AP and host the better it works as there is more bandwidth - but I > could make things fail even at close range. > > What I'm looking at is the ability for skype and Google voice to work where SIP does not. I have a > lot more tests to run. I've bumped into this claim several times. > > ( I did find confounding problems if zrtp was turned on with a client without support - this is even > on 1000BaseT not wifi related ) > > I'm also reading that CISCO has APs that make SIP work better - might just be marketing hype - but > could be careful sharing of bandwidth and QOS. > > >> My biggest issue with wifi based sip phones is the >> battery life sucks. > > Even when the phone is used as a phone it is a problem - wifi tends to burn even more power. I've > wanted to make it work with something in the 7" format that would have more battery. Newer chipsets > can use a lot less power if the drivers are right. > > > Don't bother w/ IAX2 its a solution looking for a problem. > > That is what I've heard - but I do see some products supporting it - makes me think there is a > reason. ( my searches did not find happy freeswitch users using IAX2 ) > > > -------------------------------------------------------------------------------- > Karl Schmidt EMail Karl at xtronics.com > Transtronics, Inc. WEB http://secure.transtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 FAX (785) 841-0434 > > In a business deal, both parties enter for their own gain. > It is only when dealing with the government, because of the > threat of physical force, that we enter in deals that are not in our interest. > kps > -------------------------------------------------------------------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ssinyagin at yahoo.com Sat Oct 12 07:30:04 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Fri, 11 Oct 2013 20:30:04 -0700 (PDT) Subject: [Freeswitch-users] Intel Atom performance In-Reply-To: References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52587A17.1050009@freeswitch.org> Message-ID: <1381548604.88448.YahooMailNeo@web126204.mail.ne1.yahoo.com> the ones I've got for testing are about 3 years old. 20 concurrent calls with G722/PCMA transcoding run quite OK. I will write a detailed test report within the weekend. ________________________________ From: Guillermo Ruiz Camauer To: FreeSWITCH Users Help Sent: Saturday, October 12, 2013 1:15 AM Subject: Re: [Freeswitch-users] Intel Atom performance The newest ATOMs have out-of-order execution, and even come in 4 and 8 core versions. ?Different amounts of cache also. ?Wildly varying TDPs also, from 4W to 20W, On Fri, Oct 11, 2013 at 7:22 PM, I put the Who? in Mishehu wrote: Also note that at least up until one generation ago on Atom - and possibly still true for the current generation - the processors are all in-order processing.? Something like a Via Nano X2/X4 is going to be out-of-order processing, which *might* give you more bang for your buck.? I have a Via Nano X2 here but all I've done with it so far was some testing of OpenSIPS on it, not Freeswitch yet. > >In general knowing the specs necessary for a given build-out depends highly on your usage scenario.? There's no simple equation to plug in. > >-Yossi > > >On 10/10/2013 04:12 PM, Guillermo Ruiz Camauer wrote: > >You might want to specify the codecs for transcoding since they are not all the same and also the ATOM processor type, as there are many generations now. >> >> >>Guillermo >> >> >> >>On Thu, Oct 10, 2013 at 5:50 PM, Stanislav Sinyagin wrote: >> >>hi, >>> >>>can anyone share some useful experience with Intel Atom CPU? If used with FreeSWITCH, >>> >>>-- what's the maximum number of concurrent transcoding calls without service degradation? >>>-- maximum CPS without service degradation? >>>-- maximum size of a conference call (with or without transcoding) >>> >>>There's a choice of fanless appliances with 1.6 GHz Atom processors, and I wonder how well they can perform as PBX. >>> >>>thanks >>> >>> >>> >>> >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >> >> >> >> -- >>Guillermo Ruiz Camauer >> >> >> >>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Guillermo Ruiz Camauer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131011/e426261b/attachment-0001.html From daemonserj at gmail.com Sat Oct 12 07:03:22 2013 From: daemonserj at gmail.com (daemonserj) Date: Sat, 12 Oct 2013 10:03:22 +0700 Subject: [Freeswitch-users] *_after_bridge not works In-Reply-To: References: Message-ID: <1418065956.20131012100322@gmail.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131012/6000b044/attachment.html From krice at freeswitch.org Sat Oct 12 09:41:28 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 12 Oct 2013 00:41:28 -0500 Subject: [Freeswitch-users] *_after_bridge not works In-Reply-To: <1418065956.20131012100322@gmail.com> Message-ID: You are setting them on the A leg probably... If you want the B leg to follow those you need to set it there On 10/11/13 10:03 PM, "daemonserj" wrote: > Hello All > > Simple question: Why exec_after_bridge , transfer_after_bridge not works if > leg A terminates the call ? > > Thank you -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131012/0769a985/attachment.html From james.mortensen at synclio.com Sat Oct 12 23:44:34 2013 From: james.mortensen at synclio.com (James Mortensen) Date: Sat, 12 Oct 2013 12:44:34 -0700 Subject: [Freeswitch-users] Sending SAVPF INVITE to Opensips In-Reply-To: References: Message-ID: I wanted to follow up on Kristian's question about OpenSIPS and WebSocket support. According to what I've read, OpenSIPS only supports WS/WSS Via header parsing but doesn't actually have a WebSocket module to accept WebSocket connections. According to the OpenSIPS documentation as well as the release notes, one must front-end OpenSIPS with OverSIP. http://www.opensips.org/Documentation/Tutorials-WebSocket Hope this helps! James On Sat, Sep 28, 2013 at 4:22 PM, Kristian Kielhofner wrote: > I'm glad it worked for you. > > Are you aware that Opensips supports SIP over websockets and secure > websockets natively? > > > On Saturday, September 28, 2013, James Mortensen wrote: > >> Hi Kristian, >> >> You're right, what I want is a WebRTC SDP, which FreeSWITCH is clearly >> capable of generating as it does this with the default configuration by >> default if port 5066 is enabled for the ws-binding parameter. >> >> To answer your question, OverSIP sits in front of Opensips between the >> Chrome client and Opensips. Chrome can only register with a SIP endpoint >> using websockets, so OverSIP listens on a TCP/WS port and passes the SIP >> messages to Opensips. >> >> BANDWIDTH.com --udp--> FreeSWITCH -udp--> Opensips -udp--> OverSIP >> --ws--> Chrome WebRTC client. >> >> It looks like what I was missing was the > data="media_webrtc=true"/> and using application export instead of set. I >> was able to get an inbound call routed to my Chrome client, answer it, and >> experience two way audio. >> >> >> Thanks again! :) >> >> >> James >> >> >> >> On Sat, Sep 28, 2013 at 12:46 PM, Kristian Kielhofner wrote: >> >> James, >> >> There's quite a bit of detail omitted here but a few points: >> >> - avpf=yes on your gateway definition isn't doing anything. >> - Have you tried exporting media_webrtc=true before bridging back to >> Opensips? >> - Try exporting the actual variables, not including variable_ : >> >> >> >> >> >> >> >> > data="sofia/external/ws-Opensips/11234 at 54.X.X.75"/> >> >> >> >> - While you do want an "SAVPF INVITE" you really want a WebRTC SDP, >> which includes much much more than just SAVPF. >> - You'll probably want to do all other sorts of codec manipulation, >> fixups, etc when bridging between typical SIP endpoints and WebRTC >> endpoints. Look into late negotiation and every codec >> variable/setting you can find. >> - You may want to re-consider the interaction and authentication >> between OpenSIPS and FreeSWITCH. >> >> Also, what are you using OverSIP for in this scenario? >> >> >> On Fri, Sep 27, 2013 at 8:26 PM, James Mortensen >> wrote: >> > Hello, >> > >> > I have a bandwidth.com number pointed to opensips, and a WebRTC peer >> > registered with Opensips. I'm trying to dial the 10 digit number from a >> > cell phone and connect the call through FreeSWITCH to the Chrome WebRTC >> > client. >> > >> > >> > I defined opensips as a gateway, in the external profile: >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > In the public dialplan context, I added in a condition to catch the >> INVITE >> > coming in from opensips and pass it to a context I've called >> > "default-inbound". See the second condition: >> > >> > >> > > > break="never"> >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > Then, in the default-inbound context, I match the dialed number, answer >> the >> > call leg from the PSTN, and then try to transfer back through opensips >> to >> > oversip and to Chrome. The problem is that I either end up sending >> back AVP >> > INVITES, or Opensips refuses to authenticate the user. >> > >> > References: <525BE3E4.9070201@digitalmail.com> Message-ID: Try uuid_dump -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Alex Lake Gesendet: Montag, 14. Oktober 2013 14:30 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] Inspecting sessions from fs_cli Is there a way I can look at all session variables of a live call from the fs_cli prompt? eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" cheers, Alex _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From alex at digitalmail.com Mon Oct 14 16:58:53 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 14 Oct 2013 13:58:53 +0100 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <525BE3E4.9070201@digitalmail.com> References: <525BE3E4.9070201@digitalmail.com> Message-ID: <525BEA8D.7020003@digitalmail.com> BTW, it's channel variables I mean, of course ;-) > Is there a way I can look at all session variables of a live call from > the fs_cli prompt? > > eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" > > cheers, > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Mon Oct 14 17:00:07 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 14 Oct 2013 13:00:07 +0000 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <525BE3E4.9070201@digitalmail.com> References: <525BE3E4.9070201@digitalmail.com> Message-ID: <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump Or "info" app from within the dialplan. -Avi On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake wrote: > Is there a way I can look at all session variables of a live call from > the fs_cli prompt? > > eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" > > cheers, > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/c47152d5/attachment.html From alex at digitalmail.com Mon Oct 14 17:15:31 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 14 Oct 2013 14:15:31 +0100 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> References: <525BE3E4.9070201@digitalmail.com> <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> Message-ID: <525BEE73.20804@digitalmail.com> Ah thanks (and to Gerald as well). Unfortunately the information I was really looking for doesn't seem to be in there. I was after current (live) values for the rtp_audio_ variables. Well, actually, I was after an indication of line quality. Any suggestions? Rgds, Alex > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump > Or "info" app from within the dialplan. > -Avi > > On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake > wrote: > > Is there a way I can look at all session variables of a live call from > the fs_cli prompt? > > eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" > > cheers, > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/c4fcbed8/attachment.html From vbvbrj at gmail.com Mon Oct 14 18:28:16 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 17:28:16 +0300 Subject: [Freeswitch-users] What kind of attack is this? Message-ID: <525BFF80.8020805@gmail.com> Hello. recently I see ddos on one interface and FS module callcenter is working irregularly. tcpdump shows this: 17:17:42.410306 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 364) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 E..l.. at .0...2.% MY.".....XW.REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3646224729;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 360911671 Max-Forwards: 70 17:17:42.415504 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 365) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 E..m.. at .0...2.% MY.".....Y,.REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-1538287390;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 3912185912 Max-Forwards: 70 17:17:42.420997 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 365) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 E..m.. at .0...2.% MY.".....Y7.REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3729326239;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 2188845586 Max-Forwards: 70 17:17:42.425886 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 365) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 E..m.. at .0...2.% MY.".....Y3.REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-2208974380;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 4149361432 Max-Forwards: 70 17:17:42.431126 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 364) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 E..l.. at .0...2.% MY.".....X..REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-725880732;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 1466795680 Max-Forwards: 70 17:17:42.436476 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 365) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 E..m.. at .0...2.% MY.".....Y6.REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3259665948;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 3328716097 Max-Forwards: 70 17:17:42.441541 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto UDP (17), length 364) 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 E..l.. at .0...2.% MY.".....XT.REGISTER sip:A.B.C.D SIP/2.0 Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-2487219966;rport Content-Length: 0 From: "6796" Accept: application/sdp User-Agent: friendly-scanner To: "6796" Contact: sip:123 at 1.1.1.1 CSeq: 1 REGISTER Call-ID: 684380132 Max-Forwards: 70 In iptables I have this: 1637 597K DROP all -- * * 50.30.37.10 0.0.0.0/0 0 0 DROP all -- * * 62.75.212.215 0.0.0.0/0 So packets form that IP are not dropped. How is that? Does FS has a bag? -- Mimiko desu. From martin.cmelik at gmail.com Mon Oct 14 18:47:36 2013 From: martin.cmelik at gmail.com (=?UTF-8?B?TWFydGluIMSMbWVsw61r?=) Date: Mon, 14 Oct 2013 16:47:36 +0200 Subject: [Freeswitch-users] Configuration with dynamic public IP Message-ID: Hi all, I have very easy (easy for you :]) question. Let me describe in short the setup. FreeSWITCH installed on local network computer. No extra functions are required, I will just install FS, create users in directory and then need capability to make call with users inside same internal network, as well as with clients on Internet which will be logged/registered on same PBX. External IP is changing in time, but my router have UPnP capability and I also redirect all incoming traffic to this PBX. Users place calls from mobile clients (Linphone). I have two issues: 1] Internally everything works fine (from internal to internal). I have problem with clients registered from Internet, who cant call anyone else (mailbox, IVR, echo test, ... is OK). Error message in FS states USER_NOT_REGISTERED and "Client busy" on side of caller. I try couple of things/setup in vars.xml but still nothing helped me (external_rtp_ip, external_sip_ip). Does anyone have working example? 2] Which is in relation with problem "1". External IP of my router is changing with almost every restart of my router. Is there some recommendation how FS can check, from time to time, current IP and change setup accordingly? I can also write script which will check current IP, rewrite config and restart FS if needed, but if something already exist, why reinvent the wheel :] If this is described already (I was unsuccessful on google), please just send me a link. Thank you very much! ? Martin ?mel?k From krice at freeswitch.org Mon Oct 14 18:54:28 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Oct 2013 09:54:28 -0500 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: <525BFF80.8020805@gmail.com> Message-ID: This is sipvicious, its a brute force scanner... See http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with FreeSWITCH to defeat this attack On 10/14/13 9:28 AM, "Mimiko" wrote: > Hello. > > recently I see ddos on one interface and FS module callcenter is working > irregularly. tcpdump shows this: > > 17:17:42.410306 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 364) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 > E..l.. at .0...2.% > MY.".....XW.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3646224729;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 360911671 > Max-Forwards: 70 > > > 17:17:42.415504 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y,.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-1538287390;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 3912185912 > Max-Forwards: 70 > > > 17:17:42.420997 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y7.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3729326239;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 2188845586 > Max-Forwards: 70 > > > 17:17:42.425886 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y3.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-2208974380;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 4149361432 > Max-Forwards: 70 > > > 17:17:42.431126 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 364) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 > E..l.. at .0...2.% > MY.".....X..REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-725880732;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 1466795680 > Max-Forwards: 70 > > > 17:17:42.436476 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y6.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3259665948;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 3328716097 > Max-Forwards: 70 > > > 17:17:42.441541 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 364) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 > E..l.. at .0...2.% > MY.".....XT.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-2487219966;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 684380132 > Max-Forwards: 70 > > > In iptables I have this: > 1637 597K DROP all -- * * 50.30.37.10 > 0.0.0.0/0 > 0 0 DROP all -- * * 62.75.212.215 > 0.0.0.0/0 > > So packets form that IP are not dropped. How is that? Does FS has a bag? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From vbvbrj at gmail.com Mon Oct 14 18:57:48 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 17:57:48 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: Message-ID: <525C066C.20102@gmail.com> On 14.10.2013 17:54, Ken Rice wrote: > This is sipvicious, its a brute force scanner... See > http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with > FreeSWITCH to defeat this attack Ken thank you. I am planning to set up Fail2ban. But for now need to drop any packet from offending IP. -- Mimiko desu. From jleung at v10networks.ca Mon Oct 14 19:00:03 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 14 Oct 2013 23:00:03 +0800 Subject: [Freeswitch-users] What kind of attack is this? Message-ID: <001f01cec8ee$155683cc$7c07000a@smb.curriegrad2004.ca> You can also use iptables to drop udp packets with the string friendly-scanner destined to port 5060 Ken Rice wrote: This is sipvicious, its a brute force scanner... See http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with FreeSWITCH to defeat this attack On 10/14/13 9:28 AM, "Mimiko" wrote: > Hello. > > recently I see ddos on one interface and FS module callcenter is working > irregularly. tcpdump shows this: > > 17:17:42.410306 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 364) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 > E..l.. at .0...2.% > MY.".....XW.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3646224729;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 360911671 > Max-Forwards: 70 > > > 17:17:42.415504 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y,.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-1538287390;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 3912185912 > Max-Forwards: 70 > > > 17:17:42.420997 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y7.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3729326239;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 2188845586 > Max-Forwards: 70 > > > 17:17:42.425886 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y3.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-2208974380;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 4149361432 > Max-Forwards: 70 > > > 17:17:42.431126 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 364) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 > E..l.. at .0...2.% > MY.".....X..REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-725880732;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 1466795680 > Max-Forwards: 70 > > > 17:17:42.436476 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 365) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 337 > E..m.. at .0...2.% > MY.".....Y6.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-3259665948;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 3328716097 > Max-Forwards: 70 > > > 17:17:42.441541 IP (tos 0x0, ttl 48, id 0, offset 0, flags [DF], proto > UDP (17), length 364) > 50.30.37.10.5064 > A.B.C.D.5060: [udp sum ok] UDP, length 336 > E..l.. at .0...2.% > MY.".....XT.REGISTER sip:A.B.C.D SIP/2.0 > Via: SIP/2.0/UDP 62.75.212.215:5064;branch=z9hG4bK-2487219966;rport > Content-Length: 0 > From: "6796" > Accept: application/sdp > User-Agent: friendly-scanner > To: "6796" > Contact: sip:123 at 1.1.1.1 > CSeq: 1 REGISTER > Call-ID: 684380132 > Max-Forwards: 70 > > > In iptables I have this: > 1637 597K DROP all -- * * 50.30.37.10 > 0.0.0.0/0 > 0 0 DROP all -- * * 62.75.212.215 > 0.0.0.0/0 > > So packets form that IP are not dropped. How is that? Does FS has a bag? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Mon Oct 14 19:05:53 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Oct 2013 10:05:53 -0500 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: <525C066C.20102@gmail.com> Message-ID: Google around a little... You can drop any packet with friendly-scanner in it with iptables as a way to defeat the attack also On 10/14/13 9:57 AM, "Mimiko" wrote: > On 14.10.2013 17:54, Ken Rice wrote: >> This is sipvicious, its a brute force scanner... See >> http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with >> FreeSWITCH to defeat this attack > > Ken thank you. I am planning to set up Fail2ban. But for now need to > drop any packet from offending IP. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From jleung at v10networks.ca Mon Oct 14 19:10:41 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 14 Oct 2013 23:10:41 +0800 Subject: [Freeswitch-users] What kind of attack is this? Message-ID: <002001cec8ef$9fa589ff$7c07000a@smb.curriegrad2004.ca> Also lately there has been scanners spoofing Cisco IOS devices. Stay safe people. Ken Rice wrote: Google around a little... You can drop any packet with friendly-scanner in it with iptables as a way to defeat the attack also On 10/14/13 9:57 AM, "Mimiko" wrote: > On 14.10.2013 17:54, Ken Rice wrote: >> This is sipvicious, its a brute force scanner... See >> http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with >> FreeSWITCH to defeat this attack > > Ken thank you. I am planning to set up Fail2ban. But for now need to > drop any packet from offending IP. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From vbvbrj at gmail.com Mon Oct 14 19:15:39 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 18:15:39 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: Message-ID: <525C0A9B.1020502@gmail.com> On 14.10.2013 18:05, Ken Rice wrote: > You can drop any packet with friendly-scanner in > it with iptables as a way to defeat the attack also I use this command: iptables -I INPUT -j DROP -p udp ?dport 5060 -m string ?string ?friendly-scanner? ?algo bm and my iptables looks like this: Chain INPUT (policy DROP 27 packets, 8596 bytes) pkts bytes target prot opt in out source destination 2482 905K DROP udp -- * * 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 STRING match "friendly-scanner" ALGO name bm TO 65535 0 0 DROP all -- * * 50.7.251.123 0.0.0.0/0 0 0 DROP all -- * * 199.241.187.214 0.0.0.0/0 0 0 DROP all -- * * 85.25.199.142 0.0.0.0/0 0 0 DROP all -- * * 50.30.37.10 0.0.0.0/0 0 0 DROP all -- * * 62.75.212.215 0.0.0.0/0 But this does not help. -- Mimiko desu. From vbvbrj at gmail.com Mon Oct 14 19:15:39 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 18:15:39 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: Message-ID: <525C0A9B.1020502@gmail.com> On 14.10.2013 18:05, Ken Rice wrote: > You can drop any packet with friendly-scanner in > it with iptables as a way to defeat the attack also I use this command: iptables -I INPUT -j DROP -p udp ?dport 5060 -m string ?string ?friendly-scanner? ?algo bm and my iptables looks like this: Chain INPUT (policy DROP 27 packets, 8596 bytes) pkts bytes target prot opt in out source destination 2482 905K DROP udp -- * * 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 STRING match "friendly-scanner" ALGO name bm TO 65535 0 0 DROP all -- * * 50.7.251.123 0.0.0.0/0 0 0 DROP all -- * * 199.241.187.214 0.0.0.0/0 0 0 DROP all -- * * 85.25.199.142 0.0.0.0/0 0 0 DROP all -- * * 50.30.37.10 0.0.0.0/0 0 0 DROP all -- * * 62.75.212.215 0.0.0.0/0 But this does not help. -- Mimiko desu. From jleung at v10networks.ca Mon Oct 14 19:18:38 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 14 Oct 2013 23:18:38 +0800 Subject: [Freeswitch-users] What kind of attack is this? Message-ID: <002101cec8f0$ad1a2866$7c07000a@smb.curriegrad2004.ca> It's working as intended Mimiko wrote: On 14.10.2013 18:05, Ken Rice wrote: > You can drop any packet with friendly-scanner in > it with iptables as a way to defeat the attack also I use this command: iptables -I INPUT -j DROP -p udp ?dport 5060 -m string ?string ?friendly-scanner? ?algo bm and my iptables looks like this: Chain INPUT (policy DROP 27 packets, 8596 bytes) pkts bytes target prot opt in out source destination 2482 905K DROP udp -- * * 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 STRING match "friendly-scanner" ALGO name bm TO 65535 0 0 DROP all -- * * 50.7.251.123 0.0.0.0/0 0 0 DROP all -- * * 199.241.187.214 0.0.0.0/0 0 0 DROP all -- * * 85.25.199.142 0.0.0.0/0 0 0 DROP all -- * * 50.30.37.10 0.0.0.0/0 0 0 DROP all -- * * 62.75.212.215 0.0.0.0/0 But this does not help. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Mon Oct 14 19:31:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Oct 2013 08:31:38 -0700 Subject: [Freeswitch-users] lua session question In-Reply-To: References: Message-ID: Do you mean the default dialplan context? If so, why does that matter for an outbound channel that you've explicitly created? Perhaps you could expound a little on what you're doing here. -MC On Fri, Oct 11, 2013 at 9:26 AM, Ira Tessler wrote: > I have a lua script that I am creating a new session: > > legA = freeswitch.Session(dialA); > > Where dial is: > > {dtmf_type=none,originate_timeout=60,instant_ringback=true,origination_caller_id_name=xxxxxxxxxx,origination_caller_id_number=xxxxxxxxxx,followme_callid=e9886a3f-58c5-456a-949a-b5bd89b11b38}sofia/gateway/sr/xxxxxxxxxx|sofia/gateway/voipinnovations.com/xxxxxxxxxx > > When the session gets created, it is in the "default" context. How can I > set the context for the channel to be created in? > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/1e672ec6/attachment.html From wstephen80 at gmail.com Mon Oct 14 19:53:37 2013 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 14 Oct 2013 17:53:37 +0200 Subject: [Freeswitch-users] Performance issue Message-ID: Hi all, we have an issue with our FreeSwitch box that to us seems to be performance related. The effect is that in FreeSwitch the number of session reaches the limit we set in config as: The limit is reached independently of its value because when FreeSwitch is in this state, the number of sessions grows indefinitely. We have tried to upgrade the hardware of the box moving from a Xeon 2 CPU E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but with this more powerful hardware it happens that the limit is reached with less sessions. It seems that performance are related to the speed of single core instead of speed of the box. Make sense? A example of "status" issued before the crash is: 14969852 session(s) since startup 13765 session(s) - 538 out of max 1000 per sec 30000 session(s) max min idle cpu 0.00/36.00 Current Stack Size/Max 240K/8192K Any advice? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/09ed994c/attachment.html From anthony.minessale at gmail.com Mon Oct 14 20:03:24 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Oct 2013 11:03:24 -0500 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: Try latest HEAD for a probable fix. On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: > Hi all, > we have an issue with our FreeSwitch box that to us seems to be > performance related. > The effect is that in FreeSwitch the number of session reaches the limit > we set in config as: > > > > The limit is reached independently of its value because when FreeSwitch is > in this state, the number of sessions grows indefinitely. > > We have tried to upgrade the hardware of the box moving from a Xeon 2 CPU > E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but with > this more powerful hardware it happens that the limit is reached with less > sessions. > It seems that performance are related to the speed of single core instead > of speed of the box. > Make sense? > > A example of "status" issued before the crash is: > > 14969852 session(s) since startup > 13765 session(s) - 538 out of max 1000 per sec > 30000 session(s) max > min idle cpu 0.00/36.00 > Current Stack Size/Max 240K/8192K > > > Any advice? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/abd7ec38/attachment.html From wstephen80 at gmail.com Mon Oct 14 20:21:23 2013 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 14 Oct 2013 18:21:23 +0200 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: Thank you for your advice. I have tried the 1.2.stable but I have due to abandon this branch due to a some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting x sessions ..." messages. I have seen that this issue (FS-5848) has already been solved so I can try with 1.2.stable but I don't see any fix that can affect my issue. On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Try latest HEAD for a probable fix. > > > > On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: > >> Hi all, >> we have an issue with our FreeSwitch box that to us seems to be >> performance related. >> The effect is that in FreeSwitch the number of session reaches the limit >> we set in config as: >> >> >> >> The limit is reached independently of its value because when FreeSwitch >> is in this state, the number of sessions grows indefinitely. >> >> We have tried to upgrade the hardware of the box moving from a Xeon 2 CPU >> E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but with >> this more powerful hardware it happens that the limit is reached with less >> sessions. >> It seems that performance are related to the speed of single core instead >> of speed of the box. >> Make sense? >> >> A example of "status" issued before the crash is: >> >> 14969852 session(s) since startup >> 13765 session(s) - 538 out of max 1000 per sec >> 30000 session(s) max >> min idle cpu 0.00/36.00 >> Current Stack Size/Max 240K/8192K >> >> >> Any advice? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/48687b71/attachment-0001.html From anthony.minessale at gmail.com Mon Oct 14 20:27:03 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Oct 2013 11:27:03 -0500 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: That's all the same thing...... Do you have this problem on master branch as well? On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde wrote: > Thank you for your advice. > I have tried the 1.2.stable but I have due to abandon this branch due to a > some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting > x sessions ..." messages. > I have seen that this issue (FS-5848) has already been solved so I can > try with 1.2.stable but I don't see any fix that can affect my issue. > > > On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Try latest HEAD for a probable fix. >> >> >> >> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: >> >>> Hi all, >>> we have an issue with our FreeSwitch box that to us seems to be >>> performance related. >>> The effect is that in FreeSwitch the number of session reaches the limit >>> we set in config as: >>> >>> >>> >>> The limit is reached independently of its value because when FreeSwitch >>> is in this state, the number of sessions grows indefinitely. >>> >>> We have tried to upgrade the hardware of the box moving from a Xeon 2 >>> CPU E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but >>> with this more powerful hardware it happens that the limit is reached with >>> less sessions. >>> It seems that performance are related to the speed of single core >>> instead of speed of the box. >>> Make sense? >>> >>> A example of "status" issued before the crash is: >>> >>> 14969852 session(s) since startup >>> 13765 session(s) - 538 out of max 1000 per sec >>> 30000 session(s) max >>> min idle cpu 0.00/36.00 >>> Current Stack Size/Max 240K/8192K >>> >>> >>> Any advice? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/f2a33994/attachment.html From krice at freeswitch.org Mon Oct 14 20:28:18 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Oct 2013 11:28:18 -0500 Subject: [Freeswitch-users] Performance issue In-Reply-To: Message-ID: That zombie sessions issue has been fixed... On 10/14/13 11:21 AM, "Stephen Wilde" wrote: > Thank you for your advice. > I have tried the 1.2.stable but I have due to abandon this branch due to a > some "zombie sessions" in FreeSwitch visible also during shutdown with a > "Waiting x sessions ..." messages. > I have seen that this issue (FS-5848)?has already been solved so I can try > with 1.2.stable but I don't see any fix that can affect my issue. > > > On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale > wrote: >> Try latest HEAD for a probable fix. >> >> >> >> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: >>> Hi all, >>> we have an issue with our FreeSwitch box that to us seems to be performance >>> related. >>> The effect is that in FreeSwitch the number of session reaches the limit we >>> set in config as: >>> >>> >>> >>> The limit is reached independently of its value because when FreeSwitch is >>> in this state, the number of sessions grows indefinitely. >>> >>> We have tried to upgrade the hardware of the box moving from a Xeon 2 CPU >>> E5649 (12 core?2.53GHz) to a Xeon 4 CPU?E5-4640 (32 core?2.40GHz) but with >>> this more powerful hardware it happens that the limit is reached with less >>> sessions. >>> It seems that performance are related to the speed of single core instead of >>> speed of the box. >>> Make sense? >>> >>> A example of "status" issued before the crash is: >>> >>> 14969852 session(s) since startup >>> 13765 session(s) - 538 out of max 1000 per sec >>> 30000 session(s) max >>> min idle cpu 0.00/36.00 >>> Current Stack Size/Max 240K/8192K >>> >>> >>> Any advice? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/cb2f4956/attachment.html From vbvbrj at gmail.com Mon Oct 14 20:30:14 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 19:30:14 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: <002101cec8f0$ad1a2866$7c07000a@smb.curriegrad2004.ca> References: <002101cec8f0$ad1a2866$7c07000a@smb.curriegrad2004.ca> Message-ID: <525C1C16.5030706@gmail.com> On 14.10.2013 18:18, Jeff Leung wrote: > It's working as intended So this line: 2482 905K DROP udp -- * * 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 STRING match "friendly-scanner" ALGO name bm TO 65535 means that none of the packet reached FS? -- Mimiko desu. From wstephen80 at gmail.com Mon Oct 14 20:33:45 2013 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 14 Oct 2013 18:33:45 +0200 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: I have tried only branch v1.2.stable not master. Is the master good for a production environment? On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That's all the same thing...... > > Do you have this problem on master branch as well? > > > > On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde wrote: > >> Thank you for your advice. >> I have tried the 1.2.stable but I have due to abandon this branch due to >> a some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting >> x sessions ..." messages. >> I have seen that this issue (FS-5848) has already been solved so I can >> try with 1.2.stable but I don't see any fix that can affect my issue. >> >> >> On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Try latest HEAD for a probable fix. >>> >>> >>> >>> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: >>> >>>> Hi all, >>>> we have an issue with our FreeSwitch box that to us seems to be >>>> performance related. >>>> The effect is that in FreeSwitch the number of session reaches the >>>> limit we set in config as: >>>> >>>> >>>> >>>> The limit is reached independently of its value because when FreeSwitch >>>> is in this state, the number of sessions grows indefinitely. >>>> >>>> We have tried to upgrade the hardware of the box moving from a Xeon 2 >>>> CPU E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but >>>> with this more powerful hardware it happens that the limit is reached with >>>> less sessions. >>>> It seems that performance are related to the speed of single core >>>> instead of speed of the box. >>>> Make sense? >>>> >>>> A example of "status" issued before the crash is: >>>> >>>> 14969852 session(s) since startup >>>> 13765 session(s) - 538 out of max 1000 per sec >>>> 30000 session(s) max >>>> min idle cpu 0.00/36.00 >>>> Current Stack Size/Max 240K/8192K >>>> >>>> >>>> Any advice? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/a6a0f456/attachment-0001.html From anthony.minessale at gmail.com Mon Oct 14 20:38:11 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Oct 2013 11:38:11 -0500 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: It is for many. It depends on the general goals etc. The problem you describe was only on stable and all the symptoms, not shutting down, channels count being wrong, zombies etc are all the same 1 problem now fixed. On Mon, Oct 14, 2013 at 11:33 AM, Stephen Wilde wrote: > I have tried only branch v1.2.stable not master. Is the master good for a > production environment? > > > On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> That's all the same thing...... >> >> Do you have this problem on master branch as well? >> >> >> >> On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde wrote: >> >>> Thank you for your advice. >>> I have tried the 1.2.stable but I have due to abandon this branch due to >>> a some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting >>> x sessions ..." messages. >>> I have seen that this issue (FS-5848) has already been solved so I can >>> try with 1.2.stable but I don't see any fix that can affect my issue. >>> >>> >>> On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Try latest HEAD for a probable fix. >>>> >>>> >>>> >>>> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: >>>> >>>>> Hi all, >>>>> we have an issue with our FreeSwitch box that to us seems to be >>>>> performance related. >>>>> The effect is that in FreeSwitch the number of session reaches the >>>>> limit we set in config as: >>>>> >>>>> >>>>> >>>>> The limit is reached independently of its value because when >>>>> FreeSwitch is in this state, the number of sessions grows indefinitely. >>>>> >>>>> We have tried to upgrade the hardware of the box moving from a Xeon 2 >>>>> CPU E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but >>>>> with this more powerful hardware it happens that the limit is reached with >>>>> less sessions. >>>>> It seems that performance are related to the speed of single core >>>>> instead of speed of the box. >>>>> Make sense? >>>>> >>>>> A example of "status" issued before the crash is: >>>>> >>>>> 14969852 session(s) since startup >>>>> 13765 session(s) - 538 out of max 1000 per sec >>>>> 30000 session(s) max >>>>> min idle cpu 0.00/36.00 >>>>> Current Stack Size/Max 240K/8192K >>>>> >>>>> >>>>> Any advice? >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/37a7e629/attachment.html From steveayre at gmail.com Mon Oct 14 20:59:59 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Oct 2013 17:59:59 +0100 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: <525C066C.20102@gmail.com> References: <525C066C.20102@gmail.com> Message-ID: Remember unless it gets blocked upstream the packet will still hit your firewall/server. Not FreeSWITCH sure, but it'll still consume some resources to receive, identify and block it. The scanner does not care that you're not responding, it'll continue to send anyway. On 14 October 2013 15:57, Mimiko wrote: > On 14.10.2013 17:54, Ken Rice wrote: > > This is sipvicious, its a brute force scanner... See > > http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with > > FreeSWITCH to defeat this attack > > Ken thank you. I am planning to set up Fail2ban. But for now need to > drop any packet from offending IP. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/d891064a/attachment-0001.html From nneul at mst.edu Mon Oct 14 21:13:32 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 14 Oct 2013 12:13:32 -0500 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: <525C066C.20102@gmail.com> Message-ID: <525C263C.6080106@mst.edu> What happens to traffic levels if it does get in? i.e. would it be a more effective countermeasure to let it in, but have it not be able to actually do anything? (honeypot/dummy accounts) -- Nathan On 10/14/2013 11:59 AM, Steven Ayre wrote: > Remember unless it gets blocked upstream the packet will still hit your firewall/server. Not FreeSWITCH sure, but it'll > still consume some resources to receive, identify and block it. The scanner does not care that you're not responding, > it'll continue to send anyway. > > > On 14 October 2013 15:57, Mimiko > wrote: > > On 14.10.2013 17:54, Ken Rice wrote: > > This is sipvicious, its a brute force scanner... See > > http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with > > FreeSWITCH to defeat this attack > > Ken thank you. I am planning to set up Fail2ban. But for now need to > drop any packet from offending IP. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From vbvbrj at gmail.com Mon Oct 14 21:17:42 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 20:17:42 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: <525C066C.20102@gmail.com> Message-ID: <525C2736.5070607@gmail.com> On 14.10.2013 19:59, Steven Ayre wrote: > Remember unless it gets blocked upstream the packet will still hit your > firewall/server. Not FreeSWITCH sure, but it'll still consume some > resources to receive, identify and block it. The scanner does not care > that you're not responding, it'll continue to send anyway. Yep. Now I called internet provider to inform about the attack and them to block that IP because we have monthly limit on this provider. -- Mimiko desu. From krice at freeswitch.org Mon Oct 14 21:25:38 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Oct 2013 12:25:38 -0500 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: <525C2736.5070607@gmail.com> Message-ID: The amount of traffic isnt that much... The real load on the system is the sip server trying to auth all those attempts.. On 10/14/13 12:17 PM, "Mimiko" wrote: > On 14.10.2013 19:59, Steven Ayre wrote: >> Remember unless it gets blocked upstream the packet will still hit your >> firewall/server. Not FreeSWITCH sure, but it'll still consume some >> resources to receive, identify and block it. The scanner does not care >> that you're not responding, it'll continue to send anyway. > > Yep. Now I called internet provider to inform about the attack and them > to block that IP because we have monthly limit on this provider. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From grcamauer at gmail.com Mon Oct 14 21:26:39 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 14 Oct 2013 14:26:39 -0300 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <525BEE73.20804@digitalmail.com> References: <525BE3E4.9070201@digitalmail.com> <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> <525BEE73.20804@digitalmail.com> Message-ID: <-3057579751474145159@unknownmsgid> Look into RTCP. Sent from my iPhone On 14/10/2013, at 10:23, Alex Lake wrote: Ah thanks (and to Gerald as well). Unfortunately the information I was really looking for doesn't seem to be in there. I was after current (live) values for the rtp_audio_ variables. Well, actually, I was after an indication of line quality. Any suggestions? Rgds, Alex http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump Or "info" app from within the dialplan. -Avi On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake wrote: > Is there a way I can look at all session variables of a live call from > the fs_cli prompt? > > eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" > > cheers, > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/0d74dd9d/attachment.html From martin.cmelik at gmail.com Mon Oct 14 21:21:04 2013 From: martin.cmelik at gmail.com (Martin Cmelik) Date: Mon, 14 Oct 2013 19:21:04 +0200 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: <525C066C.20102@gmail.com> Message-ID: Wouldnt be possible to have some IDS countermeasure and attack mitigation directly build in FS? For example Snort have more than 100 signatures for SIP attacks, but Snort is resource intensive and also cant inspect encrypted traffic. Thank you S pozdravem / Best regards, Martin ?mel?k Sent from ? > On 14. 10. 2013, at 18:59, Steven Ayre wrote: > > Remember unless it gets blocked upstream the packet will still hit your firewall/server. Not FreeSWITCH sure, but it'll still consume some resources to receive, identify and block it. The scanner does not care that you're not responding, it'll continue to send anyway. > > >> On 14 October 2013 15:57, Mimiko wrote: >> On 14.10.2013 17:54, Ken Rice wrote: >> > This is sipvicious, its a brute force scanner... See >> > http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with >> > FreeSWITCH to defeat this attack >> >> Ken thank you. I am planning to set up Fail2ban. But for now need to >> drop any packet from offending IP. >> >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/8494a59e/attachment-0001.html From nneul at mst.edu Mon Oct 14 21:43:33 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 14 Oct 2013 12:43:33 -0500 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: Message-ID: <525C2D45.8050602@mst.edu> Yeah, that's my point - does the load decrease if it "gets in", or does it just continue brute forcing? -- Nathan On 10/14/2013 12:25 PM, Ken Rice wrote: > The amount of traffic isnt that much... The real load on the system is the > sip server trying to auth all those attempts.. > > > On 10/14/13 12:17 PM, "Mimiko" wrote: > >> On 14.10.2013 19:59, Steven Ayre wrote: >>> Remember unless it gets blocked upstream the packet will still hit your >>> firewall/server. Not FreeSWITCH sure, but it'll still consume some >>> resources to receive, identify and block it. The scanner does not care >>> that you're not responding, it'll continue to send anyway. >> >> Yep. Now I called internet provider to inform about the attack and them >> to block that IP because we have monthly limit on this provider. > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From krice at freeswitch.org Mon Oct 14 21:55:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Oct 2013 12:55:41 -0500 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: Message-ID: There are counter attacks for sipvicious, the author has intentionally left ways to counter strike... This is something that fail2ban and others are out there for... Building them into freeswitch itself would create even more overhead and probably couldn?t do as good of a job as a purpose build IDS On 10/14/13 12:21 PM, "Martin Cmelik" wrote: > Wouldnt be possible to have some IDS countermeasure and attack mitigation > directly build in FS? > For example Snort have more than 100 signatures for SIP attacks, but Snort is > resource intensive and also cant inspect encrypted traffic. > > Thank you > > S pozdravem / Best regards, > Martin ?mel?k > > Sent from ? > > On 14. 10. 2013, at 18:59, Steven Ayre wrote: > >> Remember unless it gets blocked upstream the packet will still hit your >> firewall/server. Not FreeSWITCH sure, but it'll still consume some resources >> to receive, identify and block it. The scanner does not care that you're not >> responding, it'll continue to send anyway. >> >> >> On 14 October 2013 15:57, Mimiko wrote: >>> On 14.10.2013 17:54, Ken Rice wrote: >>>> > This is sipvicious, its a brute force scanner... See >>>> > http://wiki.freeswitch.org/wiki/Fail2ban on how to setup Fail2ban with >>>> > FreeSWITCH to defeat this attack >>> >>> Ken thank you. I am planning to set up Fail2ban. But for now need to >>> drop any packet from offending IP. >>> >>> -- >>> Mimiko desu. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131014/d6a28212/attachment.html From vbvbrj at gmail.com Mon Oct 14 22:05:39 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 14 Oct 2013 21:05:39 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: References: Message-ID: <525C3273.3080509@gmail.com> On 14.10.2013 20:55, Ken Rice wrote: > There are counter attacks for sipvicious, the author has intentionally > left ways to counter strike... This is something that fail2ban and > others are out there for... Building them into freeswitch itself would > create even more overhead and probably couldn?t do as good of a job as a > purpose build IDS Fail2ban must be apart from FS, as it can be used to monitor other services hosted on same machine. Even SSH server. Although the attacker's IP was blocked I still see in logs: Where is sofia/internal_A.B.C.D/100 at A.B.C.D appears? 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [NOTICE] switch_channel.c:1034 New Channel sofia/internal_A.B.C.D/100 at A.B.C.D [35a84dc8-0a11-449a-9a81-aa0a6ad75ab6] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] switch_core_session.c:1010 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] switch_core_session.c:1010 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] switch_core_state_machine.c:418 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_NEW 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] switch_core_state_machine.c:436 (sofia/internal_A.B.C.D/100 at A.B.C.D) State NEW 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.810383 [DEBUG] switch_core_session.c:1010 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 2013-10-14 20:48:51.810383 [DEBUG] sofia.c:1787 detaching session 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [WARNING] switch_core_state_machine.c:517 9ed841f1-b75d-4462-b8c4-100b976d3567 sofia/internal_A.B.C.D/100 at A.B.C.D Abandoned 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [NOTICE] switch_core_state_machine.c:520 Hangup sofia/internal_A.B.C.D/100 at A.B.C.D [CS_NEW] [WRONG_CALL_STATE] 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_channel.c:3139 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [KILL] 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_session.c:1345 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:418 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_HANGUP 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:681 (sofia/internal_A.B.C.D/100 at A.B.C.D) State HANGUP 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] mod_sofia.c:465 Channel sofia/internal_A.B.C.D/100 at A.B.C.D hanging up, cause: WRONG_CALL_STATE 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:48 sofia/internal_A.B.C.D/100 at A.B.C.D Standard HANGUP, cause: WRONG_CALL_STATE 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:681 (sofia/internal_A.B.C.D/100 at A.B.C.D) State HANGUP going to sleep 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:694 (sofia/internal_A.B.C.D/100 at A.B.C.D) Callstate Change DOWN -> HANGUP 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:449 (sofia/internal_A.B.C.D/100 at A.B.C.D) State Change CS_HANGUP -> CS_REPORTING 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_session.c:1345 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:418 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_REPORTING 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:766 (sofia/internal_A.B.C.D/100 at A.B.C.D) State REPORTING 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:92 sofia/internal_A.B.C.D/100 at A.B.C.D Standard REPORTING, cause: WRONG_CALL_STATE 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:766 (sofia/internal_A.B.C.D/100 at A.B.C.D) State REPORTING going to sleep 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:443 (sofia/internal_A.B.C.D/100 at A.B.C.D) State Change CS_REPORTING -> CS_DESTROY 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_session.c:1345 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_session.c:1553 Session 85 (sofia/internal_A.B.C.D/100 at A.B.C.D) Locked, Waiting on external entities 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [NOTICE] switch_core_session.c:1571 Session 85 (sofia/internal_A.B.C.D/100 at A.B.C.D) Ended 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [NOTICE] switch_core_session.c:1575 Close Channel sofia/internal_A.B.C.D/100 at A.B.C.D [CS_DESTROY] 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:568 (sofia/internal_A.B.C.D/100 at A.B.C.D) Callstate Change HANGUP -> DOWN 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:571 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_DESTROY 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:581 (sofia/internal_A.B.C.D/100 at A.B.C.D) State DESTROY 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] mod_sofia.c:375 sofia/internal_A.B.C.D/100 at A.B.C.D SOFIA DESTROY 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:99 sofia/internal_A.B.C.D/100 at A.B.C.D Standard DESTROY 9ed841f1-b75d-4462-b8c4-100b976d3567 2013-10-14 20:48:56.370468 [DEBUG] switch_core_state_machine.c:581 (sofia/internal_A.B.C.D/100 at A.B.C.D) State DESTROY going to sleep 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [WARNING] switch_core_state_machine.c:517 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 sofia/internal_A.B.C.D/100 at A.B.C.D Abandoned 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [NOTICE] switch_core_state_machine.c:520 Hangup sofia/internal_A.B.C.D/100 at A.B.C.D [CS_NEW] [WRONG_CALL_STATE] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_channel.c:3139 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [KILL] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_session.c:1345 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:418 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_HANGUP 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:681 (sofia/internal_A.B.C.D/100 at A.B.C.D) State HANGUP 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] mod_sofia.c:465 Channel sofia/internal_A.B.C.D/100 at A.B.C.D hanging up, cause: WRONG_CALL_STATE 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:48 sofia/internal_A.B.C.D/100 at A.B.C.D Standard HANGUP, cause: WRONG_CALL_STATE 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:681 (sofia/internal_A.B.C.D/100 at A.B.C.D) State HANGUP going to sleep 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:694 (sofia/internal_A.B.C.D/100 at A.B.C.D) Callstate Change DOWN -> HANGUP 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:449 (sofia/internal_A.B.C.D/100 at A.B.C.D) State Change CS_HANGUP -> CS_REPORTING 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_session.c:1345 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:418 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_REPORTING 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:766 (sofia/internal_A.B.C.D/100 at A.B.C.D) State REPORTING 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:92 sofia/internal_A.B.C.D/100 at A.B.C.D Standard REPORTING, cause: WRONG_CALL_STATE 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:766 (sofia/internal_A.B.C.D/100 at A.B.C.D) State REPORTING going to sleep 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:443 (sofia/internal_A.B.C.D/100 at A.B.C.D) State Change CS_REPORTING -> CS_DESTROY 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_session.c:1345 Send signal sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_session.c:1553 Session 86 (sofia/internal_A.B.C.D/100 at A.B.C.D) Locked, Waiting on external entities 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [NOTICE] switch_core_session.c:1571 Session 86 (sofia/internal_A.B.C.D/100 at A.B.C.D) Ended 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [NOTICE] switch_core_session.c:1575 Close Channel sofia/internal_A.B.C.D/100 at A.B.C.D [CS_DESTROY] 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:568 (sofia/internal_A.B.C.D/100 at A.B.C.D) Callstate Change HANGUP -> DOWN 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:571 (sofia/internal_A.B.C.D/100 at A.B.C.D) Running State Change CS_DESTROY 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:581 (sofia/internal_A.B.C.D/100 at A.B.C.D) State DESTROY 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] mod_sofia.c:375 sofia/internal_A.B.C.D/100 at A.B.C.D SOFIA DESTROY 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:99 sofia/internal_A.B.C.D/100 at A.B.C.D Standard DESTROY 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:49:01.830470 [DEBUG] switch_core_state_machine.c:581 (sofia/internal_A.B.C.D/100 at A.B.C.D) State DESTROY going to sleep -- Mimiko desu. From andretodd at verizon.net Tue Oct 15 03:40:15 2013 From: andretodd at verizon.net (Andre) Date: Mon, 14 Oct 2013 19:40:15 -0400 Subject: [Freeswitch-users] multiple sip profiles Message-ID: <041201cec936$bcdcb180$36961480$@verizon.net> What values need to be changed to create multiple sip profiles? Let's say my box has 2 IP's and I want only some traffic on one IP. In Internal I see these fields, should I change these to my other IP? Also on a real server with a static IP should I also remove auto-nat? Thanks From alex at digitalmail.com Tue Oct 15 11:42:58 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 15 Oct 2013 08:42:58 +0100 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <-3057579751474145159@unknownmsgid> References: <525BE3E4.9070201@digitalmail.com> <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> <525BEE73.20804@digitalmail.com> <-3057579751474145159@unknownmsgid> Message-ID: <525CF202.8040403@digitalmail.com> The Wiki is pretty skimpy on that - I set the interval, but now what? ;-) > Look into RTCP. > > > Sent from my iPhone > > On 14/10/2013, at 10:23, Alex Lake > wrote: > >> Ah thanks (and to Gerald as well). >> Unfortunately the information I was really looking for doesn't seem >> to be in there. >> I was after current (live) values for the rtp_audio_ variables. >> Well, actually, I was after an indication of line quality. >> Any suggestions? >> Rgds, >> Alex >>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump >>> Or "info" app from within the dialplan. >>> -Avi >>> >>> On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake >> > wrote: >>> >>> Is there a way I can look at all session variables of a live >>> call from >>> the fs_cli prompt? >>> >>> eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" >>> >>> cheers, >>> Alex >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/fbcaf9eb/attachment.html From alex at digitalmail.com Tue Oct 15 12:14:17 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 15 Oct 2013 09:14:17 +0100 Subject: [Freeswitch-users] Migrating from SQLite to MySQL Message-ID: <525CF959.4040408@digitalmail.com> Just wondered if there is an "easy" way to change my DB from the default sqlite to mysql without losing data (eg. voicemail settings)? From jleung at v10networks.ca Tue Oct 15 12:31:01 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Tue, 15 Oct 2013 16:31:01 +0800 Subject: [Freeswitch-users] Migrating from SQLite to MySQL Message-ID: <002201cec980$e47c4974$7c07000a@smb.curriegrad2004.ca> You could dump the data from the sqlite database as a bunch of inserts and then import it to your favorite rdbms of your choice. Alex Lake wrote: Just wondered if there is an "easy" way to change my DB from the default sqlite to mysql without losing data (eg. voicemail settings)? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From steveayre at gmail.com Tue Oct 15 14:25:40 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 11:25:40 +0100 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV Message-ID: Hi, Does anyone know what I'm doing wrong with: $ fs_encode -v -l mod_spandsp -l mod_com_g729 call.G729 call.wav I get: 2013-10-15 11:23:47.745292 [ERR] switch_core_codec.c:654 Invalid codec L16! Couldn't initialize codec for L16 at 8000h@20i -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/957c1005/attachment.html From steveayre at gmail.com Tue Oct 15 14:29:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 11:29:48 +0100 Subject: [Freeswitch-users] multiple sip profiles In-Reply-To: <041201cec936$bcdcb180$36961480$@verizon.net> References: <041201cec936$bcdcb180$36961480$@verizon.net> Message-ID: You duplicate the entire tag, usuallly found in /etc/freeswitch/sip_profiles A profile is binds to a single sip-ip and sip-port combination. You want to change the IP, port, or both on the new profile. rtp-ip *can* be the same, but is probably best matching the sip-ip (on a multihomed server you then know the media's taking the same working route that the signalling is). Also on a real server with a static IP should I also remove auto-nat? I don't use any nat stuff on a public IP. I have no ext-*-ip params set, which means they default to using the sip-ip and rtp-ip values. I also start freeswitch with the -nonat flag (in the init.d script) which lets it start up much faster as it doesn't bother autodetecting your non-existent NAT router. -Steve On 15 October 2013 00:40, Andre wrote: > What values need to be changed to create multiple sip profiles? > Let's say my box has 2 IP's and I want only some traffic on one IP. > > In Internal I see these fields, should I change these to my other IP? > > > > > > > > Also on a real server with a static IP should I also remove auto-nat? > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/ee9c8711/attachment-0001.html From grcamauer at gmail.com Tue Oct 15 14:43:48 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 15 Oct 2013 07:43:48 -0300 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: <-1738076673133177811@unknownmsgid> I think you need to tell it both the input and output filenames. Sent from my iPhone On 15/10/2013, at 07:27, Steven Ayre wrote: > Hi, > > Does anyone know what I'm doing wrong with: > > $ fs_encode -v -l mod_spandsp -l mod_com_g729 call.G729 call.wav > > I get: > 2013-10-15 11:23:47.745292 [ERR] switch_core_codec.c:654 Invalid codec L16! > Couldn't initialize codec for L16 at 8000h@20i > > -Steve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From grcamauer at gmail.com Tue Oct 15 14:45:26 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 15 Oct 2013 07:45:26 -0300 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <525CF202.8040403@digitalmail.com> References: <525BE3E4.9070201@digitalmail.com> <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> <525BEE73.20804@digitalmail.com> <-3057579751474145159@unknownmsgid> <525CF202.8040403@digitalmail.com> Message-ID: <-8327635250516141972@unknownmsgid> Did you compile and load the module? If so, a wires hark capture should show rtcp packets. Sent from my iPhone On 15/10/2013, at 04:54, Alex Lake wrote: The Wiki is pretty skimpy on that - I set the interval, but now what? ;-) Look into RTCP. Sent from my iPhone On 14/10/2013, at 10:23, Alex Lake wrote: Ah thanks (and to Gerald as well). Unfortunately the information I was really looking for doesn't seem to be in there. I was after current (live) values for the rtp_audio_ variables. Well, actually, I was after an indication of line quality. Any suggestions? Rgds, Alex http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump Or "info" app from within the dialplan. -Avi On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake wrote: > Is there a way I can look at all session variables of a live call from > the fs_cli prompt? > > eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" > > cheers, > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/cc997dc1/attachment.html From jaybinks at gmail.com Tue Oct 15 15:29:07 2013 From: jaybinks at gmail.com (jay binks) Date: Tue, 15 Oct 2013 21:29:07 +1000 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <-8327635250516141972@unknownmsgid> References: <525BE3E4.9070201@digitalmail.com> <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> <525BEE73.20804@digitalmail.com> <-3057579751474145159@unknownmsgid> <525CF202.8040403@digitalmail.com> <-8327635250516141972@unknownmsgid> Message-ID: what module do you speak of ? Jay On 15 October 2013 20:45, Guillermo Ruiz Camauer wrote: > Did you compile and load the module? If so, a wires hark capture should > show rtcp packets. > > Sent from my iPhone > > On 15/10/2013, at 04:54, Alex Lake wrote: > > The Wiki is pretty skimpy on that - I set the interval, but now what? ;-) > > Look into RTCP. > > > Sent from my iPhone > > On 14/10/2013, at 10:23, Alex Lake wrote: > > Ah thanks (and to Gerald as well). > Unfortunately the information I was really looking for doesn't seem to be > in there. > I was after current (live) values for the rtp_audio_ variables. > Well, actually, I was after an indication of line quality. > Any suggestions? > Rgds, > Alex > > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump > Or "info" app from within the dialplan. > -Avi > > On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake wrote: > >> Is there a way I can look at all session variables of a live call from >> the fs_cli prompt? >> >> eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" >> >> cheers, >> Alex >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/166467bb/attachment-0001.html From alex at digitalmail.com Tue Oct 15 15:38:23 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 15 Oct 2013 12:38:23 +0100 Subject: [Freeswitch-users] Inspecting sessions from fs_cli In-Reply-To: <-8327635250516141972@unknownmsgid> References: <525BE3E4.9070201@digitalmail.com> <00000141b70d5812-d889a4cb-c6cd-4abd-aca7-7a399df3bc93-000000@email.amazonses.com> <525BEE73.20804@digitalmail.com> <-3057579751474145159@unknownmsgid> <525CF202.8040403@digitalmail.com> <-8327635250516141972@unknownmsgid> Message-ID: <525D292F.2080100@digitalmail.com> Ah - so I'd need something else to look at and collate the data. Having been impressed with the rtp_audio_ variables, I wondered if some freeswitch module might do more of the work for me ;-) > Did you compile and load the module? If so, a wires hark capture > should show rtcp packets. > > Sent from my iPhone > > On 15/10/2013, at 04:54, Alex Lake > wrote: > >> The Wiki is pretty skimpy on that - I set the interval, but now what? >> ;-) >>> Look into RTCP. >>> >>> >>> Sent from my iPhone >>> >>> On 14/10/2013, at 10:23, Alex Lake >> > wrote: >>> >>>> Ah thanks (and to Gerald as well). >>>> Unfortunately the information I was really looking for doesn't seem >>>> to be in there. >>>> I was after current (live) values for the rtp_audio_ variables. >>>> Well, actually, I was after an indication of line quality. >>>> Any suggestions? >>>> Rgds, >>>> Alex >>>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dump >>>>> Or "info" app from within the dialplan. >>>>> -Avi >>>>> >>>>> On Mon, Oct 14, 2013 at 3:30 PM, Alex Lake >>>> > wrote: >>>>> >>>>> Is there a way I can look at all session variables of a live >>>>> call from >>>>> the fs_cli prompt? >>>>> >>>>> eg. "show session fb413c9a-c82b-4c4d-bb8c-5101aacb5ed5" >>>>> >>>>> cheers, >>>>> Alex >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/0ebe5f0e/attachment.html From andretodd at verizon.net Tue Oct 15 16:03:31 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 15 Oct 2013 08:03:31 -0400 Subject: [Freeswitch-users] multiple sip profiles Message-ID: <0MUP00C2ILIF8B50@vms173015.mailsrvcs.net> Thanks I'll try that out. I assume its internal profile only? Also -nonat is just an argument right when launching freeswitch.exe -----Original Message----- From: "Steven Ayre" Sent: ?10/?15/?2013 6:29 AM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] multiple sip profiles You duplicate the entire tag, usuallly found in /etc/freeswitch/sip_profiles A profile is binds to a single sip-ip and sip-port combination. You want to change the IP, port, or both on the new profile. rtp-ip *can* be the same, but is probably best matching the sip-ip (on a multihomed server you then know the media's taking the same working route that the signalling is). Also on a real server with a static IP should I also remove auto-nat? I don't use any nat stuff on a public IP. I have no ext-*-ip params set, which means they default to using the sip-ip and rtp-ip values. I also start freeswitch with the -nonat flag (in the init.d script) which lets it start up much faster as it doesn't bother autodetecting your non-existent NAT router. -Steve On 15 October 2013 00:40, Andre wrote: What values need to be changed to create multiple sip profiles? Let's say my box has 2 IP's and I want only some traffic on one IP. In Internal I see these fields, should I change these to my other IP? Also on a real server with a static IP should I also remove auto-nat? Thanks _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/c142b723/attachment-0001.html From steveayre at gmail.com Tue Oct 15 18:42:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 15:42:02 +0100 Subject: [Freeswitch-users] multiple sip profiles In-Reply-To: <0MUP00C2ILIF8B50@vms173015.mailsrvcs.net> References: <0MUP00C2ILIF8B50@vms173015.mailsrvcs.net> Message-ID: > > Thanks I'll try that out. I assume its internal profile only? Internal is just a profile in the example vanilla config. You can have any number of profiles you like, with any name. My advice goes for any profile where the bind (ie local on the server) address and external (ie seen by the sip client) address are the same. > Also -nonat is just an argument right when launching freeswitch.exe Yes. You'll still want the nat support if any of your profiles are behind NAT. On 15 October 2013 13:03, Andre Demattia wrote: > Thanks I'll try that out. I assume its internal profile only? > Also -nonat is just an argument right when launching freeswitch.exe > ------------------------------ > From: Steven Ayre > Sent: 10/15/2013 6:29 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] multiple sip profiles > > You duplicate the entire tag, usuallly found > in /etc/freeswitch/sip_profiles > > A profile is binds to a single sip-ip and sip-port combination. You want > to change the IP, port, or both on the new profile. rtp-ip *can* be the > same, but is probably best matching the sip-ip (on a multihomed server you > then know the media's taking the same working route that the signalling is). > > Also on a real server with a static IP should I also remove auto-nat? > > > I don't use any nat stuff on a public IP. I have no ext-*-ip params set, > which means they default to using the sip-ip and rtp-ip values. > > I also start freeswitch with the -nonat flag (in the init.d script) which > lets it start up much faster as it doesn't bother autodetecting your > non-existent NAT router. > > -Steve > > > On 15 October 2013 00:40, Andre wrote: > >> What values need to be changed to create multiple sip profiles? >> Let's say my box has 2 IP's and I want only some traffic on one IP. >> >> In Internal I see these fields, should I change these to my other IP? >> >> >> >> >> >> >> >> Also on a real server with a static IP should I also remove auto-nat? >> Thanks >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/ecbc97b8/attachment.html From steveayre at gmail.com Tue Oct 15 18:42:56 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 15:42:56 +0100 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: <-1738076673133177811@unknownmsgid> References: <-1738076673133177811@unknownmsgid> Message-ID: > > $ fs_encode -v -l mod_spandsp -l mod_com_g729 call.G729 call.wav I am... input of call.G729, output of call.wav On 15 October 2013 11:43, Guillermo Ruiz Camauer wrote: > I think you need to tell it both the input and output filenames. > > Sent from my iPhone > > On 15/10/2013, at 07:27, Steven Ayre wrote: > > > Hi, > > > > Does anyone know what I'm doing wrong with: > > > > $ fs_encode -v -l mod_spandsp -l mod_com_g729 call.G729 call.wav > > > > I get: > > 2013-10-15 11:23:47.745292 [ERR] switch_core_codec.c:654 Invalid codec > L16! > > Couldn't initialize codec for L16 at 8000h@20i > > > > -Steve > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/591ecd8a/attachment.html From smrdoshi at gmail.com Tue Oct 15 18:59:33 2013 From: smrdoshi at gmail.com (Samir Doshi) Date: Tue, 15 Oct 2013 07:59:33 -0700 (PDT) Subject: [Freeswitch-users] Call hangup not working in perl session Message-ID: <1381849173815-7595714.post@n2.nabble.com> Hi Guys, I am using perl session to handle call flow. I need to hangup call at exact 1 minute. For that i have set below line, but somehow its not working. Call keeps running. $session->execute( "set", "execute_on_answer=sched_hangup +60" ); Can anybody please point out the issue or something I missing. Thanks in advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-perl-session-tp7595714.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dvl36.ripe.nick at gmail.com Tue Oct 15 19:39:30 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 15 Oct 2013 18:39:30 +0300 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: Hi, When I tried I didn't used '-l mod_spandsp' and it worked fine. (1.2.stable) Dmitry. 2013/10/15 Steven Ayre > Hi, > > Does anyone know what I'm doing wrong with: > > $ fs_encode -v -l mod_spandsp -l mod_com_g729 call.G729 call.wav > > I get: > 2013-10-15 11:23:47.745292 [ERR] switch_core_codec.c:654 Invalid codec L16! > Couldn't initialize codec for L16 at 8000h@20i > > -Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/8447e237/attachment.html From steveayre at gmail.com Tue Oct 15 19:48:28 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 16:48:28 +0100 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: $ fs_encode -v -l mod_com_g729 a.G729 a.wav ...has the same error (actually the log shows it's still loading mod_spandsp mod_sndfile and mod_native_file anyway). On 15 October 2013 16:39, Dmitry Lysenko wrote: > Hi, > > When I tried I didn't used '-l mod_spandsp' and it worked fine. > (1.2.stable) > > Dmitry. > > > 2013/10/15 Steven Ayre > >> Hi, >> >> Does anyone know what I'm doing wrong with: >> >> $ fs_encode -v -l mod_spandsp -l mod_com_g729 call.G729 call.wav >> >> I get: >> 2013-10-15 11:23:47.745292 [ERR] switch_core_codec.c:654 Invalid codec >> L16! >> Couldn't initialize codec for L16 at 8000h@20i >> >> -Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/6900c95c/attachment-0001.html From steveayre at gmail.com Tue Oct 15 19:52:40 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 16:52:40 +0100 Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <1381849173815-7595714.post@n2.nabble.com> References: <1381849173815-7595714.post@n2.nabble.com> Message-ID: Is $session already answered? On 15 October 2013 15:59, Samir Doshi wrote: > Hi Guys, > > I am using perl session to handle call flow. > I need to hangup call at exact 1 minute. For that i have set below line, > but > somehow its not working. Call keeps running. > $session->execute( "set", "execute_on_answer=sched_hangup +60" ); > > Can anybody please point out the issue or something I missing. > > Thanks in advance. > > > > > > ----- > Thanks, > Samir > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-perl-session-tp7595714.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/f6ffadbd/attachment.html From smontour at verizon.net Tue Oct 15 20:01:50 2013 From: smontour at verizon.net (Sam Montour) Date: Tue, 15 Oct 2013 11:01:50 -0500 Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <1381849173815-7595714.post@n2.nabble.com> References: <1381849173815-7595714.post@n2.nabble.com> Message-ID: <006f01cec9bf$dc189c60$9449d520$@verizon.net> Hi, I had this working sometime back when I was testing a pre-paid application using Perl. Here is what I had in my script. my $max_session_time = 30; $session->execute("set", "execute_on_answer=sched_hangup +" . $max_session_time); Where in the script are you placing the execute command? Sam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samir Doshi Sent: Tuesday, October 15, 2013 10:00 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Call hangup not working in perl session Hi Guys, I am using perl session to handle call flow. I need to hangup call at exact 1 minute. For that i have set below line, but somehow its not working. Call keeps running. $session->execute( "set", "execute_on_answer=sched_hangup +60" ); Can anybody please point out the issue or something I missing. Thanks in advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-per l-session-tp7595714.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From alex at digitalmail.com Tue Oct 15 20:46:12 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 15 Oct 2013 17:46:12 +0100 Subject: [Freeswitch-users] Build error Message-ID: <525D7154.9050906@digitalmail.com> Not sure if this is of any interest. Just been building v1.2.stable on a (small) ubuntu box. Would having only 512MB RAM cramp gcc's style? making all mod_flite making in ... making in include ... making in src ... making in src/audio ... making in src/utils ... making in src/regex ... making in src/hrg ... making in src/stats ... making in src/speech ... making in src/lexicon ... making in src/synth ... making in src/wavesynth ... making in src/cg ... making in lang ... making in lang/cmulex ... making in lang/usenglish ... making in lang/cmu_us_kal ... making in lang/cmu_time_awb ... making in lang/cmu_us_kal16 ... gcc: internal compiler error: Killed (program cc1) Please submit a full bug report, with preprocessed source if appropriate. See for instructions. make[8]: *** [../../build/x86_64-linux-gnu/obj/lang/cmu_us_kal16/cmu_us_kal16_res.o] Error 4 make[7]: *** [../build/x86_64-linux-gnu/obj/lang/.make_build_dirs] Error 2 make[6]: *** [build/x86_64-linux-gnu/obj//.make_build_dirs] Error 2 make[5]: *** [/usr/local/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_flite-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 From steveayre at gmail.com Tue Oct 15 21:00:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Oct 2013 18:00:47 +0100 Subject: [Freeswitch-users] Build error In-Reply-To: <525D7154.9050906@digitalmail.com> References: <525D7154.9050906@digitalmail.com> Message-ID: If you don't need it trying removing mod_flite from modules.conf to skip building it On 15 October 2013 17:46, Alex Lake wrote: > Not sure if this is of any interest. Just been building v1.2.stable on a > (small) ubuntu box. Would having only 512MB RAM cramp gcc's style? > > making all mod_flite > making in ... > making in include ... > making in src ... > making in src/audio ... > making in src/utils ... > making in src/regex ... > making in src/hrg ... > making in src/stats ... > making in src/speech ... > making in src/lexicon ... > making in src/synth ... > making in src/wavesynth ... > making in src/cg ... > making in lang ... > making in lang/cmulex ... > making in lang/usenglish ... > making in lang/cmu_us_kal ... > making in lang/cmu_time_awb ... > making in lang/cmu_us_kal16 ... > gcc: internal compiler error: Killed (program cc1) > Please submit a full bug report, > with preprocessed source if appropriate. > See for instructions. > make[8]: *** > [../../build/x86_64-linux-gnu/obj/lang/cmu_us_kal16/cmu_us_kal16_res.o] > Error 4 > make[7]: *** [../build/x86_64-linux-gnu/obj/lang/.make_build_dirs] Error 2 > make[6]: *** [build/x86_64-linux-gnu/obj//.make_build_dirs] Error 2 > make[5]: *** > > [/usr/local/src/freeswitch/libs/flite-1.5.4-current/build/libs/libflite_cmu_us_awb.a] > Error 2 > make[4]: *** [all] Error 1 > make[3]: *** [mod_flite-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/ff740327/attachment.html From avi at avimarcus.net Tue Oct 15 21:03:57 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 15 Oct 2013 17:03:57 +0000 Subject: [Freeswitch-users] Build error In-Reply-To: <525D7154.9050906@digitalmail.com> References: <525D7154.9050906@digitalmail.com> Message-ID: <00000141bd12f28d-027228f7-89e6-4d81-ada5-c1801875fc8b-000000@email.amazonses.com> Yes, I never could build flite on just 512mb. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/a604ebee/attachment.html From alex at digitalmail.com Tue Oct 15 21:07:11 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 15 Oct 2013 18:07:11 +0100 Subject: [Freeswitch-users] Build error In-Reply-To: <525D7154.9050906@digitalmail.com> References: <525D7154.9050906@digitalmail.com> Message-ID: <525D763F.3020107@digitalmail.com> ...oops it's 12.04 LTS 64 bit. Now building on a slightly larger box. From alex at digitalmail.com Tue Oct 15 21:24:03 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 15 Oct 2013 18:24:03 +0100 Subject: [Freeswitch-users] Build error In-Reply-To: <00000141bd12f28d-027228f7-89e6-4d81-ada5-c1801875fc8b-000000@email.amazonses.com> References: <525D7154.9050906@digitalmail.com> <00000141bd12f28d-027228f7-89e6-4d81-ada5-c1801875fc8b-000000@email.amazonses.com> Message-ID: <525D7A33.7020704@digitalmail.com> Seems fine on a 1GB machine. > Yes, I never could build flite on just 512mb. > -Avi > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/4c8874e3/attachment.html From smrdoshi at gmail.com Tue Oct 15 22:08:00 2013 From: smrdoshi at gmail.com (Samir Doshi) Date: Tue, 15 Oct 2013 11:08:00 -0700 (PDT) Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <006f01cec9bf$dc189c60$9449d520$@verizon.net> References: <1381849173815-7595714.post@n2.nabble.com> <006f01cec9bf$dc189c60$9449d520$@verizon.net> Message-ID: <1381860480301-7595722.post@n2.nabble.com> Yeah $session is already answered at the beginning of call. $session->answer(); I have placed it just before bridge application, like this $session->execute( "set", "execute_on_answer=sched_hangup +60" ); $session->execute( "bridge", "$data_string" ); ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-perl-session-tp7595714p7595722.html Sent from the freeswitch-users mailing list archive at Nabble.com. From smontour at verizon.net Tue Oct 15 22:25:55 2013 From: smontour at verizon.net (Sam Montour) Date: Tue, 15 Oct 2013 13:25:55 -0500 Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <1381860480301-7595722.post@n2.nabble.com> References: <1381849173815-7595714.post@n2.nabble.com> <006f01cec9bf$dc189c60$9449d520$@verizon.net> <1381860480301-7595722.post@n2.nabble.com> Message-ID: <009101cec9d3$fd1d6e90$f7584bb0$@verizon.net> Comment out the $session->answer(); and try it. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samir Doshi Sent: Tuesday, October 15, 2013 1:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call hangup not working in perl session Yeah $session is already answered at the beginning of call. $session->answer(); I have placed it just before bridge application, like this $session->execute( "set", "execute_on_answer=sched_hangup +60" ); $session->execute( "bridge", "$data_string" ); ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-per l-session-tp7595714p7595722.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dvl36.ripe.nick at gmail.com Tue Oct 15 23:47:51 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Tue, 15 Oct 2013 22:47:51 +0300 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: I checked right now, and it work. (git, 1.2.stable, Oct,11) How about other codecs? Using 'sudo'? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/fee486c7/attachment.html From fs at voice2net.ca Wed Oct 16 00:42:16 2013 From: fs at voice2net.ca (fs) Date: Tue, 15 Oct 2013 16:42:16 -0400 Subject: [Freeswitch-users] blf on parked calls goes out. References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52587A17.1050009@freeswitch.org> <1381548604.88448.YahooMailNeo@web126204.mail.ne1.yahoo.com> <1381613092.88194.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: <02d401cec9e7$09cf4510$4dd1a8c0@DARCY> I pursued this awhile back with no results, it is still a puzzling issue. Any help what so ever would be appreciated. When I park a call I get a notify message telling me the call park is confirmed. Then at a random time, 2 seconds to 20 seconds out of the clear blue I get a message showing the call park is terminated. The call is still active and can be picked up using the button on the phone. The problem does not exist with extension blf. using the console (fs_cli) at debug level 7 shows no activity at the time of the disconnect. using version FreeSWITCH Version 1.5.5b+git~20130721T031327Z~adf5e2f6ec (git adf5e2f 2013-07-21 03:13:27Z) It seem to remember it worked correctly in the past but I am no longer sure. Darcy Primrose // this is for grandstream and yealink as well as snom parking. // this is used to retrieve a call from park using snom. I use snom 8.4.34, the load that works with blfs on freeswitch. confirmed sip:1 at myhome.pbx2net.ca;proto=park sip:park+1 terminated -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/3e06edde/attachment.html From fs.user at fordior.net Wed Oct 16 00:41:41 2013 From: fs.user at fordior.net (EL) Date: Tue, 15 Oct 2013 22:41:41 +0200 Subject: [Freeswitch-users] recommendations for Wifi SIP phones? In-Reply-To: References: <523B8595.4000502@mst.edu> <523B92F1.2070407@xtronics.com> <20130921151948.GB5507@0rdior.com> Message-ID: <20131015204141.GA31695@0rdior.com> > > On Siemens Gigaset DECT phones the max is G7221 (HD > > 16000Hz). Also, I experienced trouble calling with my baresip client to > > Siemens Gigaset handsets using the G7221 codecs. I'm not sure which end > > is causing the trouble (baresip or Siemens Gigaset DECT handset). The > > weird thing is that I'm hearing the voice from the Siemens side really > > good, but the Siemens side complains they are hearing my voice in slower > > motion. This appeared to happen in two different cases... First of all, I have to correct one typo: Siemens supports G722 not G7221. > > Someone has an idea what could cause this issue or is having the same > > experience? > Most prolly ptime mismatch on the codec side. Transcode it or hardset it. I have looked at: http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg21052.html And: http://wiki.freeswitch.org/wiki/Variable_sip_codec_negotiation Is that what you meant with setting the ptime or should I use another setting? Thanks for your input. -- EL From karl at xtronics.com Wed Oct 16 01:43:17 2013 From: karl at xtronics.com (Karl Schmidt) Date: Tue, 15 Oct 2013 16:43:17 -0500 Subject: [Freeswitch-users] recommendations for Wifi SIP phones? In-Reply-To: <20131015204141.GA31695@0rdior.com> References: <523B8595.4000502@mst.edu> <523B92F1.2070407@xtronics.com> <20130921151948.GB5507@0rdior.com> <20131015204141.GA31695@0rdior.com> Message-ID: <525DB6F5.4070200@xtronics.com> I have linphone working quite well with a WIFI connection - the detail is you have to limit other computers accessing the same AP at the same time. The interesting development is 802.11ac with MU-MIMO - I don't have hardware to test with, but MU-MIMO apparently fixes the VoIP wifi problem. http://en.wikipedia.org/wiki/Multi-user_MIMO Also - 802.11ac should do good beam-forming with devices that support it. http://chimera.labs.oreilly.com/books/1234000001739/ch04.html There is also work on UWB and wavelet radio that if allowed at enough power could do some amazing things. There is a link to a working server that would show the details - but the administration wants to jerk the public around for political purposes and has it redirected to a nag page. http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-02-48A1.pdf ... but you can see it here: http://web.archive.org/web/20111016141207/http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-02-48A1.pdf I first learned of wavelet radio in 1984 from a Russian engineer - our government did everything it could to prevent it's use and development. (It is difficult to intercept or jam such signals ) - if we lived in a sane world wifi would be based on this and be much more advanced than it is now. The current regulation limit the power and usefulness to limit competition to protect particular companies. In spite of these roadblocks wifi and UWB has slowly developed.. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 There's many a bestseller that could have been prevented by a good teacher. --Flannery O'Connor -------------------------------------------------------------------------------- From andretodd at verizon.net Wed Oct 16 02:18:57 2013 From: andretodd at verizon.net (Andre) Date: Tue, 15 Oct 2013 18:18:57 -0400 Subject: [Freeswitch-users] Freeswitch Hardware Message-ID: <06af01cec9f4$8b2f0200$a18d0600$@verizon.net> Assuming Money is no object and you wanted to get the most CPS/ Ports out of one server and you didn't care that it's stupid to put all those customers on one box. What type of hardware is recommended for running a carrier class traffic for media bypass for short duration dialer traffic? I'm assuming we want to max out the hardware. Processer, Hard drives, memory etc.? This would only be for the freeswitch server not the database. Would you also have many instances of Freeswitch to use up the server or just more profiles or just one profile? Also, how many ports/cps do you think one massive beast of a server can handle? 10,000 CPS? More? Less? I know this is a vague question but I'd like to hear what others think? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/b62d9f4f/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Oct 16 03:41:21 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 16 Oct 2013 00:41:21 +0100 Subject: [Freeswitch-users] Freeswitch Hardware In-Reply-To: <06af01cec9f4$8b2f0200$a18d0600$@verizon.net> References: <06af01cec9f4$8b2f0200$a18d0600$@verizon.net> Message-ID: DISCLAIMER: These suggestions are by no means acceptable if you are looking for quality/stability, but if you are looking for a cheap/cheerful way to get dialer traffic out the door, then the above should give you some ideas. I've written this response on the basis that quality/stability is not a goal. Throwing everything onto a single instance of FS is ultimately going to cause bottlenecks anyway (for example, the throughput for IVRs/voicemal would not be the same as the throughput for a simple user bridge) and no amount of hardware would fix that. You could potentially run multiple FS instances on a beefy box then use resource separation with LXC (you're less likely to run into performance/quality/jitter issues in comparison with virutalization with ESXi) You'd also probably want to take a look at the Sangoma DSP cards such as; http://www.sangoma.com/products/d100-30-400-sessions/ For hardware, you'd want something like 2x or 4x SSDs in RAID 1 using a decent hardware RAID card (LSI Logic is good), and make sure it's a proper RAID card (e.g. PERC H800) and not these crappy soft cards (e.g. PERC S100). You could throw a dual quad core at it and put a single CPU in to begin with to save on some cost until the point where you need to expand. Maybe 32-64GB RAM, which is pretty cheap these days. You could use consumer grade equipment and save on some cost. The RAM would be cheaper as it's non ECC/FB, and the majority of components would be cheaper also, but you'd most likely have to put it in a 4U ATX which could become costly if you plan on co-location. But be careful, this can result in corrupt data being written to disk if the memory goes bad. In terms of throughput, this has been heavily discussed already and there are many threads and examples of this on the wiki - use some google-fu and you'll find them. The only time you'd need to run multiple profiles in order to boost performance would be if you are processing thousands of calls per second (because sofia, the SIP parsing library, is single threaded), but as core devs have mentioned before you would have to be pushing some crazy numbers before this started to become a problem. You'd most likely run into problems when you start running applications such as IVR/TTS/Voicemail, and your throughput will depends on exactly what you are doing. For example, the throughput for 100% of calls going into a voicemail app would not be the same throughput as 100% of calls going through a simple user bridge. Hope this helps. Cal On Tue, Oct 15, 2013 at 11:18 PM, Andre wrote: > Assuming Money is no object and you wanted to get the most CPS/ Ports out > of one server and you didn?t care that it?s stupid to put all those > customers on one box. What type of hardware is recommended for running a > carrier class traffic for media bypass for short duration dialer traffic?* > *** > > ** ** > > I?m assuming we want to max out the hardware.**** > > ** ** > > Processer, Hard drives, memory etc.? This would only be for the > freeswitch server not the database.**** > > ** ** > > Would you also have many instances of Freeswitch to use up the server or > just more profiles or just one profile? Also, how many ports/cps do you > think one massive beast of a server can handle? 10,000 CPS? More? Less?*** > * > > ** ** > > I know this is a vague question but I?d like to hear what others think?*** > * > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/9d943633/attachment-0001.html From andretodd at verizon.net Wed Oct 16 05:00:43 2013 From: andretodd at verizon.net (Andre) Date: Tue, 15 Oct 2013 21:00:43 -0400 Subject: [Freeswitch-users] Freeswitch Hardware In-Reply-To: References: <06af01cec9f4$8b2f0200$a18d0600$@verizon.net> Message-ID: <078801ceca0b$256610a0$703231e0$@verizon.net> Do you have any suggestions for a quality way too? This will only be for a single bridge call and a LRN dip lookup with media bypass. No transcoding or voice mail. Thanks From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal Leeming [Simplicity Media Ltd] Sent: Tuesday, October 15, 2013 7:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch Hardware DISCLAIMER: These suggestions are by no means acceptable if you are looking for quality/stability, but if you are looking for a cheap/cheerful way to get dialer traffic out the door, then the above should give you some ideas. I've written this response on the basis that quality/stability is not a goal. Throwing everything onto a single instance of FS is ultimately going to cause bottlenecks anyway (for example, the throughput for IVRs/voicemal would not be the same as the throughput for a simple user bridge) and no amount of hardware would fix that. You could potentially run multiple FS instances on a beefy box then use resource separation with LXC (you're less likely to run into performance/quality/jitter issues in comparison with virutalization with ESXi) You'd also probably want to take a look at the Sangoma DSP cards such as; http://www.sangoma.com/products/d100-30-400-sessions/ For hardware, you'd want something like 2x or 4x SSDs in RAID 1 using a decent hardware RAID card (LSI Logic is good), and make sure it's a proper RAID card (e.g. PERC H800) and not these crappy soft cards (e.g. PERC S100). You could throw a dual quad core at it and put a single CPU in to begin with to save on some cost until the point where you need to expand. Maybe 32-64GB RAM, which is pretty cheap these days. You could use consumer grade equipment and save on some cost. The RAM would be cheaper as it's non ECC/FB, and the majority of components would be cheaper also, but you'd most likely have to put it in a 4U ATX which could become costly if you plan on co-location. But be careful, this can result in corrupt data being written to disk if the memory goes bad. In terms of throughput, this has been heavily discussed already and there are many threads and examples of this on the wiki - use some google-fu and you'll find them. The only time you'd need to run multiple profiles in order to boost performance would be if you are processing thousands of calls per second (because sofia, the SIP parsing library, is single threaded), but as core devs have mentioned before you would have to be pushing some crazy numbers before this started to become a problem. You'd most likely run into problems when you start running applications such as IVR/TTS/Voicemail, and your throughput will depends on exactly what you are doing. For example, the throughput for 100% of calls going into a voicemail app would not be the same throughput as 100% of calls going through a simple user bridge. Hope this helps. Cal On Tue, Oct 15, 2013 at 11:18 PM, Andre > wrote: Assuming Money is no object and you wanted to get the most CPS/ Ports out of one server and you didn't care that it's stupid to put all those customers on one box. What type of hardware is recommended for running a carrier class traffic for media bypass for short duration dialer traffic? I'm assuming we want to max out the hardware. Processer, Hard drives, memory etc.? This would only be for the freeswitch server not the database. Would you also have many instances of Freeswitch to use up the server or just more profiles or just one profile? Also, how many ports/cps do you think one massive beast of a server can handle? 10,000 CPS? More? Less? I know this is a vague question but I'd like to hear what others think? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/f04ad85b/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Oct 16 05:36:22 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 16 Oct 2013 02:36:22 +0100 Subject: [Freeswitch-users] Freeswitch Hardware In-Reply-To: <078801ceca0b$256610a0$703231e0$@verizon.net> References: <06af01cec9f4$8b2f0200$a18d0600$@verizon.net> <078801ceca0b$256610a0$703231e0$@verizon.net> Message-ID: That depends :) Some people choose to have just one box for their phone system, sat in a small rack in an office somewhere. Others may have a whole rack inside a single data center. And some might even have dual site redundancy too. You need to ask yourself, what is your end goal? Can you handle downtime/outages, if so how long? Do you have a budget in mind? Are you going to be building this solution yourself? How many concurrent calls are you expecting to have within the first 3 months? Do you have the appropriate IP transit to handle that traffic? For example, for just phone switch alone you could have two beefy machines in HA but what happens is the power goes down? Do you have A/B power feeds? How about if the rack power bar dies? How about if the switching gear for both racks die because they are on the same aggregate? If the RAID dies, how long would it take to re-build? If there was damage which caused perm data loss on both servers, how would you get up and running again quickly? How long would it take to restore block snapshots of the disk? Would that backup storage be in the same data center? What if the entire data center went down? If you have dual data centers, how do you handle IP fail over? Multi homed BGP isn't going to be instant, DNS fail over sucks, and any load balancer would introduce a single point of failure. How about if something in FS HA caused both servers to crash, due to a bug, could you handle that outage? How about if FS refused to start, would you have appropriate resources? Does the system need to be running 24/7? Who will handle the outage alarms at 3am? How will you monitor the system? The server could be up, but not handling calls properly. This is just a small subset of things you need to consider when building these solutions, in reality there are literally hundreds. Where do you draw the line and say "this is acceptable", this is only something you can decide based on the product you are selling. If you are serious about this and want a quick ROI, then you could perhaps approach consulting at freeswitch.org and ask for architectural design advice, you will then have a core developer from FS giving you recommendations based on your requirements. If you are doing this for learning experience, then you'll need about 1 year to train yourself into being able to support FS in production (based on my own experience - results may differ). Hope this helps - perhaps others might chime in with their thoughts as well. Cal On Wed, Oct 16, 2013 at 2:00 AM, Andre wrote: > Do you have any suggestions for a quality way too? This will only be for a > single bridge call and a LRN dip lookup with media bypass. No transcoding > or voice mail.**** > > Thanks**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal Leeming > [Simplicity Media Ltd] > *Sent:* Tuesday, October 15, 2013 7:41 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch Hardware**** > > ** ** > > DISCLAIMER: These suggestions are by no means acceptable if you are > looking for quality/stability, but if you are looking for a cheap/cheerful > way to get dialer traffic out the door, then the above should give you some > ideas. I've written this response on the basis that quality/stability is > not a goal.**** > > ** ** > > Throwing everything onto a single instance of FS is ultimately going to > cause bottlenecks anyway (for example, the throughput for IVRs/voicemal > would not be the same as the throughput for a simple user bridge) and no > amount of hardware would fix that. **** > > ** ** > > You could potentially run multiple FS instances on a beefy box then use > resource separation with LXC (you're less likely to run into > performance/quality/jitter issues in comparison with virutalization with > ESXi)**** > > ** ** > > You'd also probably want to take a look at the Sangoma DSP cards such as;* > *** > > http://www.sangoma.com/products/d100-30-400-sessions/**** > > ** ** > > For hardware, you'd want something like 2x or 4x SSDs in RAID 1 using a > decent hardware RAID card (LSI Logic is good), and make sure it's a proper > RAID card (e.g. PERC H800) and not these crappy soft cards (e.g. PERC > S100). You could throw a dual quad core at it and put a single CPU in to > begin with to save on some cost until the point where you need to expand. > Maybe 32-64GB RAM, which is pretty cheap these days.**** > > > You could use consumer grade equipment and save on some cost. The RAM > would be cheaper as it's non ECC/FB, and the majority of components would > be cheaper also, but you'd most likely have to put it in a 4U ATX which > could become costly if you plan on co-location. But be careful, this can > result in corrupt data being written to disk if the memory goes bad.**** > > ** ** > > In terms of throughput, this has been heavily discussed already and there > are many threads and examples of this on the wiki - use some google-fu and > you'll find them.**** > > ** ** > > The only time you'd need to run multiple profiles in order to boost > performance would be if you are processing thousands of calls per second > (because sofia, the SIP parsing library, is single threaded), but as core > devs have mentioned before you would have to be pushing some crazy numbers > before this started to become a problem.**** > > ** ** > > You'd most likely run into problems when you start running applications > such as IVR/TTS/Voicemail, and your throughput will depends on exactly what > you are doing. For example, the throughput for 100% of calls going into a > voicemail app would not be the same throughput as 100% of calls going > through a simple user bridge.**** > > ** ** > > Hope this helps.**** > > ** ** > > Cal**** > > ** ** > > ** ** > > ** ** > > ** ** > > ** ** > > On Tue, Oct 15, 2013 at 11:18 PM, Andre wrote:**** > > Assuming Money is no object and you wanted to get the most CPS/ Ports out > of one server and you didn?t care that it?s stupid to put all those > customers on one box. What type of hardware is recommended for running a > carrier class traffic for media bypass for short duration dialer traffic?* > *** > > **** > > I?m assuming we want to max out the hardware.**** > > **** > > Processer, Hard drives, memory etc.? This would only be for the > freeswitch server not the database.**** > > **** > > Would you also have many instances of Freeswitch to use up the server or > just more profiles or just one profile? Also, how many ports/cps do you > think one massive beast of a server can handle? 10,000 CPS? More? Less?*** > * > > **** > > I know this is a vague question but I?d like to hear what others think?*** > * > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/4b11e673/attachment-0001.html From brian at freeswitch.org Wed Oct 16 05:42:11 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Oct 2013 20:42:11 -0500 Subject: [Freeswitch-users] Channels Getting Stuck In-Reply-To: References: Message-ID: <0919D2A7-5E9D-423D-B007-E7614366A9E0@freeswitch.org> Update please. I hear this was fixed. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 10, 2013, at 1:07 AM, Vishal Kakkar wrote: > I have setup with 2 PRIs(terminating on sangoma cards) connected to FS using wanpipe. > > During campaign my channels are getting stuck.. > show cahnnels shows 52 channels but actual calls are only few.. > > Here is output of show channels- > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num > 18a7fa23-3f47-4a31-8c0d-c37e6b12b668,outbound,2013-10-10 11:16:49,1381384009,FreeTDM/1:9/09830701954,CS_CONSUME_MEDIA,,01203896280,,09830701954,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09830701954,,18a7fa23-3f47-4a31-8c0d-c37e6b12b668,, > 44de04f1-8b22-44d5-9665-c5771d1d66b4,outbound,2013-10-10 11:21:41,1381384301,FreeTDM/1:3/09327241861,CS_EXECUTE,Outbound Call,09327241861,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,44de04f1-8b22-44d5-9665-c5771d1d66b4,, > e564edcf-d590-4c45-ab0e-3f19e08a872c,outbound,2013-10-10 11:22:02,1381384322,FreeTDM/1:7/07830354545,CS_EXECUTE,Outbound Call,07830354545,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,e564edcf-d590-4c45-ab0e-3f19e08a872c,, > b5a83d5a-b705-41ad-8abb-d9c319d2e594,outbound,2013-10-10 11:22:32,1381384352,FreeTDM/1:18/07566326690,CS_EXECUTE,Outbound Call,07566326690,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,b5a83d5a-b705-41ad-8abb-d9c319d2e594,, > 53df0a17-a450-44b8-ab70-915f8f662b47,outbound,2013-10-10 11:22:45,1381384365,FreeTDM/1:16/09937432216,CS_EXECUTE,Outbound Call,09937432216,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,53df0a17-a450-44b8-ab70-915f8f662b47,, > 510a75eb-2340-4a50-9284-6247b52f2946,outbound,2013-10-10 11:22:46,1381384366,FreeTDM/1:22/09438189454,CS_EXECUTE,Outbound Call,09438189454,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,510a75eb-2340-4a50-9284-6247b52f2946,, > ae801140-b8eb-4824-b3bf-49cefae8eef0,outbound,2013-10-10 11:22:52,1381384372,FreeTDM/1:19/09767443205,CS_EXECUTE,Outbound Call,09767443205,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,ae801140-b8eb-4824-b3bf-49cefae8eef0,, > 0fd82b66-34cb-4c72-8a0f-3e7781c4b4f3,outbound,2013-10-10 11:23:02,1381384382,FreeTDM/1:2/09777825850,CS_EXECUTE,Outbound Call,09777825850,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,0fd82b66-34cb-4c72-8a0f-3e7781c4b4f3,, > 3ba5d15b-5233-441c-9954-a8668a4f4a39,outbound,2013-10-10 11:23:02,1381384382,FreeTDM/1:12/09765341919,CS_EXECUTE,Outbound Call,09765341919,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,3ba5d15b-5233-441c-9954-a8668a4f4a39,, > bb4be954-f5a4-49ad-a02f-4d9903466d69,outbound,2013-10-10 11:23:18,1381384398,FreeTDM/1:3/09329366443,CS_CONSUME_MEDIA,,01203896280,,09329366443,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09329366443,,bb4be954-f5a4-49ad-a02f-4d9903466d69,, > d6d66ff1-13cb-4c22-a66c-246baaed03c7,outbound,2013-10-10 11:23:18,1381384398,FreeTDM/1:8/09690699408,CS_CONSUME_MEDIA,,01203896280,,09690699408,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09690699408,,d6d66ff1-13cb-4c22-a66c-246baaed03c7,, > a9adc429-6f37-4f8f-a395-c6fab5d15ca2,outbound,2013-10-10 11:23:38,1381384418,FreeTDM/1:9/09623862679,CS_EXECUTE,Outbound Call,09623862679,,obdExtension,hangup,,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,a9adc429-6f37-4f8f-a395-c6fab5d15ca2,, > 0f12198e-87fc-4836-9255-47a03644a151,outbound,2013-10-10 11:23:52,1381384432,FreeTDM/1:6/09812005112,CS_EXECUTE,Outbound Call,09812005112,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,0f12198e-87fc-4836-9255-47a03644a151,, > 1660165c-51b3-450a-a4a4-23f616dc04ab,outbound,2013-10-10 11:24:03,1381384443,FreeTDM/1:3/09754094192,CS_EXECUTE,Outbound Call,09754094192,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,1660165c-51b3-450a-a4a4-23f616dc04ab,, > 9e773534-1d16-4ff6-8760-f743c02a3973,outbound,2013-10-10 11:24:08,1381384448,FreeTDM/1:4/09420423005,CS_EXECUTE,Outbound Call,09420423005,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,9e773534-1d16-4ff6-8760-f743c02a3973,, > 5fc752f4-e34a-462d-b21d-80da1af55b73,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:6/09439082697,CS_CONSUME_MEDIA,,01203896280,,09439082697,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09439082697,,5fc752f4-e34a-462d-b21d-80da1af55b73,, > 4aaf9bc3-da10-422e-aa3d-3120a9751671,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:11/09300153456,CS_EXECUTE,Outbound Call,09300153456,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,4aaf9bc3-da10-422e-aa3d-3120a9751671,, > d96829c1-cba2-43ad-8ff3-6a6976bcf78d,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:13/09439082697,CS_EXECUTE,Outbound Call,09439082697,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,d96829c1-cba2-43ad-8ff3-6a6976bcf78d,, > 46d90236-18df-4f3d-b5ae-11aac8e9baf3,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:14/09977252323,CS_EXECUTE,Outbound Call,09977252323,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,46d90236-18df-4f3d-b5ae-11aac8e9baf3,, > 95cea2f1-4c3b-4a15-9961-5706855b427b,outbound,2013-10-10 11:24:18,1381384458,FreeTDM/1:17/09420423005,CS_CONSUME_MEDIA,,01203896280,,09420423005,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09420423005,,95cea2f1-4c3b-4a15-9961-5706855b427b,, > 2f87fdb2-ea5c-4157-a0c8-d1746cf8ad55,outbound,2013-10-10 11:24:30,1381384470,FreeTDM/1:18/09810666158,CS_CONSUME_MEDIA,,01203896280,,09810666158,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09810666158,,2f87fdb2-ea5c-4157-a0c8-d1746cf8ad55,, > c710f549-ca34-40ea-9a75-e0b675ea1b5c,outbound,2013-10-10 11:24:30,1381384470,FreeTDM/1:19/09810666158,CS_EXECUTE,Outbound Call,09810666158,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,c710f549-ca34-40ea-9a75-e0b675ea1b5c,, > bde4aea8-e9d3-4209-b186-63fe3cacf67d,outbound,2013-10-10 11:24:46,1381384486,FreeTDM/1:12/09766434011,CS_EXECUTE,Outbound Call,09766434011,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,bde4aea8-e9d3-4209-b186-63fe3cacf67d,, > c9c88437-64e1-49a4-a180-985777f265d4,outbound,2013-10-10 11:24:51,1381384491,FreeTDM/1:9/07587465873,CS_EXECUTE,Outbound Call,07587465873,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,c9c88437-64e1-49a4-a180-985777f265d4,, > 5f89a635-ba5c-42c1-a98f-573eda504372,outbound,2013-10-10 11:24:56,1381384496,FreeTDM/1:5/09921748436,CS_EXECUTE,Outbound Call,09921748436,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,5f89a635-ba5c-42c1-a98f-573eda504372,, > 0cb88568-46e7-4d4c-8180-e5cf489947ba,outbound,2013-10-10 11:24:56,1381384496,FreeTDM/1:7/09920817954,CS_CONSUME_MEDIA,,01203896280,,09920817954,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09920817954,,0cb88568-46e7-4d4c-8180-e5cf489947ba,, > 56de78a3-62b7-4710-ae96-43f35c00335c,outbound,2013-10-10 11:25:06,1381384506,FreeTDM/1:15/09617761948,CS_EXECUTE,Outbound Call,09617761948,,obdExtension,hangup,,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,56de78a3-62b7-4710-ae96-43f35c00335c,, > 8bd2bddc-72c2-460f-a731-158f86abe268,outbound,2013-10-10 11:25:18,1381384518,FreeTDM/1:4/09890804877,CS_CONSUME_MEDIA,,01203896280,,09890804877,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09890804877,,8bd2bddc-72c2-460f-a731-158f86abe268,, > 7b016887-890a-4c66-aebe-62895b6f79bd,outbound,2013-10-10 11:25:22,1381384522,FreeTDM/1:2/09536983588,CS_EXECUTE,Outbound Call,09536983588,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,7b016887-890a-4c66-aebe-62895b6f79bd,, > 5fc9f447-2f2b-4e81-9d36-f76c9c3ffb53,outbound,2013-10-10 11:25:22,1381384522,FreeTDM/1:6/09763675941,CS_EXECUTE,Outbound Call,09763675941,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,5fc9f447-2f2b-4e81-9d36-f76c9c3ffb53,, > 86983e1e-7632-47eb-a5aa-e88aff98feae,outbound,2013-10-10 11:25:22,1381384522,FreeTDM/1:8/09823961213,CS_EXECUTE,Outbound Call,09823961213,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,86983e1e-7632-47eb-a5aa-e88aff98feae,, > f838d469-bc5b-4485-9960-5760d1aa92df,outbound,2013-10-10 11:25:31,1381384531,FreeTDM/1:1/09831622477,CS_EXECUTE,Outbound Call,09831622477,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,f838d469-bc5b-4485-9960-5760d1aa92df,, > 1509d219-d910-4aa1-80db-15ccaad98d27,outbound,2013-10-10 11:25:32,1381384532,FreeTDM/1:10/09827184894,CS_EXECUTE,Outbound Call,09827184894,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,1509d219-d910-4aa1-80db-15ccaad98d27,, > 95ea1b21-146b-4e5c-a39d-a11447429eda,outbound,2013-10-10 11:25:38,1381384538,FreeTDM/1:11/09301583677,CS_EXECUTE,Outbound Call,09301583677,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,95ea1b21-146b-4e5c-a39d-a11447429eda,, > f189518b-4f59-4660-840d-4ad87e243ec5,outbound,2013-10-10 11:25:48,1381384548,FreeTDM/1:3/09423175934,CS_CONSUME_MEDIA,,01203896280,,09423175934,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09423175934,,f189518b-4f59-4660-840d-4ad87e243ec5,, > 76103c25-1509-4469-8061-181e2d9c3179,outbound,2013-10-10 11:25:48,1381384548,FreeTDM/1:13/09423175934,CS_EXECUTE,Outbound Call,09423175934,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,76103c25-1509-4469-8061-181e2d9c3179,, > fae8ffb9-03a4-436b-b194-c5bed8dd4f55,outbound,2013-10-10 11:25:51,1381384551,FreeTDM/1:4/09891012004,CS_EXECUTE,Outbound Call,09891012004,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,fae8ffb9-03a4-436b-b194-c5bed8dd4f55,, > d43115c8-88a4-4853-a2d7-6a2355f250e9,outbound,2013-10-10 11:25:52,1381384552,FreeTDM/1:14/09433214771,CS_EXECUTE,Outbound Call,09433214771,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,d43115c8-88a4-4853-a2d7-6a2355f250e9,, > ad016f35-3186-40cb-b1db-1d0b961f9945,outbound,2013-10-10 11:25:52,1381384552,FreeTDM/1:16/09883025715,CS_EXECUTE,Outbound Call,09883025715,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,ad016f35-3186-40cb-b1db-1d0b961f9945,, > 4d8fca62-d5a4-4cdb-b70d-7193a2ff2307,outbound,2013-10-10 11:26:07,1381384567,FreeTDM/1:7/09583755624,CS_EXECUTE,Outbound Call,09583755624,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,4d8fca62-d5a4-4cdb-b70d-7193a2ff2307,, > 931fb548-3ed0-4085-9c65-2196b76999a4,outbound,2013-10-10 11:26:09,1381384569,FreeTDM/1:9/09001192140,CS_EXECUTE,Outbound Call,09001192140,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,931fb548-3ed0-4085-9c65-2196b76999a4,, > c1d192aa-9589-422d-b2b6-385a837ad8fc,outbound,2013-10-10 11:26:09,1381384569,FreeTDM/1:10/09583755624,CS_CONSUME_MEDIA,,01203896280,,09583755624,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09583755624,,c1d192aa-9589-422d-b2b6-385a837ad8fc,, > 31955dd1-d701-4cb0-b76b-cc33581e1ae2,outbound,2013-10-10 11:26:09,1381384569,FreeTDM/1:12/09223317664,CS_EXECUTE,Outbound Call,09223317664,,obdExtension,playback,/usr/local/freeswitch/media/DailyODB/Oct_10_2013.wav,xml,obd,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,ACTIVE,,,RECV,31955dd1-d701-4cb0-b76b-cc33581e1ae2,, > 87570d9f-621c-4ee9-ba2c-4cacb5375adf,outbound,2013-10-10 11:26:10,1381384570,FreeTDM/1:17/09583755624,CS_CONSUME_MEDIA,,01203896280,,09583755624,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09583755624,,87570d9f-621c-4ee9-ba2c-4cacb5375adf,, > 2d254646-62bf-4505-870e-e7c5eabd7832,outbound,2013-10-10 11:26:37,1381384597,FreeTDM/1:1/09469549674,CS_CONSUME_MEDIA,,01203896280,,09469549674,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09469549674,,2d254646-62bf-4505-870e-e7c5eabd7832,, > fcbcf8ad-2660-4b6a-8310-b837305b72db,outbound,2013-10-10 11:26:38,1381384598,FreeTDM/1:3/09453287889,CS_CONSUME_MEDIA,,01203896280,,09453287889,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09453287889,,fcbcf8ad-2660-4b6a-8310-b837305b72db,, > 56a96b36-e840-4af9-9aef-bf9d44ba93eb,outbound,2013-10-10 11:26:40,1381384600,FreeTDM/1:5/09223317664,CS_CONSUME_MEDIA,,01203896280,,09223317664,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,09223317664,,56a96b36-e840-4af9-9aef-bf9d44ba93eb,, > 921446b6-944b-4e5f-999a-69091cf6c5ca,outbound,2013-10-10 11:26:43,1381384603,FreeTDM/1:10/08018158992,CS_CONSUME_MEDIA,,01203896280,,08018158992,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,08018158992,,921446b6-944b-4e5f-999a-69091cf6c5ca,, > 71c7a9ea-c897-4722-bb66-7a8d92801bef,outbound,2013-10-10 11:26:43,1381384603,FreeTDM/1:15/08018158992,CS_CONSUME_MEDIA,,01203896280,,08018158992,,,,default,PCMA,8000,64000,PCMA,8000,64000,,xx.yyy.com,,,EARLY,Outbound Call,08018158992,,71c7a9ea-c897-4722-bb66-7a8d92801bef,, > 49 channels > > Same channel is apearing multiple times as i have bolded channel 1:9 in above output. > > Please help what should i do to overcome it.. My 3 hour campaign is now taking 18 hours becasue of this issue. > > Thanks, > -Vishal. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs at voice2net.ca Wed Oct 16 05:59:35 2013 From: fs at voice2net.ca (fs) Date: Tue, 15 Oct 2013 21:59:35 -0400 Subject: [Freeswitch-users] update to blf on parked calls goes out. References: <1381438205.67143.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52587A17.1050009@freeswitch.org> <1381548604.88448.YahooMailNeo@web126204.mail.ne1.yahoo.com><1381613092.88194.YahooMailNeo@web126203.mail.ne1.yahoo.com> <02d401cec9e7$09cf4510$4dd1a8c0@DARCY> Message-ID: <030c01ceca13$5ddd0bc0$4dd1a8c0@DARCY> Here is the scenario. Park a call using park+1. Park light illuminates. As soon as a set sends another subscriber to park+1, Freeswitch responds with a notify with state terminated. That would seem to me to be a bug or I have a setting re park incorrect? Darcy Primrose SUBSCRIBE sip:park+1 at 192.168.209.153:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.209.2:8202;branch=z9hG4bK2012800508;rport From: ;tag=1250658599 To: ;tag=LcDlvVf0Tjxb Call-ID: 171407988-8202-10 at BJC.BGI.CAJ.C CSeq: 20125 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2110 1.0.5.26 Expires: 60 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 759 bytes to udp/[192.168.209.2]:8202 at 01:55:02.425203: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.209.2:8202;branch=z9hG4bK2012800508;rport=8202 From: ;tag=1250658599 To: ;tag=LcDlvVf0Tjxb Call-ID: 171407988-8202-10 at BJC.BGI.CAJ.C CSeq: 20125 SUBSCRIBE Contact: Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130721T031327Z~adf5e2f6ec Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=60 Content-Length: 0 ------------------------------------------------------------------------ send 1090 bytes to udp/[192.168.209.78]:8204 at 01:55:02.427511: ------------------------------------------------------------------------ NOTIFY sip:102 at 192.168.209.78:8204 SIP/2.0 Via: SIP/2.0/UDP 192.168.209.153;rport;branch=z9hG4bKHe70rga87UKrj Max-Forwards: 70 From: ;tag=lPDgLOatBRrZ To: ;tag=g8cla3d4o0 Call-ID: 3c26702d8a01-sf05kqniay04 CSeq: 691946427 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130721T031327Z~adf5e2f6ec Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=2113 Content-Type: application/dialog-info+xml Content-Length: 233 terminated ----- Original Message ----- From: fs To: FreeSWITCH Users Help Sent: Tuesday, October 15, 2013 4:42 PM Subject: [Freeswitch-users] blf on parked calls goes out. I pursued this awhile back with no results, it is still a puzzling issue. Any help what so ever would be appreciated. When I park a call I get a notify message telling me the call park is confirmed. Then at a random time, 2 seconds to 20 seconds out of the clear blue I get a message showing the call park is terminated. The call is still active and can be picked up using the button on the phone. The problem does not exist with extension blf. using the console (fs_cli) at debug level 7 shows no activity at the time of the disconnect. using version FreeSWITCH Version 1.5.5b+git~20130721T031327Z~adf5e2f6ec (git adf5e2f 2013-07-21 03:13:27Z) It seem to remember it worked correctly in the past but I am no longer sure. Darcy Primrose // this is for grandstream and yealink as well as snom parking. // this is used to retrieve a call from park using snom. I use snom 8.4.34, the load that works with blfs on freeswitch. confirmed sip:1 at myhome.pbx2net.ca;proto=park sip:park+1 terminated ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4158 / Virus Database: 3609/6738 - Release Date: 10/10/13 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131015/c15b218b/attachment-0001.html From smrdoshi at gmail.com Wed Oct 16 11:29:52 2013 From: smrdoshi at gmail.com (Samir Doshi) Date: Wed, 16 Oct 2013 00:29:52 -0700 (PDT) Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <009101cec9d3$fd1d6e90$f7584bb0$@verizon.net> References: <1381849173815-7595714.post@n2.nabble.com> <006f01cec9bf$dc189c60$9449d520$@verizon.net> <1381860480301-7595722.post@n2.nabble.com> <009101cec9d3$fd1d6e90$f7584bb0$@verizon.net> Message-ID: <1381908592318-7595737.post@n2.nabble.com> That works. Thanks However I notice one thing, If I set 60 seconds sched time then my call hang up at 57 seconds. I tried many times and all the time it hangup at 57 seconds. Not sure if that's something related to fs. Any thoughts? ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-perl-session-tp7595714p7595737.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Wed Oct 16 16:28:24 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 16 Oct 2013 13:28:24 +0100 Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <1381908592318-7595737.post@n2.nabble.com> References: <1381849173815-7595714.post@n2.nabble.com> <006f01cec9bf$dc189c60$9449d520$@verizon.net> <1381860480301-7595722.post@n2.nabble.com> <009101cec9d3$fd1d6e90$f7584bb0$@verizon.net> <1381908592318-7595737.post@n2.nabble.com> Message-ID: You could also export it instead so it executes on the bleg not the aleg. That should work regardless of whether you have the aleg answered already or not. I'd personally prefer not answering the aleg unless you need to - most systems start billing from time answered. So the caller'll get charged for ringing time even if the call isn't answered. Not sure what would cause the 3s difference. If anything there should be a small delay on answer before executing sched_hangup - I could understand 61s much more than 57s. -Steve On 16 October 2013 08:29, Samir Doshi wrote: > That works. Thanks > > However I notice one thing, If I set 60 seconds sched time then my call > hang > up at 57 seconds. I tried many times and all the time it hangup at 57 > seconds. Not sure if that's something related to fs. Any thoughts? > > > > ----- > Thanks, > Samir > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-perl-session-tp7595714p7595737.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/60420b9f/attachment.html From vbvbrj at gmail.com Wed Oct 16 16:45:53 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 16 Oct 2013 15:45:53 +0300 Subject: [Freeswitch-users] New channel call direction. Message-ID: <525E8A81.9010806@gmail.com> Hi. In the logs I see: [NOTICE] switch_channel.c:1034 New Channel sofia/.... How can I see that this new channel is an incoming call to FS or is a call made by FS? I can't determine from this notice that is a connecton to FS or from FS to other phone/pbx. -- Mimiko desu. From callum.guy at x-on.co.uk Wed Oct 16 20:27:26 2013 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 16 Oct 2013 17:27:26 +0100 Subject: [Freeswitch-users] Cracking hangup_after_bridge for busy destinations Message-ID: Hi All, I'm having issues with inbound call which bridges to an external SIP handset, gets busy and then hangs up. All i want is for the A-leg not to hang up when the bridge fails (/completes with result BUSY) I am using ESL inbound and my event driven script will answer the call, set the ringback to a UK tone, play 1 second of silence, then issue the bridge as follows: {ignore_early_media=true,continue_on_fail=true,park_after_bridge=true,hangup_after_bridge=false}sofia/external/ 200010006 at sipserv.net Once this has all gone through everything works correctly when the user is available. When busy however the calling channel is ending the call rather than returning to park. I had anticipated collecting the CHANNEL_EXECUTE_COMPLETE event for the bridge command, reading the variable_DIALSTATUS for "BUSY" and then playing busy tone but it is just hanging up. Is there a clean way to resolve this? Even if its just setting the "busytone" equivalent of "ringback" that would probably do? I'm using version 1.2.9 for this testing Any help would be appreciated, maybe i'm just misunderstanding "the proper way" to handle this event :) Thanks, Callum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/eff22f76/attachment.html From iland at cs.ucsb.edu Wed Oct 16 22:04:21 2013 From: iland at cs.ucsb.edu (Danny Iland) Date: Wed, 16 Oct 2013 11:04:21 -0700 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen Message-ID: I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers. When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase. 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated. Thanks, Danny Iland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/56bb7f08/attachment.html From vbvbrj at gmail.com Wed Oct 16 23:40:14 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 16 Oct 2013 22:40:14 +0300 Subject: [Freeswitch-users] What kind of attack is this? In-Reply-To: <525C3273.3080509@gmail.com> References: <525C3273.3080509@gmail.com> Message-ID: <525EEB9E.7080301@gmail.com> On 14.10.2013 21:05, Mimiko wrote: > 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [NOTICE] > switch_channel.c:1034 New Channel sofia/internal_A.B.C.D/100 at A.B.C.D > [35a84dc8-0a11-449a-9a81-aa0a6ad75ab6] > 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] > switch_core_session.c:1010 Send signal > sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] > 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] > switch_core_session.c:1010 Send signal > sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] > 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] > switch_core_state_machine.c:418 (sofia/internal_A.B.C.D/100 at A.B.C.D) > Running State Change CS_NEW > 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.690475 [DEBUG] > switch_core_state_machine.c:436 (sofia/internal_A.B.C.D/100 at A.B.C.D) > State NEW > 35a84dc8-0a11-449a-9a81-aa0a6ad75ab6 2013-10-14 20:48:51.810383 [DEBUG] > switch_core_session.c:1010 Send signal > sofia/internal_A.B.C.D/100 at A.B.C.D [BREAK] I did found that this logs were due to an attacker trying to hack the system and setting bad invite packets with 19 mins interval. Its hard to detect and block such attacks, event for Fail2ban, because the log does not contain remote IP from which the packet came. It were great for FS to log IP not only for authentication failures, but also for calls. -- Mimiko desu. From wstephen80 at gmail.com Thu Oct 17 00:13:51 2013 From: wstephen80 at gmail.com (Stephen Wilde) Date: Wed, 16 Oct 2013 22:13:51 +0200 Subject: [Freeswitch-users] Disable core dump Message-ID: There is a way to run FreeSwitch with core dumping disabled? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/1705abb3/attachment.html From nneul at mst.edu Thu Oct 17 00:18:16 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 16 Oct 2013 15:18:16 -0500 Subject: [Freeswitch-users] Disable core dump In-Reply-To: References: Message-ID: <525EF488.4030700@mst.edu> ulimit -c 0 before running freeswitch or specify a user for freeswitch to run as, at least on linux, by default, processes with a different effective uid will not dump core by default. -- Nathan On 10/16/2013 03:13 PM, Stephen Wilde wrote: > There is a way to run FreeSwitch with core dumping disabled? > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From wstephen80 at gmail.com Thu Oct 17 00:26:43 2013 From: wstephen80 at gmail.com (Stephen Wilde) Date: Wed, 16 Oct 2013 22:26:43 +0200 Subject: [Freeswitch-users] Disable core dump In-Reply-To: <525EF488.4030700@mst.edu> References: <525EF488.4030700@mst.edu> Message-ID: I have tried with ulimit and I can see with "ulimit -a": core file size (blocks, -c) 0 then I run FreeSwitch but if I do "cat /proc/freeswitch_pid/limits" I see: Limit Soft Limit Hard Limit Units Max core file size unlimited unlimited bytes On Wed, Oct 16, 2013 at 10:18 PM, Nathan Neulinger wrote: > > ulimit -c 0 > > before running freeswitch or specify a user for freeswitch to run as, at > least on linux, by default, processes with a different effective uid will > not dump core by default. > > -- Nathan > > > On 10/16/2013 03:13 PM, Stephen Wilde wrote: > >> There is a way to run FreeSwitch with core dumping disabled? >> >> > -- > ------------------------------**------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/4ab4887d/attachment.html From anthony.minessale at gmail.com Thu Oct 17 00:32:12 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Oct 2013 15:32:12 -0500 Subject: [Freeswitch-users] Disable core dump In-Reply-To: References: <525EF488.4030700@mst.edu> Message-ID: switch.conf.xml has a dump-cores param as well On Wed, Oct 16, 2013 at 3:26 PM, Stephen Wilde wrote: > I have tried with ulimit and I can see with "ulimit -a": > > core file size (blocks, -c) 0 > > then I run FreeSwitch but if I do "cat /proc/freeswitch_pid/limits" I see: > > Limit Soft Limit Hard Limit Units > Max core file size unlimited unlimited bytes > > > On Wed, Oct 16, 2013 at 10:18 PM, Nathan Neulinger wrote: > >> >> ulimit -c 0 >> >> before running freeswitch or specify a user for freeswitch to run as, at >> least on linux, by default, processes with a different effective uid will >> not dump core by default. >> >> -- Nathan >> >> >> On 10/16/2013 03:13 PM, Stephen Wilde wrote: >> >>> There is a way to run FreeSwitch with core dumping disabled? >>> >>> >> -- >> ------------------------------**------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/7dee88a2/attachment.html From nneul at mst.edu Thu Oct 17 00:32:28 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 16 Oct 2013 15:32:28 -0500 Subject: [Freeswitch-users] Disable core dump In-Reply-To: References: <525EF488.4030700@mst.edu> Message-ID: <525EF7DC.4080703@mst.edu> It looks like there is also a 'dump-cores' option in freeswitch config/code that causes FS to raise the limits - that may be what you're seeing. Check in conf/autolog_configs/switch.conf.xml. -- Nathan On 10/16/2013 03:26 PM, Stephen Wilde wrote: > I have tried with ulimit and I can see with "ulimit -a": > > core file size (blocks, -c) 0 > > then I run FreeSwitch but if I do "cat /proc/freeswitch_pid/limits" I see: > > Limit Soft Limit Hard Limit Units > Max core file size unlimited unlimited bytes > > > On Wed, Oct 16, 2013 at 10:18 PM, Nathan Neulinger > wrote: > > > ulimit -c 0 > > before running freeswitch or specify a user for freeswitch to run as, at least on linux, by default, processes with > a different effective uid will not dump core by default. > > -- Nathan > > > On 10/16/2013 03:13 PM, Stephen Wilde wrote: > > There is a way to run FreeSwitch with core dumping disabled? > > > -- > ------------------------------__------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From wstephen80 at gmail.com Thu Oct 17 00:36:08 2013 From: wstephen80 at gmail.com (Stephen Wilde) Date: Wed, 16 Oct 2013 22:36:08 +0200 Subject: [Freeswitch-users] Disable core dump In-Reply-To: <525EF7DC.4080703@mst.edu> References: <525EF488.4030700@mst.edu> <525EF7DC.4080703@mst.edu> Message-ID: Thank you! There was in my switch.conf.xml the line: On Wed, Oct 16, 2013 at 10:32 PM, Nathan Neulinger wrote: > It looks like there is also a 'dump-cores' option in freeswitch > config/code that causes FS to raise the limits - that may be what you're > seeing. Check in conf/autolog_configs/switch.**conf.xml. > > -- Nathan > > > On 10/16/2013 03:26 PM, Stephen Wilde wrote: > >> I have tried with ulimit and I can see with "ulimit -a": >> >> core file size (blocks, -c) 0 >> >> then I run FreeSwitch but if I do "cat /proc/freeswitch_pid/limits" I see: >> >> Limit Soft Limit Hard Limit Units >> Max core file size unlimited unlimited bytes >> >> >> On Wed, Oct 16, 2013 at 10:18 PM, Nathan Neulinger > nneul at mst.edu>> wrote: >> >> >> ulimit -c 0 >> >> before running freeswitch or specify a user for freeswitch to run as, >> at least on linux, by default, processes with >> a different effective uid will not dump core by default. >> >> -- Nathan >> >> >> On 10/16/2013 03:13 PM, Stephen Wilde wrote: >> >> There is a way to run FreeSwitch with core dumping disabled? >> >> >> -- >> ------------------------------**__----------------------------**-- >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> >> > -- > ------------------------------**------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/a6420cf4/attachment-0001.html From michel at targointernet.com Thu Oct 17 00:38:33 2013 From: michel at targointernet.com (Michel Blais - Support technique - Targo communication) Date: Wed, 16 Oct 2013 16:38:33 -0400 Subject: [Freeswitch-users] Build error stop at vdbeaux.c Message-ID: <525EF949.2050700@targointernet.com> Hi, I have a build error blocking at vdbeaux.c, build output following at bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a Cavium octeon+ 50XX. I was not able to find anything on this build error. It would be appreciated if anybody could help. Thanks Michel # make config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk config.status: creating build/pkg/pkginfo config.status: creating apr-1-config config.status: WARNING: 'apr-config.in' seems to ignore the --datarootdir settg config.status: creating apr.pc config.status: creating test/Makefile config.status: creating test/internal/Makefile config.status: creating include/arch/unix/apr_private.h config.status: executing libtool commands rm: can't remove 'libtoolT': No such file or directory config.status: executing default commands config.status: include/apr.h is unchanged config.status: include/arch/unix/apr_private.h is unchanged touch src/include/switch.h make[1]: Entering directory `/usr/src/freeswitch/libs/apr' make[2]: Entering directory `/usr/src/freeswitch/libs/apr' make[2]: Nothing to be done for `local-all'. make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' touch src/include/switch.h make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' Making all in xml/expat make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util/xml/expat' make[3]: Entering directory `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' make[3]: Nothing to be done for `all'. make[3]: Leaving directory `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util/xml/expat' make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' make[2]: Nothing to be done for `local-all'. make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' touch src/include/switch.h make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c ./mkkeywordhash >keywordhash.h ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1c libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 -DHAVE_FDATASYo ./src/vdbeaux.c: In function 'opcodeNoPush': ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in this funct) ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported only once ./src/vdbeaux.c:247: error: for each function it appears in.) ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in this funct) ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in this funct) ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in this funct) ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in this funct) ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in this funct) ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in this funct) ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in this funct) ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in this funct) ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in this funct) make[1]: *** [vdbeaux.lo] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' make: *** [libs/sqlite/libsqlite3.la] Error 2 From mike at jerris.com Thu Oct 17 00:50:07 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Oct 2013 13:50:07 -0700 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: <525EF949.2050700@targointernet.com> References: <525EF949.2050700@targointernet.com> Message-ID: libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d 0x%04x\n", i, nopush[i] libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c $(TOP)/mkopcodeh.awk libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | $(NAWK) -f $(TOP)/mkopcodeh.awk >opcodes.h <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h #define NOPUSH_MASK_0 0x3fbc #define NOPUSH_MASK_1 0x3e5b #define NOPUSH_MASK_2 0x71ef #define NOPUSH_MASK_3 0x7fce #define NOPUSH_MASK_4 0xffff #define NOPUSH_MASK_5 0xcdef #define NOPUSH_MASK_6 0xb6d7 #define NOPUSH_MASK_7 0x33af #define NOPUSH_MASK_8 0xf86f #define NOPUSH_MASK_9 0x0000 take a look at the contents of your generated opcodes.h and see what you find in there. The logic would be in mkopcodeh.awk. Also possible you don't have awk installed? check out the definition of NAWK var: <11>:grep -R ^NAWK Makefile Makefile:NAWK = awk Mike On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo communication wrote: > Hi, > > I have a build error blocking at vdbeaux.c, build output following at > bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a > Cavium octeon+ 50XX. I was not able to find anything on this build > error. It would be appreciated if anybody could help. > > Thanks > Michel > > # make > config.status: creating Makefile > config.status: creating include/apr.h > config.status: creating build/apr_rules.mk > config.status: creating build/pkg/pkginfo > config.status: creating apr-1-config > config.status: WARNING: 'apr-config.in' seems to ignore the > --datarootdir settg > config.status: creating apr.pc > config.status: creating test/Makefile > config.status: creating test/internal/Makefile > config.status: creating include/arch/unix/apr_private.h > config.status: executing libtool commands > rm: can't remove 'libtoolT': No such file or directory > config.status: executing default commands > config.status: include/apr.h is unchanged > config.status: include/arch/unix/apr_private.h is unchanged > touch src/include/switch.h > make[1]: Entering directory `/usr/src/freeswitch/libs/apr' > make[2]: Entering directory `/usr/src/freeswitch/libs/apr' > make[2]: Nothing to be done for `local-all'. > make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' > make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' > touch src/include/switch.h > make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' > Making all in xml/expat > make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util/xml/expat' > make[3]: Entering directory > `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > make[3]: Nothing to be done for `all'. > make[3]: Leaving directory > `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util/xml/expat' > make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' > make[2]: Nothing to be done for `local-all'. > make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > touch src/include/switch.h > make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c > ./mkkeywordhash >keywordhash.h > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > -DHAVE_USLEEP=1c > libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > -DHAVE_FDATASYo > ./src/vdbeaux.c: In function 'opcodeNoPush': > ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in > this funct) > ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported only > once > ./src/vdbeaux.c:247: error: for each function it appears in.) > ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in > this funct) > ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in > this funct) > ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in > this funct) > ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in > this funct) > ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in > this funct) > ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in > this funct) > ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in > this funct) > ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in > this funct) > ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in > this funct) > make[1]: *** [vdbeaux.lo] Error 1 > make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' > make: *** [libs/sqlite/libsqlite3.la] Error 2 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From michel at targointernet.com Thu Oct 17 01:01:42 2013 From: michel at targointernet.com (Michel Blais - Support technique - Targo communication) Date: Wed, 16 Oct 2013 17:01:42 -0400 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: References: <525EF949.2050700@targointernet.com> Message-ID: <525EFEB6.2090607@targointernet.com> The awk command is the one include by busybox, same for grep. Could it be the problem ? # grep -R ^NAWK Makefile grep: invalid option -- 'R' BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. Thanks Le 2013-10-16 16:50, Michael Jerris a ?crit : > libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d 0x%04x\n", i, nopush[i] > > libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c $(TOP)/mkopcodeh.awk > libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | $(NAWK) -f $(TOP)/mkopcodeh.awk >opcodes.h > > <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h > #define NOPUSH_MASK_0 0x3fbc > #define NOPUSH_MASK_1 0x3e5b > #define NOPUSH_MASK_2 0x71ef > #define NOPUSH_MASK_3 0x7fce > #define NOPUSH_MASK_4 0xffff > #define NOPUSH_MASK_5 0xcdef > #define NOPUSH_MASK_6 0xb6d7 > #define NOPUSH_MASK_7 0x33af > #define NOPUSH_MASK_8 0xf86f > #define NOPUSH_MASK_9 0x0000 > > take a look at the contents of your generated opcodes.h and see what you find in there. The logic would be in mkopcodeh.awk. Also possible you don't have awk installed? check out the definition of NAWK var: > > <11>:grep -R ^NAWK Makefile > Makefile:NAWK = awk > > Mike > > > On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo communication wrote: > >> Hi, >> >> I have a build error blocking at vdbeaux.c, build output following at >> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a >> Cavium octeon+ 50XX. I was not able to find anything on this build >> error. It would be appreciated if anybody could help. >> >> Thanks >> Michel >> >> # make >> config.status: creating Makefile >> config.status: creating include/apr.h >> config.status: creating build/apr_rules.mk >> config.status: creating build/pkg/pkginfo >> config.status: creating apr-1-config >> config.status: WARNING: 'apr-config.in' seems to ignore the >> --datarootdir settg >> config.status: creating apr.pc >> config.status: creating test/Makefile >> config.status: creating test/internal/Makefile >> config.status: creating include/arch/unix/apr_private.h >> config.status: executing libtool commands >> rm: can't remove 'libtoolT': No such file or directory >> config.status: executing default commands >> config.status: include/apr.h is unchanged >> config.status: include/arch/unix/apr_private.h is unchanged >> touch src/include/switch.h >> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' >> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' >> make[2]: Nothing to be done for `local-all'. >> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' >> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' >> touch src/include/switch.h >> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' >> Making all in xml/expat >> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util/xml/expat' >> make[3]: Entering directory >> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' >> make[3]: Nothing to be done for `all'. >> make[3]: Leaving directory >> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' >> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util/xml/expat' >> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' >> make[2]: Nothing to be done for `local-all'. >> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' >> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' >> touch src/include/switch.h >> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c >> ./mkkeywordhash >keywordhash.h >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1c >> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >> -DHAVE_FDATASYo >> ./src/vdbeaux.c: In function 'opcodeNoPush': >> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported only >> once >> ./src/vdbeaux.c:247: error: for each function it appears in.) >> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in >> this funct) >> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in >> this funct) >> make[1]: *** [vdbeaux.lo] Error 1 >> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' >> make: *** [libs/sqlite/libsqlite3.la] Error 2 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Oct 17 01:11:13 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Oct 2013 14:11:13 -0700 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: <525EFEB6.2090607@targointernet.com> References: <525EF949.2050700@targointernet.com> <525EFEB6.2090607@targointernet.com> Message-ID: <97C88A31-2BC6-4B6F-995E-76752E224296@jerris.com> Not sure why your grep is funky.. but the -R isn't necessary? This is going to be limitations of the tools in busybox I suspect. grep NAWK Makefile ? On Oct 16, 2013, at 2:01 PM, Michel Blais - Support technique - Targo communication wrote: > The awk command is the one include by busybox, same for grep. Could it > be the problem ? > > # grep -R ^NAWK Makefile > grep: invalid option -- 'R' > BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. > > Thanks > > Le 2013-10-16 16:50, Michael Jerris a ?crit : >> libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d 0x%04x\n", i, nopush[i] >> >> libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c $(TOP)/mkopcodeh.awk >> libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | $(NAWK) -f $(TOP)/mkopcodeh.awk >opcodes.h >> >> <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h >> #define NOPUSH_MASK_0 0x3fbc >> #define NOPUSH_MASK_1 0x3e5b >> #define NOPUSH_MASK_2 0x71ef >> #define NOPUSH_MASK_3 0x7fce >> #define NOPUSH_MASK_4 0xffff >> #define NOPUSH_MASK_5 0xcdef >> #define NOPUSH_MASK_6 0xb6d7 >> #define NOPUSH_MASK_7 0x33af >> #define NOPUSH_MASK_8 0xf86f >> #define NOPUSH_MASK_9 0x0000 >> >> take a look at the contents of your generated opcodes.h and see what you find in there. The logic would be in mkopcodeh.awk. Also possible you don't have awk installed? check out the definition of NAWK var: >> >> <11>:grep -R ^NAWK Makefile >> Makefile:NAWK = awk >> >> Mike >> >> >> On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo communication wrote: >> >>> Hi, >>> >>> I have a build error blocking at vdbeaux.c, build output following at >>> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a >>> Cavium octeon+ 50XX. I was not able to find anything on this build >>> error. It would be appreciated if anybody could help. >>> >>> Thanks >>> Michel >>> >>> # make >>> config.status: creating Makefile >>> config.status: creating include/apr.h >>> config.status: creating build/apr_rules.mk >>> config.status: creating build/pkg/pkginfo >>> config.status: creating apr-1-config >>> config.status: WARNING: 'apr-config.in' seems to ignore the >>> --datarootdir settg >>> config.status: creating apr.pc >>> config.status: creating test/Makefile >>> config.status: creating test/internal/Makefile >>> config.status: creating include/arch/unix/apr_private.h >>> config.status: executing libtool commands >>> rm: can't remove 'libtoolT': No such file or directory >>> config.status: executing default commands >>> config.status: include/apr.h is unchanged >>> config.status: include/arch/unix/apr_private.h is unchanged >>> touch src/include/switch.h >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' >>> make[2]: Nothing to be done for `local-all'. >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' >>> touch src/include/switch.h >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' >>> Making all in xml/expat >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util/xml/expat' >>> make[3]: Entering directory >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' >>> make[3]: Nothing to be done for `all'. >>> make[3]: Leaving directory >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util/xml/expat' >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' >>> make[2]: Nothing to be done for `local-all'. >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' >>> touch src/include/switch.h >>> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c >>> ./mkkeywordhash >keywordhash.h >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >>> -DHAVE_USLEEP=1c >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 >>> -DHAVE_FDATASYo >>> ./src/vdbeaux.c: In function 'opcodeNoPush': >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported only >>> once >>> ./src/vdbeaux.c:247: error: for each function it appears in.) >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in >>> this funct) >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in >>> this funct) >>> make[1]: *** [vdbeaux.lo] Error 1 >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' >>> make: *** [libs/sqlite/libsqlite3.la] Error 2 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jdodson at acm.org Thu Oct 17 01:28:52 2013 From: jdodson at acm.org (Joel Dodson) Date: Wed, 16 Oct 2013 14:28:52 -0700 Subject: [Freeswitch-users] ftp-proxy interferes with event_socket on mac os x 10.8.5 Message-ID: Hi, First off, thank you Mario for the great documentation for installing freeswitch on Mac OS X. It was very helpful. I ran into a problem starting freeswitch locally on my mac. The problem was port 8021 was in use (default event_socket port). I had some misdirections figuring out what was using 8021 so I thought I'd write up a few notes with my experience (I've included the stupid things I did hoping others can learn from it) and maybe they can be posted on wiki in the trouble shooting section (or maybe they are already and I missed it :) ). The first thing I did was run netstat: netstat -an | grep 8021 output: tcp6 0 0 fe80::1%lo0.8021 *.* LISTEN tcp4 0 0 127.0.0.1.8021 *.* LISTEN tcp6 0 0 ::1.8021 *.* LISTEN and see that yes, something else was using port 8021. But how do I get the PID? Got to love stackoverflow: http://stackoverflow.com/questions/4421633/who-is-listening-on-a-given-tcp-port-on-mac-os-x sudo lsof -iTCP:8021 -sTCP:LISTEN I ran that to find it's launchd, output: COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME launchd 1 root 83u IPv6 0x6977044a72cceb65 0t0 TCP localhost:intu-ec-client (LISTEN) launchd 1 root 84u IPv4 0x6977044a74952245 0t0 TCP localhost:intu-ec-client (LISTEN) launchd 1 root 85u IPv6 0x6977044a72cce785 0t0 TCP localhost:intu-ec-client (LISTEN) here's where my big detour started. I wondered, what is intu-ec-client? Well, if you google it, you'll find the first several hits are people saying their ios devices have been hacked. So I thought, oh ****, my laptop has been hacked (though it's OS X 10.8.5, not ios). I poked around some more, ran a complete virus checker, a lot more googling... What I've finally concluded is the only reason intu-ec-client is listed there is because 8021 is registered with IANA as the default port for some intuit protocol and lsof is doing port number to name conversion. Turns out -P is used in lsof to not convert port numbers, -n is only for not converting address. Once I decided my system was probably not hacked, and it's just a coincidence with the intu-ec-client port, I set out to figure out what really is using that port. I'm not a systems expert by any stretch so others reading this might think, what a dumbass, why didn't you do that in the first place. Though considering it's launchd, with PID 1 and user root using that port, I guess I panicked thinking I'd better get this resolved while my system is still working... Anyway, after googling around some more and learning something about launchd, I tried to telnet to the port (another suggestion from stackoverflow which in hind sight I should have done immediately) to see what was running there. I found there's an ftp-proxy running there. Okay, so why is there an ftp-proxy running there? I've been running FS on this laptop for several weeks and haven't run into this before. I checked all the programs that start automatically and removed some that I thought might possibly be starting an ftp-proxy (though I couldn't imagine why they would be). Rebooted and still there's that ftp-proxy. I guess, again in hindsight, none of those would have launched as root via launchd. >From the launchd man page, I found the configurations are in: ~/Library/LaunchAgents Per-user agents provided by the user. /Library/LaunchAgents Per-user agents provided by the administrator. /Library/LaunchDaemons System-wide daemons provided by the administrator. /System/Library/LaunchAgents Per-user agents provided by Mac OS X. /System/Library/LaunchDaemons System-wide daemons provided by Mac OS X. And from that I found the ftp-proxy in: /System/Library/LaunchDaemons/com.apple.ftp-proxy.plist which, sure enough, has localhost and 8021 defined as a listener. What I'm still wondering is why I hadn't hit that before. In the last few weeks, I haven't added an ftp-proxy. And that ftp-proxy.plist file is dated July, 2012. I did recently launch the apple installed version of apache on my laptop. I'm suspicious that also added the ftp-proxy to launchd. I probably should look into that but for now, I'm comfortable my system has not been hacked, I understand why that port is in use so I'll just change the setting in event_socket.conf.xml and get back to work :) thanks, Joel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/9eab82ae/attachment.html From alipey at gmail.com Thu Oct 17 02:59:12 2013 From: alipey at gmail.com (Ali Pey) Date: Wed, 16 Oct 2013 18:59:12 -0400 Subject: [Freeswitch-users] Voicemail beep Message-ID: Hello, Is there a way to change the voicemail beep sound? The existing beep is too long and too loud. Thanks, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/e047be39/attachment-0001.html From grcamauer at gmail.com Thu Oct 17 03:11:03 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 16 Oct 2013 20:11:03 -0300 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: You can only use the -l switch once. Just separate the modules you want to load with commas: fs_encode -v -l mod_spandsp , mod_com_g729 call.G729 call.wav On Tue, Oct 15, 2013 at 4:47 PM, Dmitry Lysenko wrote: > I checked right now, and it work. (git, 1.2.stable, Oct,11) > How about other codecs? Using 'sudo'? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/ba40ffb2/attachment.html From ssa1357 at yahoo.com Thu Oct 17 03:13:37 2013 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Wed, 16 Oct 2013 16:13:37 -0700 (PDT) Subject: [Freeswitch-users] mod_spandsp compile error Message-ID: <1381965217.28081.YahooMailNeo@web121303.mail.ne1.yahoo.com> Hi, I am try to install a FreeSWITCH stable version 1.2.13 on a fresh CentOS. I installed all prereqired packages as they are in wiki, configure goes well but while I try to do make I got this error on mod_spandsp. making all mod_spandsp Creating mod_spandsp_la-mod_spandsp.lo mkdir .libs Compiling mod_spandsp.c ... Creating mod_spandsp_la-udptl.lo Compiling udptl.c ... cc1: warnings being treated as errors udptl.c: In function 'udptl_rx_packet': udptl.c:174: warning: 'data' may be used uninitialized in this function udptl.c:173: warning: 'msg' may be used uninitialized in this function make[3]: *** [mod_spandsp_la-udptl.lo] Error 1 make[2]: *** [mod_spandsp-all] Error 1 make[1]: *** [mod_spandsp] Error 2 make: *** [mod_spandsp] Error 2 Would somebody please help me with that? Best Regards Sadjad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/f3d72690/attachment.html From mario_fs at mgtech.com Thu Oct 17 03:14:54 2013 From: mario_fs at mgtech.com (Mario G) Date: Wed, 16 Oct 2013 16:14:54 -0700 Subject: [Freeswitch-users] ftp-proxy interferes with event_socket on mac os x 10.8.5 In-Reply-To: References: Message-ID: <7BCCE38C-06C2-4899-8FB7-50D2DF41C3CA@mgtech.com> Your welcome, I am thinking of adding a page for debugging and/or current issues/gotchas, your comments would be useful. BTW, if you look at my startup script http://wiki.freeswitch.org/wiki/Installation_and_Setup_on_OS_X#Create_the_FreeSWITCH.E2.84.A2_Command_Script you can find how I got the correct PID. Your should also see http://jira.freeswitch.org/browse/FS-5223#comment-45367 which is still an issue which is why the script has to get the PID. Mario G On Oct 16, 2013, at 2:28 PM, Joel Dodson wrote: > Hi, > > First off, thank you Mario for the great documentation for installing freeswitch on Mac OS X. It was very helpful. > > I ran into a problem starting freeswitch locally on my mac. The problem was port 8021 was in use (default event_socket port). I had some misdirections figuring out what was using 8021 so I thought I'd write up a few notes with my experience (I've included the stupid things I did hoping others can learn from it) and maybe they can be posted on wiki in the trouble shooting section (or maybe they are already and I missed it :) ). > > The first thing I did was run netstat: > > netstat -an | grep 8021 > > output: > > tcp6 0 0 fe80::1%lo0.8021 *.* LISTEN > tcp4 0 0 127.0.0.1.8021 *.* LISTEN > tcp6 0 0 ::1.8021 *.* LISTEN > > and see that yes, something else was using port 8021. But how do I get the PID? Got to love stackoverflow: > > http://stackoverflow.com/questions/4421633/who-is-listening-on-a-given-tcp-port-on-mac-os-x > > sudo lsof -iTCP:8021 -sTCP:LISTEN > > I ran that to find it's launchd, output: > > COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME > launchd 1 root 83u IPv6 0x6977044a72cceb65 0t0 TCP localhost:intu-ec-client (LISTEN) > launchd 1 root 84u IPv4 0x6977044a74952245 0t0 TCP localhost:intu-ec-client (LISTEN) > launchd 1 root 85u IPv6 0x6977044a72cce785 0t0 TCP localhost:intu-ec-client (LISTEN) > > here's where my big detour started. I wondered, what is intu-ec-client? Well, if you google it, you'll find the first several hits are people saying their ios devices have been hacked. So I thought, oh ****, my laptop has been hacked (though it's OS X 10.8.5, not ios). I poked around some more, ran a complete virus checker, a lot more googling... > > What I've finally concluded is the only reason intu-ec-client is listed there is because 8021 is registered with IANA as the default port for some intuit protocol and lsof is doing port number to name conversion. Turns out -P is used in lsof to not convert port numbers, -n is only for not converting address. > > Once I decided my system was probably not hacked, and it's just a coincidence with the intu-ec-client port, I set out to figure out what really is using that port. I'm not a systems expert by any stretch so others reading this might think, what a dumbass, why didn't you do that in the first place. > > Though considering it's launchd, with PID 1 and user root using that port, I guess I panicked thinking I'd better get this resolved while my system is still working... > > Anyway, after googling around some more and learning something about launchd, I tried to telnet to the port (another suggestion from stackoverflow which in hind sight I should have done immediately) to see what was running there. I found there's an ftp-proxy running there. > > Okay, so why is there an ftp-proxy running there? I've been running FS on this laptop for several weeks and haven't run into this before. > > I checked all the programs that start automatically and removed some that I thought might possibly be starting an ftp-proxy (though I couldn't imagine why they would be). Rebooted and still there's that ftp-proxy. I guess, again in hindsight, none of those would have launched as root via launchd. > > From the launchd man page, I found the configurations are in: > > ~/Library/LaunchAgents Per-user agents provided by the user. > /Library/LaunchAgents Per-user agents provided by the administrator. > /Library/LaunchDaemons System-wide daemons provided by the administrator. > /System/Library/LaunchAgents Per-user agents provided by Mac OS X. > /System/Library/LaunchDaemons System-wide daemons provided by Mac OS X. > > And from that I found the ftp-proxy in: > > /System/Library/LaunchDaemons/com.apple.ftp-proxy.plist > > which, sure enough, has localhost and 8021 defined as a listener. > > What I'm still wondering is why I hadn't hit that before. In the last few weeks, I haven't added an ftp-proxy. And that ftp-proxy.plist file is dated July, 2012. I did recently launch the apple installed version of apache on my laptop. I'm suspicious that also added the ftp-proxy to launchd. I probably should look into that but for now, I'm comfortable my system has not been hacked, I understand why that port is in use so I'll just change the setting in event_socket.conf.xml and get back to work :) > > thanks, > Joel > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/a49124d8/attachment.html From max at nysolutions.com Thu Oct 17 04:51:00 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 17 Oct 2013 00:51:00 +0000 Subject: [Freeswitch-users] Voicemail beep In-Reply-To: References: Message-ID: Just take they .wav file and modify it with a sound file editor. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Wednesday, October 16, 2013 6:59 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Voicemail beep Hello, Is there a way to change the voicemail beep sound? The existing beep is too long and too loud. Thanks, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a5234a29/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a5234a29/attachment-0001.jpg From michel at targointernet.com Thu Oct 17 05:05:43 2013 From: michel at targointernet.com (Michel Blais) Date: Wed, 16 Oct 2013 21:05:43 -0400 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: <97C88A31-2BC6-4B6F-995E-76752E224296@jerris.com> References: <525EF949.2050700@targointernet.com> <525EFEB6.2090607@targointernet.com> <97C88A31-2BC6-4B6F-995E-76752E224296@jerris.com> Message-ID: Busybox don't include a full version of every command. It's a shell that include most commande with only the most used option but often lack usefull feature. It's use to save space on embedded device. I didn't know the -R option so I was not sur about what you where looking for but the're no line with NAWK in the make file but if I look for AWK, I get this : # grep AWK Makefile $(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " $$1; \ AWK = awk $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1 } \ $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1; } \ $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ Thanks 2013/10/16 Michael Jerris > Not sure why your grep is funky.. but the -R isn't necessary? This is > going to be limitations of the tools in busybox I suspect. > > grep NAWK Makefile > > ? > > On Oct 16, 2013, at 2:01 PM, Michel Blais - Support technique - Targo > communication wrote: > > > The awk command is the one include by busybox, same for grep. Could it > > be the problem ? > > > > # grep -R ^NAWK Makefile > > grep: invalid option -- 'R' > > BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. > > > > Thanks > > > > Le 2013-10-16 16:50, Michael Jerris a ?crit : > >> libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d > 0x%04x\n", i, nopush[i] > >> > >> libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c > $(TOP)/mkopcodeh.awk > >> libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | $(NAWK) > -f $(TOP)/mkopcodeh.awk >opcodes.h > >> > >> <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h > >> #define NOPUSH_MASK_0 0x3fbc > >> #define NOPUSH_MASK_1 0x3e5b > >> #define NOPUSH_MASK_2 0x71ef > >> #define NOPUSH_MASK_3 0x7fce > >> #define NOPUSH_MASK_4 0xffff > >> #define NOPUSH_MASK_5 0xcdef > >> #define NOPUSH_MASK_6 0xb6d7 > >> #define NOPUSH_MASK_7 0x33af > >> #define NOPUSH_MASK_8 0xf86f > >> #define NOPUSH_MASK_9 0x0000 > >> > >> take a look at the contents of your generated opcodes.h and see what > you find in there. The logic would be in mkopcodeh.awk. Also possible you > don't have awk installed? check out the definition of NAWK var: > >> > >> <11>:grep -R ^NAWK Makefile > >> Makefile:NAWK = awk > >> > >> Mike > >> > >> > >> On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo > communication wrote: > >> > >>> Hi, > >>> > >>> I have a build error blocking at vdbeaux.c, build output following at > >>> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a > >>> Cavium octeon+ 50XX. I was not able to find anything on this build > >>> error. It would be appreciated if anybody could help. > >>> > >>> Thanks > >>> Michel > >>> > >>> # make > >>> config.status: creating Makefile > >>> config.status: creating include/apr.h > >>> config.status: creating build/apr_rules.mk > >>> config.status: creating build/pkg/pkginfo > >>> config.status: creating apr-1-config > >>> config.status: WARNING: 'apr-config.in' seems to ignore the > >>> --datarootdir settg > >>> config.status: creating apr.pc > >>> config.status: creating test/Makefile > >>> config.status: creating test/internal/Makefile > >>> config.status: creating include/arch/unix/apr_private.h > >>> config.status: executing libtool commands > >>> rm: can't remove 'libtoolT': No such file or directory > >>> config.status: executing default commands > >>> config.status: include/apr.h is unchanged > >>> config.status: include/arch/unix/apr_private.h is unchanged > >>> touch src/include/switch.h > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' > >>> make[2]: Nothing to be done for `local-all'. > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' > >>> touch src/include/switch.h > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' > >>> Making all in xml/expat > >>> make[2]: Entering directory > `/usr/src/freeswitch/libs/apr-util/xml/expat' > >>> make[3]: Entering directory > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > >>> make[3]: Nothing to be done for `all'. > >>> make[3]: Leaving directory > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > >>> make[2]: Leaving directory > `/usr/src/freeswitch/libs/apr-util/xml/expat' > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' > >>> make[2]: Nothing to be done for `local-all'. > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > >>> touch src/include/switch.h > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c > >>> ./mkkeywordhash >keywordhash.h > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./src/vdbeaux.c: In function 'opcodeNoPush': > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported > only > >>> once > >>> ./src/vdbeaux.c:247: error: for each function it appears in.) > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in > >>> this funct) > >>> make[1]: *** [vdbeaux.lo] Error 1 > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' > >>> make: *** [libs/sqlite/libsqlite3.la] Error 2 > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/441ec66f/attachment-0001.html From steveayre at gmail.com Thu Oct 17 05:13:53 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Oct 2013 02:13:53 +0100 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: Same happens when only using a single '-l mod_com_g729' arg. Also does anyone know a way to run this without having to be root? Strace shows the reason is it tries to open the /etc/freeswitch configs and write them to /var/log/freeswitch/freeswitch.xml.fsxml (which needs the freeswitch or root user). -Steve On 17 October 2013 00:11, Guillermo Ruiz Camauer wrote: > You can only use the -l switch once. Just separate the modules you want > to load with commas: > > fs_encode -v -l mod_spandsp , mod_com_g729 call.G729 call.wav > > > On Tue, Oct 15, 2013 at 4:47 PM, Dmitry Lysenko > wrote: > >> I checked right now, and it work. (git, 1.2.stable, Oct,11) >> How about other codecs? Using 'sudo'? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/acf2c9f7/attachment.html From steveayre at gmail.com Thu Oct 17 05:14:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Oct 2013 02:14:52 +0100 Subject: [Freeswitch-users] fs_encode to convert G729 to WAV In-Reply-To: References: Message-ID: Ok for some reason today it works! I did just update to 1.2.14 though and reboot though, so perhaps something changed there. :) Does anyone know how to run the program as any user? On 17 October 2013 02:13, Steven Ayre wrote: > Same happens when only using a single '-l mod_com_g729' arg. > > Also does anyone know a way to run this without having to be root? > > Strace shows the reason is it tries to open the /etc/freeswitch configs > and write them to /var/log/freeswitch/freeswitch.xml.fsxml (which needs the > freeswitch or root user). > > -Steve > > > On 17 October 2013 00:11, Guillermo Ruiz Camauer wrote: > >> You can only use the -l switch once. Just separate the modules you want >> to load with commas: >> >> fs_encode -v -l mod_spandsp , mod_com_g729 call.G729 call.wav >> >> >> On Tue, Oct 15, 2013 at 4:47 PM, Dmitry Lysenko < >> dvl36.ripe.nick at gmail.com> wrote: >> >>> I checked right now, and it work. (git, 1.2.stable, Oct,11) >>> How about other codecs? Using 'sudo'? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/d7e091e2/attachment.html From andretodd at verizon.net Thu Oct 17 05:18:01 2013 From: andretodd at verizon.net (Andre) Date: Wed, 16 Oct 2013 21:18:01 -0400 Subject: [Freeswitch-users] Change response code Message-ID: <043001cecad6$b9c59970$2d50cc50$@verizon.net> HI, how would I change the response code back to my customer? Let's say I receive a 500 from my provider and I wanted to tell my customer it's a 503, how would I do that? Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131016/ff9bd1a2/attachment.html From nandy1925 at gmail.com Thu Oct 17 06:27:58 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 17 Oct 2013 10:27:58 +0800 Subject: [Freeswitch-users] FreeTDM tones.conf Disconnect Supervision Message-ID: Hello all, Our telco recently changed their call progress tones: Dialtone: 450Hz continuous Ringback tone: 450Hz/1 sec ON 4 sec OFF Busy tone: 450Hz/360ms ON 360 OFF Attn: 950Hz/continuous (0db) The way I understand it, FXO listens for the dialtone then sends DTMF tones. After this, it expects either the Ringback or Busy tone. I read Michael Collin's post that FreeTDM doesn't detect (as of now) cadence. http://permalink.gmane.org/gmane.comp.telephony.freeswitch.user/36531 Questions: 1. What state will FreeTDM go when it detects 450Hz? Ringing or Busy? I presume it would be Ringing or it would hangup. 2. Can I use tone_detect to distinguish them? Can I use a series of tone_detects? Any suggestion where to go? If not ... My last recourse will be - wait for the Attn tone. But ... will FreeTDM jump to hangup the call upon detecting this? Thanks, /Nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/5882b487/attachment.html From jleung at v10networks.ca Thu Oct 17 07:25:02 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Thu, 17 Oct 2013 11:25:02 +0800 Subject: [Freeswitch-users] Build error stop at vdbeaux.c Message-ID: <002501cecae8$7b72277a$7c07000a@smb.curriegrad2004.ca> Install gawk and have that gawk's path take precedence over the default path variable. Michel Blais wrote: Busybox don't include a full version of every command. It's a shell that include most commande with only the most used option but often lack usefull feature. It's use to save space on embedded device. I didn't know the -R option so I was not sur about what you where looking for but the're no line with NAWK in the make file but if I look for AWK, I get this : # grep AWK Makefile $(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " $$1; \ AWK = awk $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1 } \ $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1; } \ $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ Thanks 2013/10/16 Michael Jerris > Not sure why your grep is funky.. but the -R isn't necessary? This is > going to be limitations of the tools in busybox I suspect. > > grep NAWK Makefile > > ? > > On Oct 16, 2013, at 2:01 PM, Michel Blais - Support technique - Targo > communication wrote: > > > The awk command is the one include by busybox, same for grep. Could it > > be the problem ? > > > > # grep -R ^NAWK Makefile > > grep: invalid option -- 'R' > > BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. > > > > Thanks > > > > Le 2013-10-16 16:50, Michael Jerris a ?crit : > >> libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d > 0x%04x\n", i, nopush[i] > >> > >> libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c > $(TOP)/mkopcodeh.awk > >> libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | $(NAWK) > -f $(TOP)/mkopcodeh.awk >opcodes.h > >> > >> <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h > >> #define NOPUSH_MASK_0 0x3fbc > >> #define NOPUSH_MASK_1 0x3e5b > >> #define NOPUSH_MASK_2 0x71ef > >> #define NOPUSH_MASK_3 0x7fce > >> #define NOPUSH_MASK_4 0xffff > >> #define NOPUSH_MASK_5 0xcdef > >> #define NOPUSH_MASK_6 0xb6d7 > >> #define NOPUSH_MASK_7 0x33af > >> #define NOPUSH_MASK_8 0xf86f > >> #define NOPUSH_MASK_9 0x0000 > >> > >> take a look at the contents of your generated opcodes.h and see what > you find in there. The logic would be in mkopcodeh.awk. Also possible you > don't have awk installed? check out the definition of NAWK var: > >> > >> <11>:grep -R ^NAWK Makefile > >> Makefile:NAWK = awk > >> > >> Mike > >> > >> > >> On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo > communication wrote: > >> > >>> Hi, > >>> > >>> I have a build error blocking at vdbeaux.c, build output following at > >>> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a > >>> Cavium octeon+ 50XX. I was not able to find anything on this build > >>> error. It would be appreciated if anybody could help. > >>> > >>> Thanks > >>> Michel > >>> > >>> # make > >>> config.status: creating Makefile > >>> config.status: creating include/apr.h > >>> config.status: creating build/apr_rules.mk > >>> config.status: creating build/pkg/pkginfo > >>> config.status: creating apr-1-config > >>> config.status: WARNING: 'apr-config.in' seems to ignore the > >>> --datarootdir settg > >>> config.status: creating apr.pc > >>> config.status: creating test/Makefile > >>> config.status: creating test/internal/Makefile > >>> config.status: creating include/arch/unix/apr_private.h > >>> config.status: executing libtool commands > >>> rm: can't remove 'libtoolT': No such file or directory > >>> config.status: executing default commands > >>> config.status: include/apr.h is unchanged > >>> config.status: include/arch/unix/apr_private.h is unchanged > >>> touch src/include/switch.h > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' > >>> make[2]: Nothing to be done for `local-all'. > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' > >>> touch src/include/switch.h > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' > >>> Making all in xml/expat > >>> make[2]: Entering directory > `/usr/src/freeswitch/libs/apr-util/xml/expat' > >>> make[3]: Entering directory > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > >>> make[3]: Nothing to be done for `all'. > >>> make[3]: Leaving directory > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > >>> make[2]: Leaving directory > `/usr/src/freeswitch/libs/apr-util/xml/expat' > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' > >>> make[2]: Nothing to be done for `local-all'. > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > >>> touch src/include/switch.h > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c > >>> ./mkkeywordhash >keywordhash.h > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > >>> -DHAVE_USLEEP=1c > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > >>> -DHAVE_FDATASYo > >>> ./src/vdbeaux.c: In function 'opcodeNoPush': > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported > only > >>> once > >>> ./src/vdbeaux.c:247: error: for each function it appears in.) > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in > >>> this funct) > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in > >>> this funct) > >>> make[1]: *** [vdbeaux.lo] Error 1 > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' > >>> make: *** [libs/sqlite/libsqlite3.la] Error 2 > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From smontour at verizon.net Thu Oct 17 07:38:19 2013 From: smontour at verizon.net (Sam Montour) Date: Wed, 16 Oct 2013 22:38:19 -0500 Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <1381908592318-7595737.post@n2.nabble.com> References: <1381849173815-7595714.post@n2.nabble.com> <006f01cec9bf$dc189c60$9449d520$@verizon.net> <1381860480301-7595722.post@n2.nabble.com> <009101cec9d3$fd1d6e90$f7584bb0$@verizon.net> <1381908592318-7595737.post@n2.nabble.com> Message-ID: <008d01cecaea$52b2a340$f817e9c0$@verizon.net> Where are you noting the hang up time? Is it in FS , a test tool, or a network trace? I remember seeing something similar on a test tool. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Samir Doshi Sent: Wednesday, October 16, 2013 2:30 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call hangup not working in perl session That works. Thanks However I notice one thing, If I set 60 seconds sched time then my call hang up at 57 seconds. I tried many times and all the time it hangup at 57 seconds. Not sure if that's something related to fs. Any thoughts? ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-per l-session-tp7595714p7595737.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From alipey at gmail.com Thu Oct 17 08:28:18 2013 From: alipey at gmail.com (Ali Pey) Date: Thu, 17 Oct 2013 00:28:18 -0400 Subject: [Freeswitch-users] Voicemail beep In-Reply-To: References: Message-ID: Hi Moishe, Is there a wav file that is used for voicemail beep? Where is this .wav file? what's its name? Thanks, Ali On Wed, Oct 16, 2013 at 8:51 PM, Moishe Grunstein wrote: > Just take they .wav file and modify it with a sound file editor.**** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Wednesday, October 16, 2013 6:59 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Voicemail beep**** > > ** ** > > Hello,**** > > ** ** > > Is there a way to change the voicemail beep sound?**** > > ** ** > > The existing beep is too long and too loud.**** > > ** ** > > Thanks,**** > > Ali**** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/025556c3/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/025556c3/attachment.jpe From gmaruzz at gmail.com Thu Oct 17 12:00:18 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 17 Oct 2013 10:00:18 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen In-Reply-To: References: Message-ID: mod_skypopen uses signed raw linear 16 bit (eg: SL16) natively, but you seem to force it to use PCMU or PCMA (eg: g711). FreeSWITCH will automatically do the conversion for you (from whatever your channel uses) when bridging to skypopen if you don't force it. -giovanni On Wed, Oct 16, 2013 at 8:04 PM, Danny Iland wrote: > I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile > phones to VoIP providers. > > When bridging a call from a cellular handset to a mod_skypopen interface, > I get the following error as soon as the Skype call is answered. The > handset then hangs up. It appears the problem lies in the 'audio codec > compare' phase. > > 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal > sofia/internal/IMSI[me]@[ip] [BREAK] > 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel > [skypopen/skype101/echo123] has been answered > 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 > (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE > 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] > MSG_ID=41 > 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 > [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] > outbound_channel_answered! > 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] > ACCEPTED here you send me 32776 > 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 > SO_RCVBUF is 5120, size is 4 > 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 > SO_SNDBUF is 5120, size is 4 > 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 > [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] > READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| > 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare > [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] > 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare > [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] > 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. > Disable 2833 dtmf and switch to INFO > 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal > sofia/internal/IMSI[me]@[ip] [BREAK] > 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup > sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > Any advice on forcing a transcode or otherwise having the Audio Codec > Compare result in a successful bridge is appreciated. > > Thanks, > Danny Iland > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/01949de7/attachment.html From noc at sonerep.com Thu Oct 17 12:22:33 2013 From: noc at sonerep.com (Labolinux) Date: Thu, 17 Oct 2013 08:22:33 +0000 Subject: [Freeswitch-users] [freeswitch-port] Message-ID: <525F9E49.2090708@sonerep.com> Hello, I just installed freeswitch version 1.5.6b. I want to change default internal sip port (5060). In the conf file I find vanilla, sbc, insideout, rayo, softphone. Now in what file can I change the port? Regards, D. AMOUZOU From steveayre at gmail.com Thu Oct 17 13:02:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Oct 2013 10:02:48 +0100 Subject: [Freeswitch-users] Change response code In-Reply-To: <043001cecad6$b9c59970$2d50cc50$@verizon.net> References: <043001cecad6$b9c59970$2d50cc50$@verizon.net> Message-ID: This'll hangup on a successful bridge (answer) but continue to the hangup on a failed bridge. The 34 hangup is the ISDN clearing cause - see http://wiki.freeswitch.org/wiki/Hangup_Causes (you can use enumeration names or the numbers). 34 NORMAL_CIRCUIT_CONGESTION will return 503 Service Unavailable, with the 34 cause in a Reason header. See http://tools.ietf.org/html/rfc3398 for the full list of mappings. Note it's not a 1:1 mapping so 34 maps to 503 and maps back to 41 without the Reason header to override that to provide the 34. If instead you're only wanting to control the SIP code use the respond app. The Reason header'll probably come from the SIP->ISDN mapping in the above RFC. If you want to selectively control what response to send based on the bridge's hangup cause, transfer to another context where the extensions each check the hangup cause to pick which hangup to use. ${bridge_hangup_cause} is probably the variable you'd want to check. -Steve On 17 October 2013 02:18, Andre wrote: > HI, how would I change the response code back to my customer?**** > > ** ** > > Let?s say I receive a 500 from my provider and I wanted to tell my > customer it?s a 503, how would I do that?**** > > Andre**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/7a4a7a23/attachment.html From steveayre at gmail.com Thu Oct 17 13:06:43 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Oct 2013 10:06:43 +0100 Subject: [Freeswitch-users] [freeswitch-port] In-Reply-To: <525F9E49.2090708@sonerep.com> References: <525F9E49.2090708@sonerep.com> Message-ID: vanilla, sbc, insideout, rayo, softphone are all example configurations. Pick the one closest to your needs and customise it, or create your own. The sip-port on the sip profile controls what port the sip profile binds to. The vanilla config (the one most people start from) will bind to multiple ports by using multiple sip profiles. A sip profile binds to a single ip:port. You'll find them organised within the configuration directory under the sip_profiles directory. On 17 October 2013 09:22, Labolinux wrote: > Hello, > > I just installed freeswitch version 1.5.6b. > > I want to change default internal sip port (5060). > In the conf file I find vanilla, sbc, insideout, rayo, softphone. > > Now in what file can I change the port? > > Regards, > > D. AMOUZOU > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/c29910e8/attachment.html From noc at sonerep.com Thu Oct 17 13:11:53 2013 From: noc at sonerep.com (Labolinux) Date: Thu, 17 Oct 2013 09:11:53 +0000 Subject: [Freeswitch-users] [Freeswitch-port] Message-ID: <525FA9D9.3010207@sonerep.com> Hello, I just installed freeswitch version 1.5.6. I want to change default internal sip port (5060). In the conf file I find vanilla, sbc, insideout, rayo, softphone. Now in what file can I change the port? Regards, D. AMOUZOU From ssinyagin at yahoo.com Thu Oct 17 15:33:48 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 17 Oct 2013 04:33:48 -0700 (PDT) Subject: [Freeswitch-users] [Freeswitch-port] In-Reply-To: <525FA9D9.3010207@sonerep.com> References: <525FA9D9.3010207@sonerep.com> Message-ID: <1382009628.60878.YahooMailNeo@web126205.mail.ne1.yahoo.com> I guess you won't get far without reading first. I would suggest the FreeSWITCH book, at least the first 30% of it: http://www.packtpub.com/freeswitch-1-2/book ________________________________ From: Labolinux To: freeswitch-users at lists.freeswitch.org Sent: Thursday, October 17, 2013 11:11 AM Subject: [Freeswitch-users] [Freeswitch-port] Hello, I just installed freeswitch version 1.5.6. I want to change default internal sip port (5060). In the conf file I find vanilla, sbc, insideout, rayo, softphone. Now in what file can I change the port? Regards, D. AMOUZOU _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/be40c7ab/attachment.html From alex at digitalmail.com Thu Oct 17 15:45:30 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 17 Oct 2013 12:45:30 +0100 Subject: [Freeswitch-users] Getting something to run after voicemail app Message-ID: <525FCDDA.8080602@digitalmail.com> Is there a way to get some lua script to execute after voicemail app has terminated (eg. by a hangup)? The code below won't work, as the lua action will be skipped if the caller hung up in the vm menu From steveu at coppice.org Thu Oct 17 16:00:59 2013 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Oct 2013 20:00:59 +0800 Subject: [Freeswitch-users] mod_spandsp compile error In-Reply-To: <1381965217.28081.YahooMailNeo@web121303.mail.ne1.yahoo.com> References: <1381965217.28081.YahooMailNeo@web121303.mail.ne1.yahoo.com> Message-ID: <525FD17B.3010507@coppice.org> On 10/17/2013 07:13 AM, Sadjad Seyed-Ahmadian wrote: > Hi, > > I am try to install a FreeSWITCH stable version 1.2.13 on a fresh > CentOS. I installed all prereqired packages as they are in wiki, > configure goes well but while I try to do make I got this error on > mod_spandsp. > > > making all mod_spandsp > Creating mod_spandsp_la-mod_spandsp.lo > mkdir .libs > Compiling mod_spandsp.c ... > Creating mod_spandsp_la-udptl.lo > Compiling udptl.c ... > cc1: warnings being treated as errors > udptl.c: In function 'udptl_rx_packet': > udptl.c:174: warning: 'data' may be used uninitialized in this function > udptl.c:173: warning: 'msg' may be used uninitialized in this function > make[3]: *** [mod_spandsp_la-udptl.lo] Error 1 > make[2]: *** [mod_spandsp-all] Error 1 > make[1]: *** [mod_spandsp] Error 2 > make: *** [mod_spandsp] Error 2 > > > Would somebody please help me with that? > > Best Regards > Sadjad > Centos 6.4 doesn't do that. Which version of Centos are you using? I might be missing something, but those two warnings seem bogus. Steve From steveayre at gmail.com Thu Oct 17 16:21:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Oct 2013 13:21:02 +0100 Subject: [Freeswitch-users] mod_spandsp compile error In-Reply-To: <525FD17B.3010507@coppice.org> References: <1381965217.28081.YahooMailNeo@web121303.mail.ne1.yahoo.com> <525FD17B.3010507@coppice.org> Message-ID: I'm guessing the warning is because decode_open_type gets a pointer to them before they're initialised. Not a problem if they're only for output but guessing the compiler can't tell that. On 17 October 2013 13:00, Steve Underwood wrote: > On 10/17/2013 07:13 AM, Sadjad Seyed-Ahmadian wrote: > > Hi, > > > > I am try to install a FreeSWITCH stable version 1.2.13 on a fresh > > CentOS. I installed all prereqired packages as they are in wiki, > > configure goes well but while I try to do make I got this error on > > mod_spandsp. > > > > > > making all mod_spandsp > > Creating mod_spandsp_la-mod_spandsp.lo > > mkdir .libs > > Compiling mod_spandsp.c ... > > Creating mod_spandsp_la-udptl.lo > > Compiling udptl.c ... > > cc1: warnings being treated as errors > > udptl.c: In function 'udptl_rx_packet': > > udptl.c:174: warning: 'data' may be used uninitialized in this function > > udptl.c:173: warning: 'msg' may be used uninitialized in this function > > make[3]: *** [mod_spandsp_la-udptl.lo] Error 1 > > make[2]: *** [mod_spandsp-all] Error 1 > > make[1]: *** [mod_spandsp] Error 2 > > make: *** [mod_spandsp] Error 2 > > > > > > Would somebody please help me with that? > > > > Best Regards > > Sadjad > > > Centos 6.4 doesn't do that. Which version of Centos are you using? I > might be missing something, but those two warnings seem bogus. > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a5a84ea2/attachment.html From nneul at mst.edu Thu Oct 17 16:25:27 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 17 Oct 2013 07:25:27 -0500 Subject: [Freeswitch-users] Getting something to run after voicemail app In-Reply-To: <525FCDDA.8080602@digitalmail.com> References: <525FCDDA.8080602@digitalmail.com> Message-ID: <525FD737.9020000@mst.edu> I think 'hangup_after_bridge=false' might do what you need, but have not verified that. -- Nathan On 10/17/2013 06:45 AM, Alex Lake wrote: > Is there a way to get some lua script to execute after voicemail app has > terminated (eg. by a hangup)? > > The code below won't work, as the lua action will be skipped if the > caller hung up in the vm menu > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From jleung at v10networks.ca Thu Oct 17 16:25:51 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Thu, 17 Oct 2013 20:25:51 +0800 Subject: [Freeswitch-users] [Freeswitch-port] Message-ID: <002a01cecb34$25945e2a$7c07000a@smb.curriegrad2004.ca> Every copy sold gives the project more exposure ;) Stanislav Sinyagin wrote: I guess you won't get far without reading first. I would suggest the FreeSWITCH book, at least the first 30% of it: http://www.packtpub.com/freeswitch-1-2/book ________________________________ From: Labolinux To: freeswitch-users at lists.freeswitch.org Sent: Thursday, October 17, 2013 11:11 AM Subject: [Freeswitch-users] [Freeswitch-port] Hello, I just installed freeswitch version 1.5.6. I want to change default internal sip port (5060). In the conf file I find vanilla, sbc, insideout, rayo, softphone. Now in what file can I change the port? Regards, D. AMOUZOU _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From lloyd.aloysius at gmail.com Thu Oct 17 16:32:09 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Thu, 17 Oct 2013 08:32:09 -0400 Subject: [Freeswitch-users] [Freeswitch-port] In-Reply-To: <002a01cecb34$25945e2a$7c07000a@smb.curriegrad2004.ca> References: <002a01cecb34$25945e2a$7c07000a@smb.curriegrad2004.ca> Message-ID: /usr/local/freeswitch/conf/sip_profiles/internal.xml Lloyd On Thu, Oct 17, 2013 at 8:25 AM, Jeff Leung wrote: > Every copy sold gives the project more exposure ;) > > Stanislav Sinyagin wrote: > > I guess you won't get far without reading first. > I would suggest the FreeSWITCH book, at least the first 30% of it: > http://www.packtpub.com/freeswitch-1-2/book > > > > > > > ________________________________ > From: Labolinux > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, October 17, 2013 11:11 AM > Subject: [Freeswitch-users] [Freeswitch-port] > > > Hello, > > I just installed freeswitch version 1.5.6. > > I want to change default internal sip port (5060). > In the conf file I find vanilla, sbc, insideout, rayo, softphone. > > Now in what file can I change the port? > > Regards, > > D. AMOUZOU > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/4cb543f8/attachment.html From steveu at coppice.org Thu Oct 17 16:32:48 2013 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Oct 2013 20:32:48 +0800 Subject: [Freeswitch-users] mod_spandsp compile error In-Reply-To: References: <1381965217.28081.YahooMailNeo@web121303.mail.ne1.yahoo.com> <525FD17B.3010507@coppice.org> Message-ID: <525FD8F0.4040509@coppice.org> You are probably right, but the only thing I see which might be non-compliant (although I don't think it is) is that in two places memcpy can be called with a bad pointer when the length is zero. I just changed the code so memcpy is only called when the length is non-zero. Let's see what happens with that. Regards, Steve On 10/17/2013 08:21 PM, Steven Ayre wrote: > I'm guessing the warning is because decode_open_type gets a pointer to > them before they're initialised. Not a problem if they're only for > output but guessing the compiler can't tell that. > > > On 17 October 2013 13:00, Steve Underwood > wrote: > > On 10/17/2013 07:13 AM, Sadjad Seyed-Ahmadian wrote: > > Hi, > > > > I am try to install a FreeSWITCH stable version 1.2.13 on a fresh > > CentOS. I installed all prereqired packages as they are in wiki, > > configure goes well but while I try to do make I got this error on > > mod_spandsp. > > > > > > making all mod_spandsp > > Creating mod_spandsp_la-mod_spandsp.lo > > mkdir .libs > > Compiling mod_spandsp.c ... > > Creating mod_spandsp_la-udptl.lo > > Compiling udptl.c ... > > cc1: warnings being treated as errors > > udptl.c: In function 'udptl_rx_packet': > > udptl.c:174: warning: 'data' may be used uninitialized in this > function > > udptl.c:173: warning: 'msg' may be used uninitialized in this > function > > make[3]: *** [mod_spandsp_la-udptl.lo] Error 1 > > make[2]: *** [mod_spandsp-all] Error 1 > > make[1]: *** [mod_spandsp] Error 2 > > make: *** [mod_spandsp] Error 2 > > > > > > Would somebody please help me with that? > > > > Best Regards > > Sadjad > > > Centos 6.4 doesn't do that. Which version of Centos are you using? I > might be missing something, but those two warnings seem bogus. > > Steve > From avi at avimarcus.net Thu Oct 17 17:09:38 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Oct 2013 13:09:38 +0000 Subject: [Freeswitch-users] Getting something to run after voicemail app In-Reply-To: <525FCDDA.8080602@digitalmail.com> References: <525FCDDA.8080602@digitalmail.com> Message-ID: <00000141c689253e-a951ec60-642a-456f-92fe-49abd5dd1548-000000@email.amazonses.com> I think you want a hangup hook or zombie exec turned on. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/3818faf2/attachment-0001.html From krice at freeswitch.org Thu Oct 17 17:15:07 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 17 Oct 2013 08:15:07 -0500 Subject: [Freeswitch-users] FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released! In-Reply-To: Message-ID: FreeSWITCHers! Today we are proud to announce FreeSWITCH 1.2.14! Available today via git, http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum repos! This is a maintenance release to address several bugs that have been identified since the last release. ? Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For more information on how to join see:?http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instruct ions -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/915ed3a6/attachment.html From alex at digitalmail.com Thu Oct 17 17:32:33 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 17 Oct 2013 14:32:33 +0100 Subject: [Freeswitch-users] Getting something to run after voicemail app In-Reply-To: <00000141c689253e-a951ec60-642a-456f-92fe-49abd5dd1548-000000@email.amazonses.com> References: <525FCDDA.8080602@digitalmail.com> <00000141c689253e-a951ec60-642a-456f-92fe-49abd5dd1548-000000@email.amazonses.com> Message-ID: <525FE6F1.5080601@digitalmail.com> Brilliant - love the name, and it seems to work. > I think you want a hangup hook or zombie exec turned on. > > -Avi > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/ec8c2dbd/attachment.html From anton.vazir at gmail.com Thu Oct 17 17:53:07 2013 From: anton.vazir at gmail.com (Anton VG) Date: Thu, 17 Oct 2013 17:53:07 +0400 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: References: Message-ID: Also every channel has a start time, you may kill old ones, which is there for hour or two, for instance. Also I differentiate bridged_calls and just calls - and I do not kill bridged. But for me bridged calls never hang. I did not dig deep where that dead channels are coming from, but notice they appear after a while, and just made a sweeper for now as that is not often not critical issue for me. 2013/10/11 Steven Ayre > If you use uuid_kill and the channel exists you'll see +OK, if it isn't > found you'll see an error. > > Similar errors probably occur with Apis such as uuid_setvar... Perhaps > that will give you a way to tell them apart. > > The reason for this is likely a database one. Those commands run queries > against the calls and channels tables in the core DB. FreeSWITCH updates > these when channels start, change state and end. > > It may be the transactions to remove the entries at the end if the call, > either with a logged error or silently. That'd make the db entry persist > even though the call no longer does. If that's the cause it's worth > investigating why. It could be load, db errors, or transaction locking > where another connection is already using the row. The ODBC options in > use might be worth checking. Also if FS is batching up updates to the DB > that might conceivably be affecting it. > > > > On Wednesday, October 9, 2013, Peter Hartmann wrote: > >> Hi, >> First, thanks for Freeswitch! I'm experiencing an issue where 'show >> calls' returns several calls that aren't actually happening both >> inbound and outbound. Has anyone seen this before? >> >> Rebooting the handset (Polycom IP 550) associated with that extension >> has no effect so it seems in FS. >> >> freeswitch at internal> show calls >> >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch >> d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 >> 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 >> 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 >> 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com >> ,CS_EXECUTE,unknown >> ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 >> 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 >> 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension >> 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound >> Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension >> 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, >> bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 >> 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 >> 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 >> 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 >> 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com >> ,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >> Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound >> Call,1000,,,,,,,,,,,,,,,,,,, >> >> 9 total. >> >> >> Running: >> FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 >> 2013-10-02 21:30:46Z) >> >> >> Thanks much! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/3fa8d2f1/attachment.html From alex at digitalmail.com Thu Oct 17 17:55:23 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 17 Oct 2013 14:55:23 +0100 Subject: [Freeswitch-users] Getting something to run after voicemail app In-Reply-To: <525FE6F1.5080601@digitalmail.com> References: <525FCDDA.8080602@digitalmail.com> <00000141c689253e-a951ec60-642a-456f-92fe-49abd5dd1548-000000@email.amazonses.com> <525FE6F1.5080601@digitalmail.com> Message-ID: <525FEC4B.10104@digitalmail.com> ...although it would appear that (quite reasonably, I suppose) session:execute doesn't work there. May try another method. > Brilliant - love the name, and it seems to work. >> I think you want a hangup hook or zombie exec turned on. >> >> -Avi >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/cbefb11e/attachment-0001.html From krice at freeswitch.org Thu Oct 17 18:00:18 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 17 Oct 2013 09:00:18 -0500 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: Message-ID: You should just have to update, there was a problem with this that was fixed earlier in the week... On 10/17/13 8:53 AM, "Anton VG" wrote: > Also every channel has a start time, you may kill old ones, which is there for > hour or two, for instance. Also I differentiate bridged_calls and just calls - > and I do not kill bridged. But for me bridged calls never hang. I did not dig > deep where that dead channels are coming from, but notice they appear after a > while, and just made a sweeper for now as that is not often not critical issue > for me. > > > 2013/10/11 Steven Ayre >> If you use uuid_kill and the channel exists you'll see +OK, if it isn't found >> you'll see an error. >> >> Similar errors probably occur with Apis such as uuid_setvar... Perhaps that >> will give you a way to tell them apart. >> >> The reason for this is likely a database one. Those commands run queries >> against the calls and channels tables in the core DB. FreeSWITCH updates >> these when channels start, change state and end. >> >> It may be the transactions to remove the entries at the end if the call, >> either with?a logged error or silently. That'd make the db entry persist even >> though the call no longer does. If that's the cause it's worth investigating >> why. It could be load, db errors, or transaction locking where another >> connection is already?using the row. The?ODBC options in use?might be worth >> checking. Also if FS is batching up updates to the DB that might >> conceivably?be affecting it. >> >> >> >> On Wednesday, October 9, 2013, Peter Hartmann wrote: >>> Hi, >>> First, thanks for Freeswitch! ?I'm experiencing an issue where 'show >>> calls' ?returns several calls that aren't actually happening both >>> inbound and outbound. ? ?Has anyone seen this before? >>> >>> Rebooting the handset (Polycom IP 550) associated with that extension >>> has no effect so it seems in FS. >>> >>> freeswitch at internal> show calls >>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des >>> t,presence_id,presence_data,callstate,callee_name,callee_num,callee_directio >>> n,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_c >>> reated,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest, >>> b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_calle >>> e_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch >>> d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 >>> 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+13 >>> 47xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 >>> 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+12 >>> 12xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 >>> 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,unk >>> nown >>> ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 >>> 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+12 >>> 12xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 >>> 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension >>> 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound >>> Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension >>> 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, >>> bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 >>> 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+12 >>> 12xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 >>> 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+12 >>> 12xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 >>> 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+13 >>> 47xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 >>> 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,+16 >>> 46xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound >>> Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound >>> Call,1000,,,,,,,,,,,,,,,,,,, >>> >>> 9 total. >>> >>> >>> Running: >>> FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 >>> 2013-10-02 21:30:46Z) >>> >>> >>> Thanks much! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/18c2eca5/attachment.html From jleung at v10networks.ca Thu Oct 17 19:02:31 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Thu, 17 Oct 2013 23:02:31 +0800 Subject: [Freeswitch-users] [Freeswitch-port] Message-ID: <002b01cecb49$f0a5a435$7c07000a@smb.curriegrad2004.ca> There's a 50 percent sale at packt for the 1.2 book right now if new users want it. Lloyd Aloysius wrote: /usr/local/freeswitch/conf/sip_profiles/internal.xml Lloyd On Thu, Oct 17, 2013 at 8:25 AM, Jeff Leung wrote: > Every copy sold gives the project more exposure ;) > > Stanislav Sinyagin wrote: > > I guess you won't get far without reading first. > I would suggest the FreeSWITCH book, at least the first 30% of it: > http://www.packtpub.com/freeswitch-1-2/book > > > > > > > ________________________________ > From: Labolinux > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, October 17, 2013 11:11 AM > Subject: [Freeswitch-users] [Freeswitch-port] > > > Hello, > > I just installed freeswitch version 1.5.6. > > I want to change default internal sip port (5060). > In the conf file I find vanilla, sbc, insideout, rayo, softphone. > > Now in what file can I change the port? > > Regards, > > D. AMOUZOU > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ga at steadfasttelecom.com Thu Oct 17 18:03:13 2013 From: ga at steadfasttelecom.com (Gilad Abada) Date: Thu, 17 Oct 2013 10:03:13 -0400 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: References: Message-ID: <83C3D9496C1C42EA842C7500D35A1EAE@steadfasttelecom.com> That issue has been fixed. Update to 1.2.14 -- SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office. On Thursday, October 17, 2013 at 9:53 AM, Anton VG wrote: > Also every channel has a start time, you may kill old ones, which is there for hour or two, for instance. Also I differentiate bridged_calls and just calls - and I do not kill bridged. But for me bridged calls never hang. I did not dig deep where that dead channels are coming from, but notice they appear after a while, and just made a sweeper for now as that is not often not critical issue for me. > > > 2013/10/11 Steven Ayre > > If you use uuid_kill and the channel exists you'll see +OK, if it isn't found you'll see an error. > > > > Similar errors probably occur with Apis such as uuid_setvar... Perhaps that will give you a way to tell them apart. > > > > The reason for this is likely a database one. Those commands run queries against the calls and channels tables in the core DB. FreeSWITCH updates these when channels start, change state and end. > > > > It may be the transactions to remove the entries at the end if the call, either with a logged error or silently. That'd make the db entry persist even though the call no longer does. If that's the cause it's worth investigating why. It could be load, db errors, or transaction locking where another connection is already using the row. The ODBC options in use might be worth checking. Also if FS is batching up updates to the DB that might conceivably be affecting it. > > > > > > > > On Wednesday, October 9, 2013, Peter Hartmann wrote: > > > Hi, > > > First, thanks for Freeswitch! I'm experiencing an issue where 'show > > > calls' returns several calls that aren't actually happening both > > > inbound and outbound. Has anyone seen this before? > > > > > > Rebooting the handset (Polycom IP 550) associated with that extension > > > has no effect so it seems in FS. > > > > > > freeswitch at internal> show calls > > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch > > > d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 > > > 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 > > > 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 > > > 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,unknown > > > ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 > > > 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 > > > 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension > > > 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound > > > Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension > > > 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, > > > bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 > > > 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 > > > 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 > > > 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 > > > 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound > > > Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound > > > Call,1000,,,,,,,,,,,,,,,,,,, > > > > > > 9 total. > > > > > > > > > Running: > > > FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 > > > 2013-10-02 21:30:46Z) > > > > > > > > > Thanks much! > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/798ac475/attachment-0001.html From hdiogenes at gmail.com Thu Oct 17 18:55:55 2013 From: hdiogenes at gmail.com (=?UTF-8?Q?Humberto_Di=C3=B3genes?=) Date: Thu, 17 Oct 2013 11:55:55 -0300 Subject: [Freeswitch-users] WAV + GSM recordings Message-ID: Does FreeSWITCH support WAV files with GSM 6.10 encoded audio? If not, how hard would it be to implement that? I had a look at the source and it seems that libsndfile already has support for it, I just didn't figure out which steps I'd have to go through to add this to FreeSWITCH. P.S.: Yes, I've tried recording raw .gsm files, but I don't know how I can play them without converting. -- Humberto Di?genes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/270c9f9f/attachment.html From anandaseem89 at gmail.com Thu Oct 17 19:22:24 2013 From: anandaseem89 at gmail.com (Aseem Anand) Date: Thu, 17 Oct 2013 20:52:24 +0530 Subject: [Freeswitch-users] Change code Message-ID: On Oct 17, 2013 6:51 AM, "Andre" wrote: > HI, how would I change the response code back to my customer?**** > > ** ** > > Let?s say I receive a 500 from my provider and I wanted to tell my > customer it?s a 503, how would I do that?**** > > Andre**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/c69bdc8b/attachment.html From anandaseem89 at gmail.com Thu Oct 17 19:22:24 2013 From: anandaseem89 at gmail.com (Aseem Anand) Date: Thu, 17 Oct 2013 20:52:24 +0530 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen In-Reply-To: References: Message-ID: On Oct 16, 2013 11:47 PM, "Danny Iland" wrote: > I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile > phones to VoIP providers. > > When bridging a call from a cellular handset to a mod_skypopen interface, > I get the following error as soon as the Skype call is answered. The > handset then hangs up. It appears the problem lies in the 'audio codec > compare' phase. > > 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal > sofia/internal/IMSI[me]@[ip] [BREAK] > 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel > [skypopen/skype101/echo123] has been answered > 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 > (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE > 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] > MSG_ID=41 > 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 > [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] > outbound_channel_answered! > 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] > ACCEPTED here you send me 32776 > 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 > SO_RCVBUF is 5120, size is 4 > 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 > [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 > SO_SNDBUF is 5120, size is 4 > 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 > [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] > READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| > 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare > [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] > 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare > [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] > 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. > Disable 2833 dtmf and switch to INFO > 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal > sofia/internal/IMSI[me]@[ip] [BREAK] > 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup > sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > Any advice on forcing a transcode or otherwise having the Audio Codec > Compare result in a successful bridge is appreciated. > > Thanks, > Danny Iland > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/62515f29/attachment.html From max at nysolutions.com Thu Oct 17 19:40:25 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 17 Oct 2013 15:40:25 +0000 Subject: [Freeswitch-users] Voicemail beep In-Reply-To: References: Message-ID: Sorry it actually plays a tone stream, tone_stream://L=1;%(1000, 0, 640)you should be able to change the length easily see https://wiki.freeswitch.org/wiki/Tone_stream Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Thursday, October 17, 2013 12:28 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail beep Hi Moishe, Is there a wav file that is used for voicemail beep? Where is this .wav file? what's its name? Thanks, Ali On Wed, Oct 16, 2013 at 8:51 PM, Moishe Grunstein > wrote: Just take they .wav file and modify it with a sound file editor. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Wednesday, October 16, 2013 6:59 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Voicemail beep Hello, Is there a way to change the voicemail beep sound? The existing beep is too long and too loud. Thanks, Ali _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://contactmonkey.com/api/v1/tracker?cm_session=971fa5ba-42a8-455e-889f-388c7d2d3e84&cm_type=open&cm_user_email=alipey at gmail.com] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a06a6959/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a06a6959/attachment-0001.jpg From a.afzali2003 at gmail.com Thu Oct 17 20:01:39 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Thu, 17 Oct 2013 19:31:39 +0330 Subject: [Freeswitch-users] Is FIFO in memeroy or in DB? In-Reply-To: <2013092316385046043114@gmail.com> References: <2013092316385046043114@gmail.com> Message-ID: look this thread: http://freeswitch-users.2379917.n2.nabble.com/mod-fifo-in-clustered-enviroment-td7594467.html Afshin On Mon, Sep 23, 2013 at 12:08 PM, absdou24 at gmail.com wrote: > ** > I hava two freeswitches ,these freeswitches share the same mysql, I make > extension1 which stands Member join the fifo named 111 of FS1 to waiting > for a Caller. > Then a comming caller from FS2 hits the diaplan that matches fifo 111, > I hope extension1 will answer the comming caller,However not. > > why?Is the data about FIFO in Memeroy,So FS2 can not get the extension1 > status from the memory fron FS2. > Please help me,Thanks a lot > > > ------------------------------ > Keep in mind Quality First > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/14bbd17e/attachment.html From alipey at gmail.com Thu Oct 17 20:04:24 2013 From: alipey at gmail.com (Ali Pey) Date: Thu, 17 Oct 2013 12:04:24 -0400 Subject: [Freeswitch-users] Voicemail beep In-Reply-To: References: Message-ID: Hi Moishe, Thank you for your response. Where does it play this tone stream? Where can I change it? In what directory and filename, I can change this behavior? Basically, what file do I need to edit, to change to beep sound for voicemail? Thanks, Ali On Thu, Oct 17, 2013 at 11:40 AM, Moishe Grunstein wrote: > Sorry it actually plays a tone stream, tone_stream://L=1;%(1000, 0, > 640)you should be able to change the length easily see > https://wiki.freeswitch.org/wiki/Tone_stream **** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Thursday, October 17, 2013 12:28 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Voicemail beep**** > > ** ** > > Hi Moishe,**** > > ** ** > > Is there a wav file that is used for voicemail beep?**** > > Where is this .wav file? what's its name?**** > > ** ** > > Thanks, > Ali**** > > ** ** > > On Wed, Oct 16, 2013 at 8:51 PM, Moishe Grunstein > wrote:**** > > Just take they .wav file and modify it with a sound file editor.**** > > **** > > **** > > Thanks,**** > > **** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Wednesday, October 16, 2013 6:59 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Voicemail beep**** > > **** > > Hello,**** > > **** > > Is there a way to change the voicemail beep sound?**** > > **** > > The existing beep is too long and too loud.**** > > **** > > Thanks,**** > > Ali**** > > **** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/7e5365df/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/7e5365df/attachment.jpe From max at nysolutions.com Thu Oct 17 20:12:40 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 17 Oct 2013 16:12:40 +0000 Subject: [Freeswitch-users] Voicemail beep In-Reply-To: References: Message-ID: Voicemail.conf.xml. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Thursday, October 17, 2013 12:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail beep Hi Moishe, Thank you for your response. Where does it play this tone stream? Where can I change it? In what directory and filename, I can change this behavior? Basically, what file do I need to edit, to change to beep sound for voicemail? Thanks, Ali On Thu, Oct 17, 2013 at 11:40 AM, Moishe Grunstein > wrote: Sorry it actually plays a tone stream, tone_stream://L=1;%(1000, 0, 640)you should be able to change the length easily see https://wiki.freeswitch.org/wiki/Tone_stream Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Thursday, October 17, 2013 12:28 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Voicemail beep Hi Moishe, Is there a wav file that is used for voicemail beep? Where is this .wav file? what's its name? Thanks, Ali On Wed, Oct 16, 2013 at 8:51 PM, Moishe Grunstein > wrote: Just take they .wav file and modify it with a sound file editor. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Wednesday, October 16, 2013 6:59 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Voicemail beep Hello, Is there a way to change the voicemail beep sound? The existing beep is too long and too loud. Thanks, Ali _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://contactmonkey.com/api/v1/tracker?cm_session=84754d34-832a-48f4-bac1-1a87954fbc8f&cm_type=open&cm_user_email=alipey at gmail.com] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a27b211f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a27b211f/attachment-0001.jpg From mario_fs at mgtech.com Thu Oct 17 20:19:38 2013 From: mario_fs at mgtech.com (Mario G) Date: Thu, 17 Oct 2013 09:19:38 -0700 Subject: [Freeswitch-users] Can a Jira summary/title be changed? Message-ID: <937D357C-4387-4ECF-A9A9-79A91EBA13E5@mgtech.com> Anyone know if changing the summary title of your own Jira issue is possible? Issue titles may need to be changed because the real/root problem was not known at the time it was created and a correct summary may be more useful for others with the same problem. For instance, I have: FS-5375 FreeSwitch crash during paging/conference <- current FreeSwitch crash using G722 on OSX <- should be FS-5223 improve message: Error Creating SIP UA for profile xxx <- old internal profile TCP port stays in use after FS shutdown on OSX <- should be Thanks, Mario G From alipey at gmail.com Thu Oct 17 20:25:14 2013 From: alipey at gmail.com (Ali Pey) Date: Thu, 17 Oct 2013 12:25:14 -0400 Subject: [Freeswitch-users] Voicemail beep In-Reply-To: References: Message-ID: Great. Thanks. I see it now. It is the "tone-spec" parameter in voicemail.conf.xml Regards, Ali On Thu, Oct 17, 2013 at 12:12 PM, Moishe Grunstein wrote: > Voicemail.conf.xml.**** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Thursday, October 17, 2013 12:04 PM > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] Voicemail beep**** > > ** ** > > Hi Moishe,**** > > ** ** > > Thank you for your response.**** > > ** ** > > Where does it play this tone stream? Where can I change it? **** > > In what directory and filename, I can change this behavior?**** > > ** ** > > Basically, what file do I need to edit, to change to beep sound for > voicemail?**** > > ** ** > > Thanks,**** > > Ali**** > > ** ** > > On Thu, Oct 17, 2013 at 11:40 AM, Moishe Grunstein > wrote:**** > > Sorry it actually plays a tone stream, tone_stream://L=1;%(1000, 0, > 640)you should be able to change the length easily see > https://wiki.freeswitch.org/wiki/Tone_stream **** > > **** > > **** > > Thanks,**** > > **** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Thursday, October 17, 2013 12:28 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Voicemail beep**** > > **** > > Hi Moishe,**** > > **** > > Is there a wav file that is used for voicemail beep?**** > > Where is this .wav file? what's its name?**** > > **** > > Thanks, > Ali**** > > **** > > On Wed, Oct 16, 2013 at 8:51 PM, Moishe Grunstein > wrote:**** > > Just take they .wav file and modify it with a sound file editor.**** > > **** > > **** > > Thanks,**** > > **** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Wednesday, October 16, 2013 6:59 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Voicemail beep**** > > **** > > Hello,**** > > **** > > Is there a way to change the voicemail beep sound?**** > > **** > > The existing beep is too long and too loud.**** > > **** > > Thanks,**** > > Ali**** > > **** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/284c622a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/284c622a/attachment-0001.jpe From ssa1357 at yahoo.com Thu Oct 17 22:18:06 2013 From: ssa1357 at yahoo.com (Sadjad Seyed-Ahmadian) Date: Thu, 17 Oct 2013 11:18:06 -0700 (PDT) Subject: [Freeswitch-users] mod_spandsp compile error Message-ID: <1382033886.80911.YahooMailNeo@web121306.mail.ne1.yahoo.com> it is CentOS release 5.9 (Final) On 10/17/2013 07:13 AM, Sadjad Seyed-Ahmadian wrote: >Hi, >>I am try to install a FreeSWITCH stable version 1.2.13 on a fresh >CentOS. I installed all prereqired packages as they are in wiki, >configure goes well but while I try to do make I got this error on >mod_spandsp. >>>making all mod_spandsp >Creating mod_spandsp_la-mod_spandsp.lo >mkdir .libs >Compiling mod_spandsp.c ... >Creating mod_spandsp_la-udptl.lo >Compiling udptl.c ... >cc1: warnings being treated as errors >udptl.c: In function 'udptl_rx_packet': >udptl.c:174: warning: 'data' may be used uninitialized in this function >udptl.c:173: warning: 'msg' may be used uninitialized in this function >make[3]: *** [mod_spandsp_la-udptl.lo] Error 1 >make[2]: *** [mod_spandsp-all] Error 1 >make[1]: *** [mod_spandsp] Error 2 >make: *** [mod_spandsp] Error 2 >>>Would somebody please help me with that? >>Best Regards >Sadjad >Centos 6.4 doesn't do that. Which version of Centos are you using? I might be missing something, but those two warnings seem bogus. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/c466b54e/attachment.html From victor.chukalovskiy at gmail.com Thu Oct 17 22:36:55 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 17 Oct 2013 14:36:55 -0400 Subject: [Freeswitch-users] How to set channel variable per SIP profile Message-ID: <52602E47.5050803@gmail.com> Hi, Subject says it all. Is there a way to set channel variable for all calls passing through a SIP profile? For example, I want all calls authorized with ACL and hitting SIP profile to have country=foo. This has to be done before hitting the dialplan. There is a bunch of "params" in the SIP profile, but these aren't channel variables. Thanks! -Victor From richard.mace at gmail.com Thu Oct 17 22:56:19 2013 From: richard.mace at gmail.com (Richard Mace) Date: Thu, 17 Oct 2013 19:56:19 +0100 Subject: [Freeswitch-users] FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released! In-Reply-To: References: Message-ID: Great news. Do the Debian packages have h323? On 17 Oct 2013 14:18, "Ken Rice" wrote: > > FreeSWITCHers! > > Today we are proud to announce FreeSWITCH 1.2.14! > > Available today via git, > http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and > yum repos! > > This is a maintenance release to address several bugs that have been > identified since the last release. > > Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For > more information on how to join see: > http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/7107897b/attachment.html From andretodd at verizon.net Thu Oct 17 23:15:41 2013 From: andretodd at verizon.net (Andre) Date: Thu, 17 Oct 2013 15:15:41 -0400 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows Message-ID: <085301cecb6d$464824d0$d2d86e70$@verizon.net> Hi, when I compile 64 bit I get this error Error 1 error C1083: Cannot open include file: 'nametab.h': No such file or directory \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c 10 1 xmltok Any idea what this means or how to fix it? Or does it even matter? Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/8765bb05/attachment.html From andretodd at verizon.net Thu Oct 17 23:36:26 2013 From: andretodd at verizon.net (Andre) Date: Thu, 17 Oct 2013 15:36:26 -0400 Subject: [Freeswitch-users] park_after_bridge Message-ID: <087901cecb70$2c41d290$84c577b0$@verizon.net> Why does park after bridge only work in media bypass mode? s.SetVariable("park_after_bridge", "true"); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/8646ca23/attachment.html From bedgar at vseinc.com Thu Oct 17 23:49:53 2013 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Thu, 17 Oct 2013 15:49:53 -0400 Subject: [Freeswitch-users] old calls hanging around In-Reply-To: <83C3D9496C1C42EA842C7500D35A1EAE@steadfasttelecom.com> References: <83C3D9496C1C42EA842C7500D35A1EAE@steadfasttelecom.com> Message-ID: <31A79B0B4414EB4B83C7EAF307ED57DA15D34B2A09@prod-exch01.corp.vseinc.com> Would this issue contribute to freetdm channels getting stuck in a gateway on calls bridged to back-end nodes? Brian From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gilad Abada Sent: Thursday, October 17, 2013 10:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] old calls hanging around That issue has been fixed. Update to 1.2.14 -- [http://steadfasttelecom.com/images/logo_steadfast.gif] SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office. On Thursday, October 17, 2013 at 9:53 AM, Anton VG wrote: Also every channel has a start time, you may kill old ones, which is there for hour or two, for instance. Also I differentiate bridged_calls and just calls - and I do not kill bridged. But for me bridged calls never hang. I did not dig deep where that dead channels are coming from, but notice they appear after a while, and just made a sweeper for now as that is not often not critical issue for me. 2013/10/11 Steven Ayre > If you use uuid_kill and the channel exists you'll see +OK, if it isn't found you'll see an error. Similar errors probably occur with Apis such as uuid_setvar... Perhaps that will give you a way to tell them apart. The reason for this is likely a database one. Those commands run queries against the calls and channels tables in the core DB. FreeSWITCH updates these when channels start, change state and end. It may be the transactions to remove the entries at the end if the call, either with a logged error or silently. That'd make the db entry persist even though the call no longer does. If that's the cause it's worth investigating why. It could be load, db errors, or transaction locking where another connection is already using the row. The ODBC options in use might be worth checking. Also if FS is batching up updates to the DB that might conceivably be affecting it. On Wednesday, October 9, 2013, Peter Hartmann wrote: Hi, First, thanks for Freeswitch! I'm experiencing an issue where 'show calls' returns several calls that aren't actually happening both inbound and outbound. Has anyone seen this before? Rebooting the handset (Polycom IP 550) associated with that extension has no effect so it seems in FS. freeswitch at internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 15:55:49,1381002949,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 09:59:34,1381154374,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 10:44:28,1381157068,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,unknown ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 10:49:05,1381157345,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 12:06:38,1381161998,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 13:37:17,1381167437,sofia/external/+1212xxxxxxx at flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 15:09:34,1381172974,sofia/external/+1347xxxxxxx at flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 15:41:20,1381174880,sofia/external/+1646xxxxxxx at flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 9 total. Running: FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 2013-10-02 21:30:46Z) Thanks much! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/d328e3f6/attachment-0001.html From victor.chukalovskiy at gmail.com Thu Oct 17 23:54:24 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 17 Oct 2013 15:54:24 -0400 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: References: <525378A5.40706@gmail.com> <52542FED.1050506@gmail.com> Message-ID: <52604070.5040505@gmail.com> Thanks to everyone who contributed! I'm attaching summary of the results On 13-10-08 09:59 PM, Henry Huang wrote: > Will you share the results? > > > > On Tue, Oct 8, 2013 at 2:27 PM, Nikolay Rogoshchenkov > > wrote: > > In second question is impossible to chose "other". > > > -- > Rogoshchenkov Nikolay > > > On Tue, Oct 8, 2013 at 12:16 PM, Victor Chukalovskiy > > wrote: > > Bump! Not sure if my message reached the list yesterday (given > all the > "tests" today) :-) > > http://www.surveymonkey.com/s/J8SL9TS > > On 13-10-07 11 :14 PM, Victor Chukalovskiy > wrote: > > Greetings! > > > > Everyone who runs some sort of billing for freeswitch, > please provide > > your feedback here: > > > > http://www.surveymonkey.com/s/J8SL9TS > > > > Just 4 question. Less then a minute to complete! > > > > Survey runs for 2-3 days. I'll post final results in this > thread for > > your benefit. > > > > Cheers, > > -Victor > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/ae2ea312/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: FreeSWITCH billing solutions.pdf Type: application/pdf Size: 96957 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/ae2ea312/attachment-0001.pdf From shishko69 at gmail.com Fri Oct 18 00:08:05 2013 From: shishko69 at gmail.com (Denis Papes) Date: Thu, 17 Oct 2013 22:08:05 +0200 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: <085301cecb6d$464824d0$d2d86e70$@verizon.net> References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> Message-ID: I successfully compiled 64-bit version using Visual Studio 2010. File nametab.h should be in same directory as xmltok.c, \libs\xmlrpc-c\lib\expat\xmltok. On Thu, Oct 17, 2013 at 9:15 PM, Andre wrote: > Hi, when I compile 64 bit I get this error**** > > ** ** > > Error 1 error C1083: Cannot open include file: 'nametab.h': No such file > or directory > \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c > 10 1 xmltok**** > > ** ** > > Any idea what this means or how to fix it? Or does it even matter?**** > > Andre**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/3af115c8/attachment.html From steveayre at gmail.com Fri Oct 18 00:15:53 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Oct 2013 21:15:53 +0100 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen In-Reply-To: References: Message-ID: Solved on IRC - mod_spandsp was failing to load because of a build error where some libraries were missing. GSM wasn't available as a result, so FS couldn't initiate media once the call rang / was answered. http://xkcd.com/979/ On 17 October 2013 16:22, Aseem Anand wrote: > On Oct 16, 2013 11:47 PM, "Danny Iland" wrote: > >> I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile >> phones to VoIP providers. >> >> When bridging a call from a cellular handset to a mod_skypopen interface, >> I get the following error as soon as the Skype call is answered. The >> handset then hangs up. It appears the problem lies in the 'audio codec >> compare' phase. >> >> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal >> sofia/internal/IMSI[me]@[ip] [BREAK] >> 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel >> [skypopen/skype101/echo123] has been answered >> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 >> (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE >> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] >> MSG_ID=41 >> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] >> outbound_channel_answered! >> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] >> ACCEPTED here you send me 32776 >> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 >> SO_RCVBUF is 5120, size is 4 >> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 >> SO_SNDBUF is 5120, size is 4 >> 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] >> READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| >> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare >> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] >> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare >> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] >> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. >> Disable 2833 dtmf and switch to INFO >> 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal >> sofia/internal/IMSI[me]@[ip] [BREAK] >> 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup >> sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >> >> Any advice on forcing a transcode or otherwise having the Audio Codec >> Compare result in a successful bridge is appreciated. >> >> Thanks, >> Danny Iland >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/e965ea36/attachment.html From andretodd at verizon.net Fri Oct 18 00:17:37 2013 From: andretodd at verizon.net (Andre) Date: Thu, 17 Oct 2013 16:17:37 -0400 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> Message-ID: <08b801cecb75$ed277000$c7765000$@verizon.net> Mine is not there. Should I just copy it there and recompile? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Papes Sent: Thursday, October 17, 2013 4:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows I successfully compiled 64-bit version using Visual Studio 2010. File nametab.h should be in same directory as xmltok.c, \libs\xmlrpc-c\lib\expat\xmltok. On Thu, Oct 17, 2013 at 9:15 PM, Andre > wrote: Hi, when I compile 64 bit I get this error Error 1 error C1083: Cannot open include file: 'nametab.h': No such file or directory \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c 10 1 xmltok Any idea what this means or how to fix it? Or does it even matter? Andre _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/dcb3b0e8/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Oct 18 00:47:58 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 17 Oct 2013 21:47:58 +0100 Subject: [Freeswitch-users] Survey - FreeSWITCH billing solutions In-Reply-To: <52604070.5040505@gmail.com> References: <525378A5.40706@gmail.com> <52542FED.1050506@gmail.com> <52604070.5040505@gmail.com> Message-ID: That's really interesting that the majority of people use in-house. Cal On Thu, Oct 17, 2013 at 8:54 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Thanks to everyone who contributed! I'm attaching summary of the results > > > > On 13-10-08 09:59 PM, Henry Huang wrote: > > Will you share the results? > > > > On Tue, Oct 8, 2013 at 2:27 PM, Nikolay Rogoshchenkov > wrote: > >> In second question is impossible to chose "other". >> >> >> -- >> Rogoshchenkov Nikolay >> >> >> On Tue, Oct 8, 2013 at 12:16 PM, Victor Chukalovskiy < >> victor.chukalovskiy at gmail.com> wrote: >> >>> Bump! Not sure if my message reached the list yesterday (given all the >>> "tests" today) :-) >>> >>> http://www.surveymonkey.com/s/J8SL9TS >>> >>> On 13-10-07 11:14 PM, Victor Chukalovskiy wrote: >>> > Greetings! >>> > >>> > Everyone who runs some sort of billing for freeswitch, please provide >>> > your feedback here: >>> > >>> > http://www.surveymonkey.com/s/J8SL9TS >>> > >>> > Just 4 question. Less then a minute to complete! >>> > >>> > Survey runs for 2-3 days. I'll post final results in this thread for >>> > your benefit. >>> > >>> > Cheers, >>> > -Victor >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/cabdc67b/attachment.html From ssinyagin at yahoo.com Fri Oct 18 00:50:13 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 17 Oct 2013 13:50:13 -0700 (PDT) Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <52602E47.5050803@gmail.com> References: <52602E47.5050803@gmail.com> Message-ID: <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> you can set variables at the gateway level, and then they? are set as channel variables for inbound calls: http://wiki.freeswitch.org/wiki/Sofia.conf.xml ________________________________ From: Victor Chukalovskiy To: FreeSWITCH Users Help Sent: Thursday, October 17, 2013 8:36 PM Subject: [Freeswitch-users] How to set channel variable per SIP profile Hi, Subject says it all. Is there a way to set channel variable for all calls passing through a SIP profile? For example, I want all calls authorized with ACL and hitting SIP profile to have country=foo. This has to be done before hitting the dialplan. There is a bunch of "params" in the SIP profile, but these aren't channel variables. Thanks! -Victor _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/d29e36a3/attachment.html From victor.chukalovskiy at gmail.com Fri Oct 18 00:59:12 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 17 Oct 2013 16:59:12 -0400 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> Message-ID: <52604FA0.1020202@gmail.com> Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. Is it possible? On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: > you can set variables at the gateway level, and then they are set as > channel variables for inbound calls: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > *From:* Victor Chukalovskiy > *To:* FreeSWITCH Users Help > *Sent:* Thursday, October 17, 2013 8:36 PM > *Subject:* [Freeswitch-users] How to set channel variable per SIP profile > > Hi, > > Subject says it all. Is there a way to set channel variable for all > calls passing through a SIP profile? > > For example, I want all calls authorized with ACL and hitting SIP > profile to have country=foo. This has to be done before hitting the > dialplan. > There is a bunch of "params" in the SIP profile, but these aren't > channel variables. > > Thanks! > -Victor > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/9fec627a/attachment-0001.html From ssinyagin at yahoo.com Fri Oct 18 01:59:36 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 17 Oct 2013 14:59:36 -0700 (PDT) Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <52604FA0.1020202@gmail.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> Message-ID: <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> if you run info() on the incoming call, you can see the profile's IP address and port and profile name in various variables. Then you can build the matching rules in your dialplan and differentiate between the profiles, and set variables that you need. But probably simply assigning different contexts to your profiles would do the job :) ________________________________ From: Victor Chukalovskiy To: freeswitch-users at lists.freeswitch.org Sent: Thursday, October 17, 2013 10:59 PM Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. Is it possible? On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: you can set variables at the gateway level, and then they? are set as channel variables for inbound calls: > >http://wiki.freeswitch.org/wiki/Sofia.conf.xml > > > > > > > > >________________________________ > From: Victor Chukalovskiy >To: FreeSWITCH Users Help >Sent: Thursday, October 17, 2013 8:36 PM >Subject: [Freeswitch-users] How to set channel variable per SIP profile > > >Hi, > >Subject says it all. Is there a way to set channel variable for all >calls passing through a SIP profile? > >For example, I want all calls authorized with ACL and hitting SIP >profile to have country=foo. This has to be done before hitting the >dialplan. >There is a bunch of "params" in the SIP profile, but these aren't >channel variables. > >Thanks! >-Victor > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > > > >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/a21e9c3b/attachment.html From iland at cs.ucsb.edu Thu Oct 17 22:36:34 2013 From: iland at cs.ucsb.edu (Danny Iland) Date: Thu, 17 Oct 2013 11:36:34 -0700 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen In-Reply-To: References: Message-ID: Giovanni, I know the Skype API takes a TCP stream of SL 16. In order to force the correct codecs, I inserted setting the following in the dialplan just before the bridge. However call establishment still fails with INCOMPATIBLE_DESTINATION after comparing GSM and L16: 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[L16:70:8000:20:128000] Any suggestions on the contents of global_codec_prefs, outbound_codec_prefs, or nolocal:absolute_codec_string? I suspect I need to define something like L16 at 13200h but when I tried this, the L16 offering did not change. Full log snippet follows Thanks, Danny Iland 2013-10-17 18:22:25.252453 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype102/echo123] has been answered 2013-10-17 18:22:25.252453 [DEBUG] switch_channel.c:3594 (skypopen/skype102/echo123) Callstate Change RINGING -> ACTIVE 2013-10-17 18:22:25.252453 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype102 ][UP,INPROGRS] MSG_ID=41 2013-10-17 18:22:25.252453 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype102 ][UP,INPROGRS] outbound_channel_answered! 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[L16:70:8000:20:128000] 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-17 18:22:25.272510 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/[me]@[ip] [BREAK] 2013-10-17 18:22:25.272510 [NOTICE] switch_channel.c:3633 Hangup sofia/internal//[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-10-17 18:22:25.272510 [DEBUG] switch_channel.c:3134 Send signal sofia/internal//[me]@[ip] [KILL] 2013-10-17 18:22:25.272510 [DEBUG] switch_core_session.c:1338 Send signal sofia/internal//[me]@[ip] [BREAK] 2013-10-17 18:22:25.272510 [DEBUG] switch_ivr_originate.c:3450 sofia/internal//[me]@[ip] Media Establishment Failed. 2013-10-17 18:22:25.272510 [NOTICE] switch_ivr_originate.c:3452 Hangup skypopen/skype102/echo123 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] On Thu, Oct 17, 2013 at 1:00 AM, Giovanni Maruzzelli wrote: > mod_skypopen uses signed raw linear 16 bit (eg: SL16) natively, but you > seem to force it to use PCMU or PCMA (eg: g711). > > FreeSWITCH will automatically do the conversion for you (from whatever > your channel uses) when bridging to skypopen if you don't force it. > > -giovanni > > > On Wed, Oct 16, 2013 at 8:04 PM, Danny Iland wrote: > >> I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile >> phones to VoIP providers. >> >> When bridging a call from a cellular handset to a mod_skypopen interface, >> I get the following error as soon as the Skype call is answered. The >> handset then hangs up. It appears the problem lies in the 'audio codec >> compare' phase. >> >> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal >> sofia/internal/IMSI[me]@[ip] [BREAK] >> 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel >> [skypopen/skype101/echo123] has been answered >> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 >> (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE >> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] >> MSG_ID=41 >> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] >> outbound_channel_answered! >> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] >> ACCEPTED here you send me 32776 >> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 >> SO_RCVBUF is 5120, size is 4 >> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 >> SO_SNDBUF is 5120, size is 4 >> 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 >> [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] >> READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| >> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare >> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] >> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare >> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] >> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. >> Disable 2833 dtmf and switch to INFO >> 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal >> sofia/internal/IMSI[me]@[ip] [BREAK] >> 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup >> sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >> >> Any advice on forcing a transcode or otherwise having the Audio Codec >> Compare result in a successful bridge is appreciated. >> >> Thanks, >> Danny Iland >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/221bbd53/attachment-0001.html From victor.chukalovskiy at gmail.com Fri Oct 18 06:31:16 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 17 Oct 2013 22:31:16 -0400 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> Message-ID: <52609D74.20906@gmail.com> Hi, yes, I already use all of these. The profile_name variable and putting different profiles into different contexts. But what I really need is to be able to set a couple custom variables per profile. This would allow to use the exact same context with different profiles....as well as couple other things. I guess it's not possible!!!! :-( On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: > if you run info() on the incoming call, you can see the profile's IP > address and port and profile name in various variables. > Then you can build the matching rules in your dialplan and > differentiate between the profiles, and set variables that you need. > > But probably simply assigning different contexts to your profiles > would do the job :) > > > > > > ------------------------------------------------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, October 17, 2013 10:59 PM > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. > > Is it possible? > On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: >> you can set variables at the gateway level, and then they are set as >> channel variables for inbound calls: >> >> http://wiki.freeswitch.org/wiki/Sofia.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ------------------------------------------------------------------------ >> *From:* Victor Chukalovskiy >> >> *To:* FreeSWITCH Users Help >> >> *Sent:* Thursday, October 17, 2013 8:36 PM >> *Subject:* [Freeswitch-users] How to set channel variable per SIP profile >> >> Hi, >> >> Subject says it all. Is there a way to set channel variable for all >> calls passing through a SIP profile? >> >> For example, I want all calls authorized with ACL and hitting SIP >> profile to have country=foo. This has to be done before hitting the >> dialplan. >> There is a bunch of "params" in the SIP profile, but these aren't >> channel variables. >> >> Thanks! >> -Victor >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131017/6e5b96ab/attachment.html From shishko69 at gmail.com Fri Oct 18 08:37:54 2013 From: shishko69 at gmail.com (Denis Papes) Date: Fri, 18 Oct 2013 06:37:54 +0200 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: <08b801cecb75$ed277000$c7765000$@verizon.net> References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> <08b801cecb75$ed277000$c7765000$@verizon.net> Message-ID: Download it again http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2 and retry. On Thu, Oct 17, 2013 at 10:17 PM, Andre wrote: > Mine is not there. Should I just copy it there and recompile?**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Denis Papes > *Sent:* Thursday, October 17, 2013 4:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows**** > > ** ** > > I successfully compiled 64-bit version using Visual Studio 2010. File > nametab.h should be in same directory as xmltok.c, > \libs\xmlrpc-c\lib\expat\xmltok.**** > > ** ** > > On Thu, Oct 17, 2013 at 9:15 PM, Andre wrote:**** > > Hi, when I compile 64 bit I get this error**** > > **** > > Error 1 error C1083: Cannot open include file: 'nametab.h': No such file > or directory > \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c > 10 1 xmltok**** > > **** > > Any idea what this means or how to fix it? Or does it even matter?**** > > Andre**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/8c0a0a4d/attachment-0001.html From jleung at v10networks.ca Fri Oct 18 10:49:28 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 18 Oct 2013 14:49:28 +0800 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows Message-ID: <002c01cecbe1$c3a6c77d$7c07000a@smb.curriegrad2004.ca> Did openssl download correctly? if not, nuke the tree ans start again. Denis Papes wrote: Download it again http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2 and retry. On Thu, Oct 17, 2013 at 10:17 PM, Andre wrote: > Mine is not there. Should I just copy it there and recompile?**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Denis Papes > *Sent:* Thursday, October 17, 2013 4:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows**** > > ** ** > > I successfully compiled 64-bit version using Visual Studio 2010. File > nametab.h should be in same directory as xmltok.c, > \libs\xmlrpc-c\lib\expat\xmltok.**** > > ** ** > > On Thu, Oct 17, 2013 at 9:15 PM, Andre wrote:**** > > Hi, when I compile 64 bit I get this error**** > > **** > > Error 1 error C1083: Cannot open include file: 'nametab.h': No such file > or directory > \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c > 10 1 xmltok**** > > **** > > Any idea what this means or how to fix it? Or does it even matter?**** > > Andre**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gmaruzz at gmail.com Fri Oct 18 14:20:41 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 18 Oct 2013 12:20:41 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen In-Reply-To: References: Message-ID: :) On Thu, Oct 17, 2013 at 10:15 PM, Steven Ayre wrote: > Solved on IRC - mod_spandsp was failing to load because of a build error > where some libraries were missing. GSM wasn't available as a result, so FS > couldn't initiate media once the call rang / was answered. > > http://xkcd.com/979/ > > > > > On 17 October 2013 16:22, Aseem Anand wrote: > >> On Oct 16, 2013 11:47 PM, "Danny Iland" wrote: >> >>> I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile >>> phones to VoIP providers. >>> >>> When bridging a call from a cellular handset to a mod_skypopen >>> interface, I get the following error as soon as the Skype call is answered. >>> The handset then hangs up. It appears the problem lies in the 'audio codec >>> compare' phase. >>> >>> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal >>> sofia/internal/IMSI[me]@[ip] [BREAK] >>> 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel >>> [skypopen/skype101/echo123] has been answered >>> 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 >>> (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE >>> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 >>> [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] >>> MSG_ID=41 >>> 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 >>> [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] >>> outbound_channel_answered! >>> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 >>> [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] >>> ACCEPTED here you send me 32776 >>> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 >>> [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 >>> SO_RCVBUF is 5120, size is 4 >>> 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 >>> [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 >>> SO_SNDBUF is 5120, size is 4 >>> 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 >>> [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] >>> READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| >>> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare >>> [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] >>> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare >>> [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] >>> 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. >>> Disable 2833 dtmf and switch to INFO >>> 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal >>> sofia/internal/IMSI[me]@[ip] [BREAK] >>> 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup >>> sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>> >>> Any advice on forcing a transcode or otherwise having the Audio Codec >>> Compare result in a successful bridge is appreciated. >>> >>> Thanks, >>> Danny Iland >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/a22fd968/attachment.html From juanito1982 at gmail.com Fri Oct 18 15:24:00 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 18 Oct 2013 13:24:00 +0200 Subject: [Freeswitch-users] Bad request answer Message-ID: Hello, I am getting "Bad request" answer from FS but I don't know why. I have phone as that (AT810) in other locations without problems. You can see one INVITE and answer here http://pastebin.com/Widd5j2i Do you know why FS doesn't like it? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/389610ed/attachment.html From anand at voxvalley.com Fri Oct 18 10:08:23 2013 From: anand at voxvalley.com (anand) Date: Fri, 18 Oct 2013 11:38:23 +0530 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <52609D74.20906@gmail.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> Message-ID: <5260D057.3060906@voxvalley.com> You can set dynamic data in dial plan as below Anand On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: > Hi, yes, I already use all of these. The profile_name variable and > putting different profiles into different contexts. > > But what I really need is to be able to set a couple custom variables > per profile. This would allow to use the exact same context with > different profiles....as well as couple other things. > > I guess it's not possible!!!! :-( > > On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: >> if you run info() on the incoming call, you can see the profile's IP >> address and port and profile name in various variables. >> Then you can build the matching rules in your dialplan and >> differentiate between the profiles, and set variables that you need. >> >> But probably simply assigning different contexts to your profiles >> would do the job :) >> >> >> >> >> >> ------------------------------------------------------------------------ >> *From:* Victor Chukalovskiy >> *To:* freeswitch-users at lists.freeswitch.org >> *Sent:* Thursday, October 17, 2013 10:59 PM >> *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP >> profile >> >> Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. >> >> Is it possible? >> On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: >>> you can set variables at the gateway level, and then they are set >>> as channel variables for inbound calls: >>> >>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------ >>> *From:* Victor Chukalovskiy >>> >>> *To:* FreeSWITCH Users Help >>> >>> *Sent:* Thursday, October 17, 2013 8:36 PM >>> *Subject:* [Freeswitch-users] How to set channel variable per SIP >>> profile >>> >>> Hi, >>> >>> Subject says it all. Is there a way to set channel variable for all >>> calls passing through a SIP profile? >>> >>> For example, I want all calls authorized with ACL and hitting SIP >>> profile to have country=foo. This has to be done before hitting the >>> dialplan. >>> There is a bunch of "params" in the SIP profile, but these aren't >>> channel variables. >>> >>> Thanks! >>> -Victor >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/09c3c59e/attachment-0001.html From mcazzador at gmail.com Fri Oct 18 15:56:20 2013 From: mcazzador at gmail.com (Matteo Cazzador) Date: Fri, 18 Oct 2013 13:56:20 +0200 Subject: [Freeswitch-users] limit incoming concurrent fax/calls Message-ID: Hi, i've configured a fax server (Centos 6.2) using ICT Fax with Plivo Framework.I'm a novice about freeswitch. All goes well, but i've a request. I need to limit concurrent incoming calls (fax receive) max 50 contemporary incoming fax. I can use "limit" parameter (from freeswitch doumentation)? Is it correct to set it in /usr/local/freeswiitch/conf/dialplan/default.xml or public.xml? I want that limit globally, max 50 concurrent calls incoming non for single DID. Thank's a lot -- Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail. ****************************************** Ing. Matteo Cazzador Email: mcazzador at gmail.com ****************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/080b886d/attachment.html From anthony.minessale at gmail.com Fri Oct 18 17:11:59 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Oct 2013 08:11:59 -0500 Subject: [Freeswitch-users] Bad request answer In-Reply-To: References: Message-ID: Maybe try factory resetting the device? On Fri, Oct 18, 2013 at 6:24 AM, Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com> wrote: > Hello, > > I am getting "Bad request" answer from FS but I don't know why. I have > phone as that (AT810) in other locations without problems. You can see one > INVITE and answer here > > http://pastebin.com/Widd5j2i > > Do you know why FS doesn't like it? > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/1b75b2c8/attachment.html From miha at softnet.si Fri Oct 18 17:22:32 2013 From: miha at softnet.si (Miha) Date: Fri, 18 Oct 2013 15:22:32 +0200 Subject: [Freeswitch-users] limit incoming concurrent fax/calls In-Reply-To: References: Message-ID: <52613618.3050705@softnet.si> Hi, it depends if you would like to limit inbound call from "outside" that you will do this in public, if you would like to limit "local" calls that you will do in in default dialplan. If you would like to limit bouth, than public and default:) miha Dne 10/18/2013 1:56 PM, pis(e Matteo Cazzador: > Hi, i've configured a fax server (Centos 6.2) using ICT Fax with Plivo > Framework.I'm a novice about freeswitch. > All goes well, but i've a request. I need to limit concurrent incoming > calls (fax receive) max 50 contemporary incoming fax. > > I can use "limit" parameter (from freeswitch doumentation)? > > Is it correct to set it in > /usr/local/freeswiitch/conf/dialplan/default.xml or public.xml? > > I want that limit globally, max 50 concurrent calls incoming non for > single DID. > > Thank's a lot > > -- > Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail. > ****************************************** > Ing. Matteo Cazzador > Email: mcazzador at gmail.com > ****************************************** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/5aaf06fe/attachment.html From andretodd at verizon.net Fri Oct 18 17:26:51 2013 From: andretodd at verizon.net (Andre) Date: Fri, 18 Oct 2013 09:26:51 -0400 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> <08b801cecb75$ed277000$c7765000$@verizon.net> Message-ID: <007901cecc05$b54df150$1fe9d3f0$@verizon.net> Thanks, I downloaded it from the that url when It was sent to freeswitch users. I did just copy and paste it and it seems to be working. Odd how it was missing. Thanks Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Denis Papes Sent: Friday, October 18, 2013 12:38 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows Download it again http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2 and retry. On Thu, Oct 17, 2013 at 10:17 PM, Andre > wrote: Mine is not there. Should I just copy it there and recompile? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Denis Papes Sent: Thursday, October 17, 2013 4:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows I successfully compiled 64-bit version using Visual Studio 2010. File nametab.h should be in same directory as xmltok.c, \libs\xmlrpc-c\lib\expat\xmltok. On Thu, Oct 17, 2013 at 9:15 PM, Andre > wrote: Hi, when I compile 64 bit I get this error Error 1 error C1083: Cannot open include file: 'nametab.h': No such file or directory \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c 10 1 xmltok Any idea what this means or how to fix it? Or does it even matter? Andre _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/a9ea2c44/attachment-0001.html From mcazzador at gmail.com Fri Oct 18 17:28:25 2013 From: mcazzador at gmail.com (Matteo Cazzador) Date: Fri, 18 Oct 2013 15:28:25 +0200 Subject: [Freeswitch-users] limit incoming concurrent fax/calls In-Reply-To: <52613618.3050705@softnet.si> References: <52613618.3050705@softnet.si> Message-ID: Great thank's a lot! i choose the first case. Is it valid for fax too i suppose, fax is a normal call from outside in fact? Today i will test this configuration. Bye 2013/10/18 Miha > Hi, > > it depends if you would like to limit inbound call from "outside" that you > will do this in public, if you would like to limit "local" calls that you > will do in in default dialplan. > > If you would like to limit bouth, than public and default:) > > miha > > Dne 10/18/2013 1:56 PM, pi?e Matteo Cazzador: > > Hi, i've configured a fax server (Centos 6.2) using ICT Fax with Plivo > Framework.I'm a novice about freeswitch. > All goes well, but i've a request. I need to limit concurrent incoming > calls (fax receive) max 50 contemporary incoming fax. > > I can use "limit" parameter (from freeswitch doumentation)? > > Is it correct to set it in > /usr/local/freeswiitch/conf/dialplan/default.xml or public.xml? > > I want that limit globally, max 50 concurrent calls incoming non for > single DID. > > Thank's a lot > > -- > Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail. > ****************************************** > Ing. Matteo Cazzador > Email: mcazzador at gmail.com > ****************************************** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail. ****************************************** Ing. Matteo Cazzador Email: mcazzador at gmail.com ****************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/6d27f4fc/attachment.html From mcazzador at gmail.com Fri Oct 18 17:28:25 2013 From: mcazzador at gmail.com (Matteo Cazzador) Date: Fri, 18 Oct 2013 15:28:25 +0200 Subject: [Freeswitch-users] limit incoming concurrent fax/calls In-Reply-To: <52613618.3050705@softnet.si> References: <52613618.3050705@softnet.si> Message-ID: Great thank's a lot! i choose the first case. Is it valid for fax too i suppose, fax is a normal call from outside in fact? Today i will test this configuration. Bye 2013/10/18 Miha > Hi, > > it depends if you would like to limit inbound call from "outside" that you > will do this in public, if you would like to limit "local" calls that you > will do in in default dialplan. > > If you would like to limit bouth, than public and default:) > > miha > > Dne 10/18/2013 1:56 PM, pi?e Matteo Cazzador: > > Hi, i've configured a fax server (Centos 6.2) using ICT Fax with Plivo > Framework.I'm a novice about freeswitch. > All goes well, but i've a request. I need to limit concurrent incoming > calls (fax receive) max 50 contemporary incoming fax. > > I can use "limit" parameter (from freeswitch doumentation)? > > Is it correct to set it in > /usr/local/freeswiitch/conf/dialplan/default.xml or public.xml? > > I want that limit globally, max 50 concurrent calls incoming non for > single DID. > > Thank's a lot > > -- > Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail. > ****************************************** > Ing. Matteo Cazzador > Email: mcazzador at gmail.com > ****************************************** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail. ****************************************** Ing. Matteo Cazzador Email: mcazzador at gmail.com ****************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/6d27f4fc/attachment-0001.html From victor.chukalovskiy at gmail.com Fri Oct 18 18:34:28 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Fri, 18 Oct 2013 10:34:28 -0400 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <5260D057.3060906@voxvalley.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> Message-ID: <526146F4.6070003@gmail.com> This is a dialplan manipulation, does not help. Need to set channel variables before hitting any dial-plan. On 13-10-18 02:08 AM, anand wrote: > You can set dynamic data in dial plan as below > > > > > Anand > > On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: >> Hi, yes, I already use all of these. The profile_name variable and >> putting different profiles into different contexts. >> >> But what I really need is to be able to set a couple custom variables >> per profile. This would allow to use the exact same context with >> different profiles....as well as couple other things. >> >> I guess it's not possible!!!! :-( >> >> On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: >>> if you run info() on the incoming call, you can see the profile's IP >>> address and port and profile name in various variables. >>> Then you can build the matching rules in your dialplan and >>> differentiate between the profiles, and set variables that you need. >>> >>> But probably simply assigning different contexts to your profiles >>> would do the job :) >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------ >>> *From:* Victor Chukalovskiy >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Sent:* Thursday, October 17, 2013 10:59 PM >>> *Subject:* Re: [Freeswitch-users] How to set channel variable per >>> SIP profile >>> >>> Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. >>> >>> Is it possible? >>> On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: >>>> you can set variables at the gateway level, and then they are set >>>> as channel variables for inbound calls: >>>> >>>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> *From:* Victor Chukalovskiy >>>> >>>> *To:* FreeSWITCH Users Help >>>> >>>> *Sent:* Thursday, October 17, 2013 8:36 PM >>>> *Subject:* [Freeswitch-users] How to set channel variable per SIP >>>> profile >>>> >>>> Hi, >>>> >>>> Subject says it all. Is there a way to set channel variable for all >>>> calls passing through a SIP profile? >>>> >>>> For example, I want all calls authorized with ACL and hitting SIP >>>> profile to have country=foo. This has to be done before hitting the >>>> dialplan. >>>> There is a bunch of "params" in the SIP profile, but these aren't >>>> channel variables. >>>> >>>> Thanks! >>>> -Victor >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/b9a69c22/attachment-0001.html From jeff at jefflenk.com Fri Oct 18 18:38:07 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 18 Oct 2013 07:38:07 -0700 (PDT) Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: <007901cecc05$b54df150$1fe9d3f0$@verizon.net> References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> <08b801cecb75$ed277000$c7765000$@verizon.net> <007901cecc05$b54df150$1fe9d3f0$@verizon.net> Message-ID: <1382107087643-7595825.post@n2.nabble.com> That is an autogenerated file. What is your build environment and how are you building? Using Visual Studio?? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freesiwtch-1-2-14-Windows-tp7595800p7595825.html Sent from the freeswitch-users mailing list archive at Nabble.com. From andretodd at verizon.net Fri Oct 18 18:46:43 2013 From: andretodd at verizon.net (Andre) Date: Fri, 18 Oct 2013 10:46:43 -0400 Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: <1382107087643-7595825.post@n2.nabble.com> References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> <08b801cecb75$ed277000$c7765000$@verizon.net> <007901cecc05$b54df150$1fe9d3f0$@verizon.net> <1382107087643-7595825.post@n2.nabble.com> Message-ID: <012a01cecc10$dd4717d0$97d54770$@verizon.net> VS 2012 with the updated Service Pack. I did a full compile in x86 first then a full in x64. Both have issues. I've complied it maybe 10 times, Clean, rebuild, build... I've tried it with 1 project to compile at a time, 8 and 16 same results for all. I downloaded the file from the download site based on the email that came out in freeswitch users. That is the only error I received. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Friday, October 18, 2013 10:38 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freesiwtch 1.2.14 Windows That is an autogenerated file. What is your build environment and how are you building? Using Visual Studio?? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freesiwtch-1-2-14-Windows-tp75 95800p7595825.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jnvines at gmail.com Fri Oct 18 18:49:13 2013 From: jnvines at gmail.com (Nick Vines) Date: Fri, 18 Oct 2013 07:49:13 -0700 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <526146F4.6070003@gmail.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> Message-ID: Two options that come to mind are the user_data command, which only works for registered endpoints, and the dialplan like anand suggested. The sip profile will dump calls into a context of your choice. Just make a file in there called 000000_set_variables.xml and set all of the variables. The 000000 is so that file will get executed first. On Fri, Oct 18, 2013 at 7:34 AM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > This is a dialplan manipulation, does not help. > > Need to set channel variables before hitting any dial-plan. > > > > On 13-10-18 02:08 AM, anand wrote: > > You can set dynamic data in dial plan as below > > > > > Anand > > On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: > > Hi, yes, I already use all of these. The profile_name variable and putting > different profiles into different contexts. > > But what I really need is to be able to set a couple custom variables per > profile. This would allow to use the exact same context with different > profiles....as well as couple other things. > > I guess it's not possible!!!! :-( > > On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: > > if you run info() on the incoming call, you can see the profile's IP > address and port and profile name in various variables. > Then you can build the matching rules in your dialplan and differentiate > between the profiles, and set variables that you need. > > But probably simply assigning different contexts to your profiles would do > the job :) > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, October 17, 2013 10:59 PM > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. > > Is it possible? > On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: > > you can set variables at the gateway level, and then they are set as > channel variables for inbound calls: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml > > > > > > > > > > > > > > > > > > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* FreeSWITCH Users Help > *Sent:* Thursday, October 17, 2013 8:36 PM > *Subject:* [Freeswitch-users] How to set channel variable per SIP profile > > Hi, > > Subject says it all. Is there a way to set channel variable for all > calls passing through a SIP profile? > > For example, I want all calls authorized with ACL and hitting SIP > profile to have country=foo. This has to be done before hitting the > dialplan. > There is a bunch of "params" in the SIP profile, but these aren't > channel variables. > > Thanks! > -Victor > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/57d46d09/attachment-0001.html From ssinyagin at yahoo.com Fri Oct 18 22:26:08 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Fri, 18 Oct 2013 11:26:08 -0700 (PDT) Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <526146F4.6070003@gmail.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> Message-ID: <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> but what's the problem with exporting needed variables in public context and transferring the call to the target context? Then you can reuse the target context as much as you need ________________________________ From: Victor Chukalovskiy To: freeswitch-users at lists.freeswitch.org Sent: Friday, October 18, 2013 4:34 PM Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile This is a dialplan manipulation, does not help. Need to set channel variables before hitting any dial-plan. On 13-10-18 02:08 AM, anand wrote: You can set dynamic data in dial plan as below Anand On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: Hi, yes, I already use all of these. The profile_name variable and putting different profiles into different contexts. But what I really need is to be able to set a couple custom variables per profile. This would allow to use the exact same context with different profiles....as well as couple other things. I guess it's not possible!!!!? :-( On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: if you run info() on the incoming call, you can see the profile's IP address and port and profile name in various variables. >Then you can build the matching rules in your dialplan and differentiate between the profiles, and set variables that you need. > >But probably simply assigning different contexts to your profiles would do the job :) > > > > > > > > > >________________________________ > From: Victor Chukalovskiy >To: freeswitch-users at lists.freeswitch.org >Sent: Thursday, October 17, 2013 10:59 PM >Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile > > > >Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. > >Is it possible? >On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: > >you can set variables at the gateway level, and then they? are set as channel variables for inbound calls: >> >>http://wiki.freeswitch.org/wiki/Sofia.conf.xml >> >> >> >> >> >> >> >> >>________________________________ >> From: Victor Chukalovskiy >>To: FreeSWITCH Users Help >>Sent: Thursday, October 17, 2013 8:36 PM >>Subject: [Freeswitch-users] How to set channel variable per SIP profile >> >> >>Hi, >> >>Subject says it all. Is there a way to set channel variable for all >>calls passing through a SIP profile? >> >>For example, I want all calls authorized with ACL and hitting SIP >>profile to have country=foo. This has to be done before hitting the >>dialplan. >>There is a bunch of "params" in the SIP profile, but these aren't >>channel variables. >> >>Thanks! >>-Victor >> >> >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> >> >> >> >>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > > > >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/4b7d5185/attachment-0001.html From andretodd at verizon.net Fri Oct 18 22:26:08 2013 From: andretodd at verizon.net (Andre) Date: Fri, 18 Oct 2013 14:26:08 -0400 Subject: [Freeswitch-users] park_after_bridge not working In-Reply-To: References: <0aaa01cec072$5538beb0$ffaa3c10$@verizon.net> Message-ID: <025501cecc2f$84c252e0$8e46f8a0$@verizon.net> Hmm is there another option to keep the leg open other than park? I want to keep the A leg open if they hang up within X seconds. Some providers require min of 6 second calls. Thanks Andre From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal Leeming [Simplicity Media Ltd] Sent: Friday, October 11, 2013 9:43 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] park_after_bridge not working Andre, As far as I know, park_after_bridge cannot be used because using bypass_media would take FS out of the media path. Have a read of this, slightly related; http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/037592.h tml Can you please explain what you mean by "doesn't work". Please include as much debugging info as possible using; sofia global siptrace on sofia loglevel all 9 sofia tracelevel alert console loglevel debug fsctl debug_level 10 Thanks Cal On Thu, Oct 3, 2013 at 8:54 PM, Andre > wrote: Hi, My Park_after_bridge doesn't work if I add bypass_media=true. How can I get Park to work when I'm bypass_media mode? Thanks _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/d3b4b062/attachment.html From smrdoshi at gmail.com Fri Oct 18 23:20:10 2013 From: smrdoshi at gmail.com (Samir Doshi) Date: Fri, 18 Oct 2013 12:20:10 -0700 (PDT) Subject: [Freeswitch-users] Call hangup not working in perl session In-Reply-To: <008d01cecaea$52b2a340$f817e9c0$@verizon.net> References: <1381849173815-7595714.post@n2.nabble.com> <006f01cec9bf$dc189c60$9449d520$@verizon.net> <1381860480301-7595722.post@n2.nabble.com> <009101cec9d3$fd1d6e90$f7584bb0$@verizon.net> <1381908592318-7595737.post@n2.nabble.com> <008d01cecaea$52b2a340$f817e9c0$@verizon.net> Message-ID: <1382124010548-7595829.post@n2.nabble.com> Thanks guys for your support. Its looks like working fine now without any change. Maybe temporary issue or something. I will keep monitor on it and get back to you in case i found anything. Thanks again. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-hangup-not-working-in-perl-session-tp7595714p7595829.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kris at kriskinc.com Fri Oct 18 23:21:33 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 18 Oct 2013 15:21:33 -0400 Subject: [Freeswitch-users] park_after_bridge not working In-Reply-To: <025501cecc2f$84c252e0$8e46f8a0$@verizon.net> References: <0aaa01cec072$5538beb0$ffaa3c10$@verizon.net> <025501cecc2f$84c252e0$8e46f8a0$@verizon.net> Message-ID: When the remote end hangs up the call is over. On Fri, Oct 18, 2013 at 2:26 PM, Andre wrote: > Hmm is there another option to keep the leg open other than park? > > > > I want to keep the A leg open if they hang up within X seconds. Some > providers require min of 6 second calls. > > > > Thanks > > Andre > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal > Leeming [Simplicity Media Ltd] > Sent: Friday, October 11, 2013 9:43 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] park_after_bridge not working > > > > Andre, > > > > As far as I know, park_after_bridge cannot be used because using > bypass_media would take FS out of the media path. > > > > Have a read of this, slightly related; > > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/037592.html > > > > Can you please explain what you mean by "doesn't work". > > > > Please include as much debugging info as possible using; > > > sofia global siptrace on > sofia loglevel all 9 > sofia tracelevel alert > console loglevel debug > fsctl debug_level 10 > > > > Thanks > > > Cal > > > > On Thu, Oct 3, 2013 at 8:54 PM, Andre wrote: > > Hi, > > > > My Park_after_bridge doesn?t work if I add bypass_media=true. How can I get > Park to work when I?m bypass_media mode? > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From michel at targointernet.com Fri Oct 18 23:44:55 2013 From: michel at targointernet.com (Michel Blais) Date: Fri, 18 Oct 2013 15:44:55 -0400 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: <002501cecae8$7b72277a$7c07000a@smb.curriegrad2004.ca> References: <002501cecae8$7b72277a$7c07000a@smb.curriegrad2004.ca> Message-ID: It seem like it builded fine with gawk instead of busybox's awk. Thank you both for your help. 2013/10/16 Jeff Leung > Install gawk and have that gawk's path take precedence over the default > path variable. > > Michel Blais wrote: > > Busybox don't include a full version of every command. It's a shell that > include most commande with only the most used option but often lack usefull > feature. It's use to save space on embedded device. > > I didn't know the -R option so I was not sur about what you where looking > for but the're no line with NAWK in the make file but if I look for AWK, I > get this : > > # grep AWK Makefile > $(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " $$1; \ > AWK = awk > $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1 } \ > $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1; } \ > $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ > $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ > $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ > > Thanks > > > > > 2013/10/16 Michael Jerris > > > Not sure why your grep is funky.. but the -R isn't necessary? This is > > going to be limitations of the tools in busybox I suspect. > > > > grep NAWK Makefile > > > > ? > > > > On Oct 16, 2013, at 2:01 PM, Michel Blais - Support technique - Targo > > communication wrote: > > > > > The awk command is the one include by busybox, same for grep. Could it > > > be the problem ? > > > > > > # grep -R ^NAWK Makefile > > > grep: invalid option -- 'R' > > > BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. > > > > > > Thanks > > > > > > Le 2013-10-16 16:50, Michael Jerris a ?crit : > > >> libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d > > 0x%04x\n", i, nopush[i] > > >> > > >> libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c > > $(TOP)/mkopcodeh.awk > > >> libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | > $(NAWK) > > -f $(TOP)/mkopcodeh.awk >opcodes.h > > >> > > >> <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h > > >> #define NOPUSH_MASK_0 0x3fbc > > >> #define NOPUSH_MASK_1 0x3e5b > > >> #define NOPUSH_MASK_2 0x71ef > > >> #define NOPUSH_MASK_3 0x7fce > > >> #define NOPUSH_MASK_4 0xffff > > >> #define NOPUSH_MASK_5 0xcdef > > >> #define NOPUSH_MASK_6 0xb6d7 > > >> #define NOPUSH_MASK_7 0x33af > > >> #define NOPUSH_MASK_8 0xf86f > > >> #define NOPUSH_MASK_9 0x0000 > > >> > > >> take a look at the contents of your generated opcodes.h and see what > > you find in there. The logic would be in mkopcodeh.awk. Also possible > you > > don't have awk installed? check out the definition of NAWK var: > > >> > > >> <11>:grep -R ^NAWK Makefile > > >> Makefile:NAWK = awk > > >> > > >> Mike > > >> > > >> > > >> On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo > > communication wrote: > > >> > > >>> Hi, > > >>> > > >>> I have a build error blocking at vdbeaux.c, build output following at > > >>> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a > > >>> Cavium octeon+ 50XX. I was not able to find anything on this build > > >>> error. It would be appreciated if anybody could help. > > >>> > > >>> Thanks > > >>> Michel > > >>> > > >>> # make > > >>> config.status: creating Makefile > > >>> config.status: creating include/apr.h > > >>> config.status: creating build/apr_rules.mk > > >>> config.status: creating build/pkg/pkginfo > > >>> config.status: creating apr-1-config > > >>> config.status: WARNING: 'apr-config.in' seems to ignore the > > >>> --datarootdir settg > > >>> config.status: creating apr.pc > > >>> config.status: creating test/Makefile > > >>> config.status: creating test/internal/Makefile > > >>> config.status: creating include/arch/unix/apr_private.h > > >>> config.status: executing libtool commands > > >>> rm: can't remove 'libtoolT': No such file or directory > > >>> config.status: executing default commands > > >>> config.status: include/apr.h is unchanged > > >>> config.status: include/arch/unix/apr_private.h is unchanged > > >>> touch src/include/switch.h > > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' > > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' > > >>> make[2]: Nothing to be done for `local-all'. > > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' > > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' > > >>> touch src/include/switch.h > > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' > > >>> Making all in xml/expat > > >>> make[2]: Entering directory > > `/usr/src/freeswitch/libs/apr-util/xml/expat' > > >>> make[3]: Entering directory > > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > > >>> make[3]: Nothing to be done for `all'. > > >>> make[3]: Leaving directory > > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > > >>> make[2]: Leaving directory > > `/usr/src/freeswitch/libs/apr-util/xml/expat' > > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' > > >>> make[2]: Nothing to be done for `local-all'. > > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > > >>> touch src/include/switch.h > > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c > > >>> ./mkkeywordhash >keywordhash.h > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./src/vdbeaux.c: In function 'opcodeNoPush': > > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported > > only > > >>> once > > >>> ./src/vdbeaux.c:247: error: for each function it appears in.) > > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in > > >>> this funct) > > >>> make[1]: *** [vdbeaux.lo] Error 1 > > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' > > >>> make: *** [libs/sqlite/libsqlite3.la] Error 2 > > >>> > > >>> > > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> FreeSWITCH-users mailing list > > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >> > > >> > > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/70fa5776/attachment-0001.html From andretodd at verizon.net Fri Oct 18 23:55:17 2013 From: andretodd at verizon.net (Andre) Date: Fri, 18 Oct 2013 15:55:17 -0400 Subject: [Freeswitch-users] park_after_bridge not working In-Reply-To: References: <0aaa01cec072$5538beb0$ffaa3c10$@verizon.net> <025501cecc2f$84c252e0$8e46f8a0$@verizon.net> Message-ID: <02c601cecc3b$f8c17d90$ea4478b0$@verizon.net> What about the caller hangs up and we don't send forward that message to the provider until ready? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Friday, October 18, 2013 3:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] park_after_bridge not working When the remote end hangs up the call is over. On Fri, Oct 18, 2013 at 2:26 PM, Andre wrote: > Hmm is there another option to keep the leg open other than park? > > > > I want to keep the A leg open if they hang up within X seconds. Some > providers require min of 6 second calls. > > > > Thanks > > Andre > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Cal Leeming [Simplicity Media Ltd] > Sent: Friday, October 11, 2013 9:43 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] park_after_bridge not working > > > > Andre, > > > > As far as I know, park_after_bridge cannot be used because using > bypass_media would take FS out of the media path. > > > > Have a read of this, slightly related; > > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/03 > 7592.html > > > > Can you please explain what you mean by "doesn't work". > > > > Please include as much debugging info as possible using; > > > sofia global siptrace on > sofia loglevel all 9 > sofia tracelevel alert > console loglevel debug > fsctl debug_level 10 > > > > Thanks > > > Cal > > > > On Thu, Oct 3, 2013 at 8:54 PM, Andre wrote: > > Hi, > > > > My Park_after_bridge doesn't work if I add bypass_media=true. How can > I get Park to work when I'm bypass_media mode? > > Thanks > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jkr888 at gmail.com Fri Oct 18 23:58:46 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Fri, 18 Oct 2013 15:58:46 -0400 Subject: [Freeswitch-users] How to failover with fail_on_single_reject construct Message-ID: I?m attempting to setup FS to failover from gw1, gw2, and gw3 subsequently. AND if any gateway response with NO_ANSWER or ALLOTED_TIMEOUT, then the next attempt should be stop. Using ?fail_on_single_reject? var, I tried following script : Bridge app still continue attempting call to each gw1, gw2, and gw3 when ALLOTED_TIMEOUT response. From searching lists, above dialplan should work out fine. Did I misses any steps? Btw, tried above on FreeSWITCH Version 1.5.6b+git~20131010T172322Z~1cd6d44b06 (git 1cd6d44 2013-10-10 17:23:22Z) windows built. Thanks, Jkwan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/b626c642/attachment.html From mike at jerris.com Sat Oct 19 00:03:28 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Oct 2013 16:03:28 -0400 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: References: <002501cecae8$7b72277a$7c07000a@smb.curriegrad2004.ca> Message-ID: <314788CF-EF56-4D10-9134-2B196EED1B47@jerris.com> Would you mind making sure the wiki has some notes on how to successfully build on busybox? On Oct 18, 2013, at 3:44 PM, Michel Blais wrote: > It seem like it builded fine with gawk instead of busybox's awk. > > Thank you both for your help. > > > 2013/10/16 Jeff Leung > Install gawk and have that gawk's path take precedence over the default path variable. > > Michel Blais wrote: > > Busybox don't include a full version of every command. It's a shell that > include most commande with only the most used option but often lack usefull > feature. It's use to save space on embedded device. > > I didn't know the -R option so I was not sur about what you where looking > for but the're no line with NAWK in the make file but if I look for AWK, I > get this : > > # grep AWK Makefile > $(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " $$1; \ > AWK = awk > $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1 } \ > $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1; } \ > $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ > $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ > $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ > > Thanks > > > > > 2013/10/16 Michael Jerris > > > Not sure why your grep is funky.. but the -R isn't necessary? This is > > going to be limitations of the tools in busybox I suspect. > > > > grep NAWK Makefile > > > > ? > > > > On Oct 16, 2013, at 2:01 PM, Michel Blais - Support technique - Targo > > communication wrote: > > > > > The awk command is the one include by busybox, same for grep. Could it > > > be the problem ? > > > > > > # grep -R ^NAWK Makefile > > > grep: invalid option -- 'R' > > > BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. > > > > > > Thanks > > > > > > Le 2013-10-16 16:50, Michael Jerris a ?crit : > > >> libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d > > 0x%04x\n", i, nopush[i] > > >> > > >> libs/sqlite/Makefile.in:373:opcodes.h: parse.h $(TOP)/src/vdbe.c > > $(TOP)/mkopcodeh.awk > > >> libs/sqlite/Makefile.in:374: cat parse.h $(TOP)/src/vdbe.c | $(NAWK) > > -f $(TOP)/mkopcodeh.awk >opcodes.h > > >> > > >> <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h > > >> #define NOPUSH_MASK_0 0x3fbc > > >> #define NOPUSH_MASK_1 0x3e5b > > >> #define NOPUSH_MASK_2 0x71ef > > >> #define NOPUSH_MASK_3 0x7fce > > >> #define NOPUSH_MASK_4 0xffff > > >> #define NOPUSH_MASK_5 0xcdef > > >> #define NOPUSH_MASK_6 0xb6d7 > > >> #define NOPUSH_MASK_7 0x33af > > >> #define NOPUSH_MASK_8 0xf86f > > >> #define NOPUSH_MASK_9 0x0000 > > >> > > >> take a look at the contents of your generated opcodes.h and see what > > you find in there. The logic would be in mkopcodeh.awk. Also possible you > > don't have awk installed? check out the definition of NAWK var: > > >> > > >> <11>:grep -R ^NAWK Makefile > > >> Makefile:NAWK = awk > > >> > > >> Mike > > >> > > >> > > >> On Oct 16, 2013, at 1:38 PM, Michel Blais - Support technique - Targo > > communication wrote: > > >> > > >>> Hi, > > >>> > > >>> I have a build error blocking at vdbeaux.c, build output following at > > >>> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, the CPU a > > >>> Cavium octeon+ 50XX. I was not able to find anything on this build > > >>> error. It would be appreciated if anybody could help. > > >>> > > >>> Thanks > > >>> Michel > > >>> > > >>> # make > > >>> config.status: creating Makefile > > >>> config.status: creating include/apr.h > > >>> config.status: creating build/apr_rules.mk > > >>> config.status: creating build/pkg/pkginfo > > >>> config.status: creating apr-1-config > > >>> config.status: WARNING: 'apr-config.in' seems to ignore the > > >>> --datarootdir settg > > >>> config.status: creating apr.pc > > >>> config.status: creating test/Makefile > > >>> config.status: creating test/internal/Makefile > > >>> config.status: creating include/arch/unix/apr_private.h > > >>> config.status: executing libtool commands > > >>> rm: can't remove 'libtoolT': No such file or directory > > >>> config.status: executing default commands > > >>> config.status: include/apr.h is unchanged > > >>> config.status: include/arch/unix/apr_private.h is unchanged > > >>> touch src/include/switch.h > > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' > > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' > > >>> make[2]: Nothing to be done for `local-all'. > > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' > > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' > > >>> touch src/include/switch.h > > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' > > >>> Making all in xml/expat > > >>> make[2]: Entering directory > > `/usr/src/freeswitch/libs/apr-util/xml/expat' > > >>> make[3]: Entering directory > > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > > >>> make[3]: Nothing to be done for `all'. > > >>> make[3]: Leaving directory > > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' > > >>> make[2]: Leaving directory > > `/usr/src/freeswitch/libs/apr-util/xml/expat' > > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' > > >>> make[2]: Nothing to be done for `local-all'. > > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' > > >>> touch src/include/switch.h > > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c > > >>> ./mkkeywordhash >keywordhash.h > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 > > >>> -DHAVE_USLEEP=1c > > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1 > > >>> -DHAVE_FDATASYo > > >>> ./src/vdbeaux.c: In function 'opcodeNoPush': > > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:247: error: (Each undeclared identifier is reported > > only > > >>> once > > >>> ./src/vdbeaux.c:247: error: for each function it appears in.) > > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared (first use in > > >>> this funct) > > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared (first use in > > >>> this funct) > > >>> make[1]: *** [vdbeaux.lo] Error 1 > > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' > > >>> make: *** [libs/sqlite/libsqlite3.la] Error 2 > > >>> > > >>> > > _________________________________________________________________________ > > >>> Professional FreeSWITCH Consulting Services: > > >>> consulting at freeswitch.org > > >>> http://www.freeswitchsolutions.com > > >>> > > >>> > > >>> > > >>> > > >>> Official FreeSWITCH Sites > > >>> http://www.freeswitch.org > > >>> http://wiki.freeswitch.org > > >>> http://www.cluecon.com > > >>> > > >>> FreeSWITCH-users mailing list > > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >>> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >>> http://www.freeswitch.org > > >> > > >> > > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/b76a3950/attachment-0001.html From michel at targointernet.com Sat Oct 19 00:54:49 2013 From: michel at targointernet.com (Michel Blais - Support technique - Targo communication) Date: Fri, 18 Oct 2013 16:54:49 -0400 Subject: [Freeswitch-users] Build error stop at vdbeaux.c In-Reply-To: <314788CF-EF56-4D10-9134-2B196EED1B47@jerris.com> References: <002501cecae8$7b72277a$7c07000a@smb.curriegrad2004.ca> <314788CF-EF56-4D10-9134-2B196EED1B47@jerris.com> Message-ID: <5261A019.3040902@targointernet.com> I already check and it seem like there nothing about installing on busybox. Seem like there only a init script compatible with it on the wiki. https://www.google.ca/search?q=wiki+freeswitch#q=site:wiki.freeswitch.org+busybox that lead me to this link : http://wiki.freeswitch.org/wiki/Freeswitch_init#BusyBox_Based_Systems The only error I found for now was to install gawk instead of using busybox's awk. The still some error starting freeswitch that seem related to user and permission. I'm debugging it right now. I will also try on openwrt, also on a MIPS CPU (Atheros AR71XX) just in case. Where should I send information for the wiki once it done ? Le 2013-10-18 16:03, Michael Jerris a ?crit : > Would you mind making sure the wiki has some notes on how to > successfully build on busybox? > > On Oct 18, 2013, at 3:44 PM, Michel Blais > wrote: > >> It seem like it builded fine with gawk instead of busybox's awk. >> >> Thank you both for your help. >> >> >> 2013/10/16 Jeff Leung > > >> >> Install gawk and have that gawk's path take precedence over the >> default path variable. >> >> Michel Blais > > wrote: >> >> Busybox don't include a full version of every command. It's a >> shell that >> include most commande with only the most used option but often >> lack usefull >> feature. It's use to save space on embedded device. >> >> I didn't know the -R option so I was not sur about what you where >> looking >> for but the're no line with NAWK in the make file but if I look >> for AWK, I >> get this : >> >> # grep AWK Makefile >> $(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " >> $$1; \ >> AWK = awk >> $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1 } \ >> $(AWK) 'BEGIN { files["."] = ""; dirs["."] = 1; } \ >> $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ >> $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ >> $(AWK) '{ files[$$0] = 1; nonempty = 1; } \ >> >> Thanks >> >> >> >> >> 2013/10/16 Michael Jerris > >> >> > Not sure why your grep is funky.. but the -R isn't necessary... >> This is >> > going to be limitations of the tools in busybox I suspect. >> > >> > grep NAWK Makefile >> > >> > ? >> > >> > On Oct 16, 2013, at 2:01 PM, Michel Blais - Support technique - >> Targo >> > communication > > wrote: >> > >> > > The awk command is the one include by busybox, same for grep. >> Could it >> > > be the problem ? >> > > >> > > # grep -R ^NAWK Makefile >> > > grep: invalid option -- 'R' >> > > BusyBox v1.19.0 (2012-08-28 21:32:52 PDT) multi-call binary. >> > > >> > > Thanks >> > > >> > > Le 2013-10-16 16:50, Michael Jerris a ?crit : >> > >> libs/sqlite/mkopcodeh.awk:125: printf "#define NOPUSH_MASK_%d >> > 0x%04x\n", i, nopush[i] >> > >> >> > >> libs/sqlite/Makefile.in:373:opcodes.h: parse.h >> $(TOP)/src/vdbe.c >> > $(TOP)/mkopcodeh.awk >> > >> libs/sqlite/Makefile.in:374: cat parse.h >> $(TOP)/src/vdbe.c | $(NAWK) >> > -f $(TOP)/mkopcodeh.awk >opcodes.h >> > >> >> > >> <7>:grep NOPUSH_MASK ./libs/sqlite/opcodes.h >> > >> #define NOPUSH_MASK_0 0x3fbc >> > >> #define NOPUSH_MASK_1 0x3e5b >> > >> #define NOPUSH_MASK_2 0x71ef >> > >> #define NOPUSH_MASK_3 0x7fce >> > >> #define NOPUSH_MASK_4 0xffff >> > >> #define NOPUSH_MASK_5 0xcdef >> > >> #define NOPUSH_MASK_6 0xb6d7 >> > >> #define NOPUSH_MASK_7 0x33af >> > >> #define NOPUSH_MASK_8 0xf86f >> > >> #define NOPUSH_MASK_9 0x0000 >> > >> >> > >> take a look at the contents of your generated opcodes.h and >> see what >> > you find in there. The logic would be in mkopcodeh.awk. Also >> possible you >> > don't have awk installed... check out the definition of NAWK var: >> > >> >> > >> <11>:grep -R ^NAWK Makefile >> > >> Makefile:NAWK = awk >> > >> >> > >> Mike >> > >> >> > >> >> > >> On Oct 16, 2013, at 1:38 PM, Michel Blais - Support >> technique - Targo >> > communication > > wrote: >> > >> >> > >>> Hi, >> > >>> >> > >>> I have a build error blocking at vdbeaux.c, build output >> following at >> > >>> bottom. OS is Vyatta based on debian 6.0.6. Arch is MIPS64, >> the CPU a >> > >>> Cavium octeon+ 50XX. I was not able to find anything on >> this build >> > >>> error. It would be appreciated if anybody could help. >> > >>> >> > >>> Thanks >> > >>> Michel >> > >>> >> > >>> # make >> > >>> config.status: creating Makefile >> > >>> config.status: creating include/apr.h >> > >>> config.status: creating build/apr_rules.mk >> >> > >>> config.status: creating build/pkg/pkginfo >> > >>> config.status: creating apr-1-config >> > >>> config.status: WARNING: 'apr-config.in >> ' seems to ignore the >> > >>> --datarootdir settg >> > >>> config.status: creating apr.pc >> > >>> config.status: creating test/Makefile >> > >>> config.status: creating test/internal/Makefile >> > >>> config.status: creating include/arch/unix/apr_private.h >> > >>> config.status: executing libtool commands >> > >>> rm: can't remove 'libtoolT': No such file or directory >> > >>> config.status: executing default commands >> > >>> config.status: include/apr.h is unchanged >> > >>> config.status: include/arch/unix/apr_private.h is unchanged >> > >>> touch src/include/switch.h >> > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr' >> > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr' >> > >>> make[2]: Nothing to be done for `local-all'. >> > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr' >> > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr' >> > >>> touch src/include/switch.h >> > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/apr-util' >> > >>> Making all in xml/expat >> > >>> make[2]: Entering directory >> > `/usr/src/freeswitch/libs/apr-util/xml/expat' >> > >>> make[3]: Entering directory >> > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' >> > >>> make[3]: Nothing to be done for `all'. >> > >>> make[3]: Leaving directory >> > >>> `/usr/src/freeswitch/libs/apr-util/xml/expat/lib' >> > >>> make[2]: Leaving directory >> > `/usr/src/freeswitch/libs/apr-util/xml/expat' >> > >>> make[2]: Entering directory `/usr/src/freeswitch/libs/apr-util' >> > >>> make[2]: Nothing to be done for `local-all'. >> > >>> make[2]: Leaving directory `/usr/src/freeswitch/libs/apr-util' >> > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/apr-util' >> > >>> touch src/include/switch.h >> > >>> make[1]: Entering directory `/usr/src/freeswitch/libs/sqlite' >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> sort -n -b -k 3 opcodes.h | awk -f ./mkopcodec.awk >opcodes.c >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> gcc -g -O2 -g -O2 -o mkkeywordhash ./tool/mkkeywordhash.c >> > >>> ./mkkeywordhash >keywordhash.h >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./libtool --tag=CC --mode=compile gcc -g -O2 -g -O2 -DOS_UNIX=1 >> > >>> -DHAVE_USLEEP=1c >> > >>> libtool: compile: gcc -g -O2 -g -O2 -DOS_UNIX=1 >> -DHAVE_USLEEP=1 >> > >>> -DHAVE_FDATASYo >> > >>> ./src/vdbeaux.c: In function 'opcodeNoPush': >> > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_0' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:247: error: (Each undeclared identifier is >> reported >> > only >> > >>> once >> > >>> ./src/vdbeaux.c:247: error: for each function it appears in.) >> > >>> ./src/vdbeaux.c:247: error: 'NOPUSH_MASK_1' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_2' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:248: error: 'NOPUSH_MASK_3' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_4' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:249: error: 'NOPUSH_MASK_5' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_6' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:250: error: 'NOPUSH_MASK_7' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_8' undeclared >> (first use in >> > >>> this funct) >> > >>> ./src/vdbeaux.c:251: error: 'NOPUSH_MASK_9' undeclared >> (first use in >> > >>> this funct) >> > >>> make[1]: *** [vdbeaux.lo] Error 1 >> > >>> make[1]: Leaving directory `/usr/src/freeswitch/libs/sqlite' >> > >>> make: *** [libs/sqlite/libsqlite3.la >> ] Error 2 >> > >>> >> > >>> >> > >> _________________________________________________________________________ >> > >>> Professional FreeSWITCH Consulting Services: >> > >>> consulting at freeswitch.org >> > >>> http://www.freeswitchsolutions.com >> >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> Official FreeSWITCH Sites >> > >>> http://www.freeswitch.org >> > >>> http://wiki.freeswitch.org >> > >>> http://www.cluecon.com >> > >>> >> > >>> FreeSWITCH-users mailing list >> > >>> FreeSWITCH-users at lists.freeswitch.org >> >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >>> UNSUBSCRIBE: >> > http://lists.freeswitch.org/mailman/options/freeswitch-users >> > >>> http://www.freeswitch.org >> > >> >> > >> >> > >> _________________________________________________________________________ >> > >> Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> > >> http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > >> Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> > >> http://wiki.freeswitch.org >> > >> http://www.cluecon.com >> > >> >> > >> FreeSWITCH-users mailing list >> > >> FreeSWITCH-users at lists.freeswitch.org >> >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE: >> > http://lists.freeswitch.org/mailman/options/freeswitch-users >> > >> http://www.freeswitch.org >> > > >> > > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131018/d88f33c5/attachment-0001.html From hardyanto.donny at gmail.com Sat Oct 19 12:47:17 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Sat, 19 Oct 2013 15:47:17 +0700 Subject: [Freeswitch-users] when using xml_curl, the sofia cannot find user spesific gateway Message-ID: Hi, FIrst I want to thank you to all member of freeswitch developer team. Freeswitch Rocks! You all have done terrific job! :) I have buy and read all the 3 Freeswitch books. Hopefully there will be another Freeswitch book that can uncover all the hidden features. The current books still only scratch the surface. I am now currently using freeswitch to do a surrogate client registration. The operator softswitch that I am connecting the freeswitch into require that all my freeswitch clients registered with them, otherwise it cannot receive the call from the softswitch. The problem is the client is dynamic and required that when client is not registered with the freeswitch, they must not registered with the operator softswitch. So only when the client is registered with the freeswitch, they must also registered with the softswitch, and when they are not, they also must not registered with the softswitch. For that problem I cannot use SIP trunking to the operator softswitch. I turn to xml_curl module and also read that the user can have own gateway on directory (from freeswitch web documentation). So I think i can use this as surrogate client registration. So I create this xml to be returned when xml_curl request the user data for registration:
The user register with freeswitch, and then freeswitch tried to register the gw_1682112207445, and complain it cannot find the gw_1682112207445. nta.c:2791 agent_recv_request() nta: received REGISTER sip:xxxx.net SIP/2.0 (CSeq 260) nta.c:3078 agent_check_request_via() nta: Via check: received=182.4.88.15 nta.c:2990 agent_recv_request() nta: REGISTER (260) going to a default leg 2013-10-19 15:42:22.468092 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5f331fdc-389a-11e3-8e18-618d0dd75901.tmp.xml *2013-10-19 15:42:22.468092 [WARNING] sofia_reg.c:2851 Gateway 'gw_1682112207445' not found.* Is the user gateway is not supported on xml_curl or I am missing something? Regards, Donny Hardyanto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131019/d380d1b6/attachment.html From pranav.lal at gmail.com Sat Oct 19 15:27:42 2013 From: pranav.lal at gmail.com (Pranav Lal) Date: Sat, 19 Oct 2013 16:57:42 +0530 Subject: [Freeswitch-users] Unable to start freeswitch: getting stuck at adding application deadlock Message-ID: <000c01ceccbe$3cc1a7e0$b644f7a0$@gmail.com> Hi all, I have updated to the latest version of freeswitch (I can't tell what it is) using the make current command. Whenever I start freeswitch, it gets stuck at adding application deadlock. I saw a thread regarding this where the suggestion was to delete all the files in the /usr/local/freeswitch/db directory which I did. No joy. Here is what I get on my screen when I run freeswitch. 2013-10-19 11:18:49.448654 [NOTICE] switch_loadable_module.c:357 Adding File Format 'fsv' 2013-10-19 11:18:49.456007 [CRIT] switch_loadable_module.c:1438 Error Loading module /usr/ local/freeswitch/mod/mod_cluechoo.so **/usr/local/freeswitch/mod/mod_cluechoo.so: undefined symbol: waddch** 2013-10-19 11:18:49.463890 [CONSOLE] switch_loadable_module.c:1456 Successfully Loaded [mo d_valet_parking] 2013-10-19 11:18:49.464202 [NOTICE] switch_loadable_module.c:261 Adding Application 'valet _park' 2013-10-19 11:18:49.464940 [NOTICE] switch_loadable_module.c:307 Adding API Function 'vale t_info' 2013-10-19 11:18:49.482006 [NOTICE] mod_httapi.c:2137 Profile [default] JSON Function [htt p://www.freeswitch.org/api/index.cgi] 2013-10-19 11:18:49.482500 [CONSOLE] switch_loadable_module.c:1456 Successfully Loaded [mo d_httapi] 2013-10-19 11:18:49.482634 [NOTICE] switch_loadable_module.c:261 Adding Application 'httap i' 2013-10-19 11:18:49.483210 [NOTICE] switch_loadable_module.c:307 Adding API Function 'htta pi' 2013-10-19 11:18:49.483724 [NOTICE] switch_loadable_module.c:357 Adding File Format 'http' 2013-10-19 11:18:49.484302 [NOTICE] switch_loadable_module.c:357 Adding File Format 'https ' 2013-10-19 11:18:49.493325 [CONSOLE] switch_loadable_module.c:1456 Successfully Loaded [mo d_dialplan_xml] 2013-10-19 11:18:49.493541 [NOTICE] switch_loadable_module.c:219 Adding Dialplan 'XML' 2013-10-19 11:18:49.499107 [CONSOLE] switch_loadable_module.c:1456 Successfully Loaded [mo d_dialplan_asterisk] 2013-10-19 11:18:49.499325 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'SIP' 2013-10-19 11:18:49.499908 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'IAX2' 2013-10-19 11:18:49.500417 [NOTICE] switch_loadable_module.c:219 Adding Dialplan 'asterisk ' 2013-10-19 11:18:49.501107 [NOTICE] switch_loadable_module.c:261 Adding Application 'Dial' 2013-10-19 11:18:49.501629 [NOTICE] switch_loadable_module.c:261 Adding Application 'Goto' 2013-10-19 11:18:49.502141 [NOTICE] switch_loadable_module.c:261 Adding Application 'Avoid ingDeadlock' Pranav From andretodd at verizon.net Sat Oct 19 17:00:07 2013 From: andretodd at verizon.net (Andre) Date: Sat, 19 Oct 2013 09:00:07 -0400 Subject: [Freeswitch-users] Limit Message-ID: <049c01cecccb$237a7b60$6a6f7220$@verizon.net> Hi, how do I set the CPS and Port limits on a bridge Can I set both on one application or do I need to set 2 limits? 5 ports 5 cps Can I do this for both 5 ports and 5 cps? From jeff at jefflenk.com Sun Oct 20 08:25:35 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Sat, 19 Oct 2013 21:25:35 -0700 (PDT) Subject: [Freeswitch-users] Freesiwtch 1.2.14 Windows In-Reply-To: <012a01cecc10$dd4717d0$97d54770$@verizon.net> References: <085301cecb6d$464824d0$d2d86e70$@verizon.net> <08b801cecb75$ed277000$c7765000$@verizon.net> <007901cecc05$b54df150$1fe9d3f0$@verizon.net> <1382107087643-7595825.post@n2.nabble.com> <012a01cecc10$dd4717d0$97d54770$@verizon.net> Message-ID: <1382243135621-7595841.post@n2.nabble.com> I'm not able to reproduce this. Would you please file a Jira and attach a full log of building the solution from a clean directory. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freesiwtch-1-2-14-Windows-tp7595800p7595841.html Sent from the freeswitch-users mailing list archive at Nabble.com. From andrew at cassidywebservices.co.uk Sun Oct 20 11:56:51 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sun, 20 Oct 2013 08:56:51 +0100 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> Message-ID: Just referring to another thread related to this, the reason it needs to be set BEFORE the dialplan is he wants to use mod_translate as a dialplan module as I have. On 18 October 2013 19:26, Stanislav Sinyagin wrote: > but what's the problem with exporting needed variables in public context > and transferring the call to the target context? > Then you can reuse the target context as much as you need > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Friday, October 18, 2013 4:34 PM > > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > This is a dialplan manipulation, does not help. > > Need to set channel variables before hitting any dial-plan. > > > On 13-10-18 02:08 AM, anand wrote: > > You can set dynamic data in dial plan as below > > > > > Anand > > On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: > > Hi, yes, I already use all of these. The profile_name variable and putting > different profiles into different contexts. > > But what I really need is to be able to set a couple custom variables per > profile. This would allow to use the exact same context with different > profiles....as well as couple other things. > > I guess it's not possible!!!! :-( > > On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: > > if you run info() on the incoming call, you can see the profile's IP > address and port and profile name in various variables. > Then you can build the matching rules in your dialplan and differentiate > between the profiles, and set variables that you need. > > But probably simply assigning different contexts to your profiles would do > the job :) > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, October 17, 2013 10:59 PM > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. > > Is it possible? > On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: > > you can set variables at the gateway level, and then they are set as > channel variables for inbound calls: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml > > > > > > > > > > > > > > > > > > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* FreeSWITCH Users Help > *Sent:* Thursday, October 17, 2013 8:36 PM > *Subject:* [Freeswitch-users] How to set channel variable per SIP profile > > Hi, > > Subject says it all. Is there a way to set channel variable for all > calls passing through a SIP profile? > > For example, I want all calls authorized with ACL and hitting SIP > profile to have country=foo. This has to be done before hitting the > dialplan. > There is a bunch of "params" in the SIP profile, but these aren't > channel variables. > > Thanks! > -Victor > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/7a036591/attachment-0001.html From juanito1982 at gmail.com Sun Oct 20 19:30:42 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sun, 20 Oct 2013 17:30:42 +0200 Subject: [Freeswitch-users] Bad request answer In-Reply-To: References: Message-ID: New routers solved the problem. The old one seems to modify SDP. New INVITE requests seems as: http://pastebin.com/hQFjDC2c We can see local IPv4 ips instead the public ones. Whan I don't know is why FS didn't like previous requests. Is there any way to know why? Regards 2013/10/18 Anthony Minessale > Maybe try factory resetting the device? > > > > > On Fri, Oct 18, 2013 at 6:24 AM, Juan Antonio Iba?ez Santorum < > juanito1982 at gmail.com> wrote: > >> Hello, >> >> I am getting "Bad request" answer from FS but I don't know why. I have >> phone as that (AT810) in other locations without problems. You can see one >> INVITE and answer here >> >> http://pastebin.com/Widd5j2i >> >> Do you know why FS doesn't like it? >> Regards >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/cac0d291/attachment.html From victor.chukalovskiy at gmail.com Sun Oct 20 19:49:42 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Sun, 20 Oct 2013 11:49:42 -0400 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> Message-ID: <5263FB96.6030807@gmail.com> Andrew, that's right. Indicating "country" to hit the right mod_translate profile is one example. However, there are other cases when I want to set a variable per SIP profile. For example, I want several profiles to use the same dialplan context but use different limit values: In this example I need to set ${value_max_calls} on per SIP profile basis so that I don't have to do different dial-plan contexts for different profiles. On 13-10-20 03:56 AM, Andrew Cassidy wrote: > Just referring to another thread related to this, the reason it needs > to be set BEFORE the dialplan is he wants to use mod_translate as a > dialplan module as I have. > > > On 18 October 2013 19:26, Stanislav Sinyagin > wrote: > > but what's the problem with exporting needed variables in public > context and transferring the call to the target context? > Then you can reuse the target context as much as you need > > > ------------------------------------------------------------------------ > *From:* Victor Chukalovskiy > > *To:* freeswitch-users at lists.freeswitch.org > > *Sent:* Friday, October 18, 2013 4:34 PM > > *Subject:* Re: [Freeswitch-users] How to set channel variable per > SIP profile > > This is a dialplan manipulation, does not help. > > Need to set channel variables before hitting any dial-plan. > > > On 13-10-18 02:08 AM, anand wrote: > You can set dynamic data in dial plan as below > > > > > Anand > > On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: > Hi, yes, I already use all of these. The profile_name variable and > putting different profiles into different contexts. > > But what I really need is to be able to set a couple custom > variables per profile. This would allow to use the exact same > context with different profiles....as well as couple other things. > > I guess it's not possible!!!! :-( > > On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: >> if you run info() on the incoming call, you can see the profile's >> IP address and port and profile name in various variables. >> Then you can build the matching rules in your dialplan and >> differentiate between the profiles, and set variables that you need. >> >> But probably simply assigning different contexts to your profiles >> would do the job :) >> >> >> >> >> >> ------------------------------------------------------------------------ >> *From:* Victor Chukalovskiy >> >> *To:* freeswitch-users at lists.freeswitch.org >> >> *Sent:* Thursday, October 17, 2013 10:59 PM >> *Subject:* Re: [Freeswitch-users] How to set channel variable per >> SIP profile >> >> Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. >> >> Is it possible? >> On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: >>> you can set variables at the gateway level, and then they are >>> set as channel variables for inbound calls: >>> >>> http://wiki.freeswitch.org/wiki/Sofia.conf.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------ >>> *From:* Victor Chukalovskiy >>> >>> *To:* FreeSWITCH Users Help >>> >>> >>> *Sent:* Thursday, October 17, 2013 8:36 PM >>> *Subject:* [Freeswitch-users] How to set channel variable per >>> SIP profile >>> >>> Hi, >>> >>> Subject says it all. Is there a way to set channel variable for all >>> calls passing through a SIP profile? >>> >>> For example, I want all calls authorized with ACL and hitting SIP >>> profile to have country=foo. This has to be done before hitting the >>> dialplan. >>> There is a bunch of "params" in the SIP profile, but these aren't >>> channel variables. >>> >>> Thanks! >>> -Victor >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/1dbe02b9/attachment-0001.html From idokan at gmail.com Sun Oct 20 19:50:48 2013 From: idokan at gmail.com (ik) Date: Sun, 20 Oct 2013 17:50:48 +0200 Subject: [Freeswitch-users] newbie: bridge control Message-ID: Hello List, I'm a newbie to Freeswitch. I need to bridge an inbound call to another call, and detect pattren of DTMF. If the the pattern happen, I need to capture it to a variable, and hangup leg B (the bridged one), and continue doing stuff with Leg A. If Leg B was disconnected, I need hangup leg A as well. I wish to better understand how to look such thing on the wiki, prior for direct answer on how to do it, in order to better learn FS and it's usage :) So any help on this matter is more then welcome. Thanks Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/88b9cad3/attachment.html From steveayre at gmail.com Sun Oct 20 21:21:44 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 20 Oct 2013 18:21:44 +0100 Subject: [Freeswitch-users] Bad request answer In-Reply-To: References: Message-ID: That internal IP in the SDP will cause one-way audio problems. RTP auto-adjust can help but you'll not hear audio until you start sending. Check if the ATCOM client can support STUN. Previously your router was attempting to correct this for you with 'SIP ALG', but that can't help with TLS and as you saw isn't perfect, so STUN would be a better solution. On 20 October 2013 16:30, Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com> wrote: > New routers solved the problem. The old one seems to modify SDP. New > INVITE requests seems as: > > http://pastebin.com/hQFjDC2c > > We can see local IPv4 ips instead the public ones. Whan I don't know is > why FS didn't like previous requests. Is there any way to know why? > > Regards > > > 2013/10/18 Anthony Minessale > >> Maybe try factory resetting the device? >> >> >> >> >> On Fri, Oct 18, 2013 at 6:24 AM, Juan Antonio Iba?ez Santorum < >> juanito1982 at gmail.com> wrote: >> >>> Hello, >>> >>> I am getting "Bad request" answer from FS but I don't know why. I have >>> phone as that (AT810) in other locations without problems. You can see one >>> INVITE and answer here >>> >>> http://pastebin.com/Widd5j2i >>> >>> Do you know why FS doesn't like it? >>> Regards >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/29ed9561/attachment.html From steveayre at gmail.com Sun Oct 20 21:26:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 20 Oct 2013 18:26:18 +0100 Subject: [Freeswitch-users] Bad request answer In-Reply-To: References: Message-ID: Looks like the old router changed the IP in the SDP without updating the Content-Length*, so the SDP length was different from what the packet claimed hence the error. *The public IP is 3 characters shorter than the internal one. The SDP is otherwise identical but both have 258 as content-length. The router was removing 6 bytes from the SDP so should have changed this to 252. On 20 October 2013 16:30, Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com> wrote: > New routers solved the problem. The old one seems to modify SDP. New > INVITE requests seems as: > > http://pastebin.com/hQFjDC2c > > We can see local IPv4 ips instead the public ones. Whan I don't know is > why FS didn't like previous requests. Is there any way to know why? > > Regards > > > 2013/10/18 Anthony Minessale > >> Maybe try factory resetting the device? >> >> >> >> >> On Fri, Oct 18, 2013 at 6:24 AM, Juan Antonio Iba?ez Santorum < >> juanito1982 at gmail.com> wrote: >> >>> Hello, >>> >>> I am getting "Bad request" answer from FS but I don't know why. I have >>> phone as that (AT810) in other locations without problems. You can see one >>> INVITE and answer here >>> >>> http://pastebin.com/Widd5j2i >>> >>> Do you know why FS doesn't like it? >>> Regards >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/35de88ae/attachment-0001.html From ssinyagin at yahoo.com Sun Oct 20 23:17:49 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Sun, 20 Oct 2013 12:17:49 -0700 (PDT) Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <5263FB96.6030807@gmail.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> <5263FB96.6030807@gmail.com> Message-ID: <1382296669.39435.YahooMailNeo@web126204.mail.ne1.yahoo.com> both Translate and Limit are parts of the dialplan, so nothing prevents you from setting all needed variables in the same dialplan in an extension with continue="true", and processing them in the following extensions. something like ????? ????? ??? ????? ????? ??? so, what's the real problem? ________________________________ From: Victor Chukalovskiy To: freeswitch-users at lists.freeswitch.org Sent: Sunday, October 20, 2013 5:49 PM Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile Andrew, that's right. Indicating "country" to hit the right mod_translate profile is one example. However, there are other cases when I want to set a variable per SIP profile. For example, I want several profiles to use the same dialplan context but use different limit values: In this example I need to set ${value_max_calls} on per SIP profile basis so that I don't have to do different dial-plan contexts for different profiles. On 13-10-20 03:56 AM, Andrew Cassidy wrote: Just referring to another thread related to this, the reason it needs to be set BEFORE the dialplan is he wants to use mod_translate as a dialplan module as I have. > > > >On 18 October 2013 19:26, Stanislav Sinyagin wrote: > >but what's the problem with exporting needed variables in public context and transferring the call to the target context? >>Then you can reuse the target context as much as you need >> >> >> >> >> >> >>________________________________ >> >>From: Victor Chukalovskiy >>To: freeswitch-users at lists.freeswitch.org >>Sent: Friday, October 18, 2013 4:34 PM >> >>Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile >> >> >> >>This is a dialplan manipulation, does not help. >> >>Need to set channel variables before hitting any dial-plan. >> >> >> >>On 13-10-18 02:08 AM, anand wrote: >> >>You can set dynamic data in dial plan as below >> >> >> >> >>Anand >> >>On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: >> >>Hi, yes, I already use all of these. The profile_name variable and putting different profiles into different contexts. >> >>But what I really need is to be able to set a couple custom variables per profile. This would allow to use the exact same context with different profiles....as well as couple other things. >> >>I guess it's not possible!!!!? :-( >> >>On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: >> >>if you run info() on the incoming call, you can see the profile's IP address and port and profile name in various variables. >>>Then you can build the matching rules in your dialplan and differentiate between the profiles, and set variables that you need. >>> >>>But probably simply assigning different contexts to your profiles would do the job :) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>________________________________ >>> From: Victor Chukalovskiy >>>To: freeswitch-users at lists.freeswitch.org >>>Sent: Thursday, October 17, 2013 10:59 PM >>>Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile >>> >>> >>> >>>Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. >>> >>>Is it possible? >>>On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: >>> >>>you can set variables at the gateway level, and then they? are set as channel variables for inbound calls: >>>> >>>>http://wiki.freeswitch.org/wiki/Sofia.conf.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>________________________________ >>>> From: Victor Chukalovskiy >>>>To: FreeSWITCH Users Help >>>>Sent: Thursday, October 17, 2013 8:36 PM >>>>Subject: [Freeswitch-users] How to set channel variable per SIP profile >>>> >>>> >>>>Hi, >>>> >>>>Subject says it all. Is there a way to set channel variable for all >>>>calls passing through a SIP profile? >>>> >>>>For example, I want all calls authorized with ACL and hitting SIP >>>>profile to have country=foo. This has to be done before hitting the >>>>dialplan. >>>>There is a bunch of "params" in the SIP profile, but these aren't >>>>channel variables. >>>> >>>>Thanks! >>>>-Victor >>>> >>>> >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting at freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> >>>> >>>>Official FreeSWITCH Sites >>>>http://www.freeswitch.org >>>>http://wiki.freeswitch.org >>>>http://www.cluecon.com >>>> >>>>FreeSWITCH-users mailing list >>>>FreeSWITCH-users at lists.freeswitch.org >>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> >>>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >>> >>> >>> >>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> >> >>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> >> >>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > > > -- >Andrew Cassidy BSc (Hons) MBCS SSCA >Managing Director > > > > >T?03300 100 960? F?03300 100 961 >E?andrew at cassidywebservices.co.uk >W?www.cassidywebservices.co.uk > > >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/05d26e21/attachment-0001.html From andrew at cassidywebservices.co.uk Mon Oct 21 00:29:34 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sun, 20 Oct 2013 21:29:34 +0100 Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: <1382296669.39435.YahooMailNeo@web126204.mail.ne1.yahoo.com> References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> <5263FB96.6030807@gmail.com> <1382296669.39435.YahooMailNeo@web126204.mail.ne1.yahoo.com> Message-ID: in sofia profile, translate is run before the xml dialplan is even considered. On 20 October 2013 20:17, Stanislav Sinyagin wrote: > both Translate and Limit are parts of the dialplan, so nothing prevents > you from setting all needed variables in the same dialplan in an extension > with continue="true", and processing them in the following extensions. > > something like > > > > > > > > > > > > > > so, what's the real problem? > > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Sunday, October 20, 2013 5:49 PM > > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > Andrew, that's right. Indicating "country" to hit the right mod_translate > profile is one example. > > However, there are other cases when I want to set a variable per SIP > profile. For example, I want several profiles to use the same dialplan > context but use different limit values: > > > > In this example I need to set ${value_max_calls} on per SIP profile basis > so that I don't have to do different dial-plan contexts for different > profiles. > > On 13-10-20 03:56 AM, Andrew Cassidy wrote: > > Just referring to another thread related to this, the reason it needs to > be set BEFORE the dialplan is he wants to use mod_translate as a dialplan > module as I have. > > > On 18 October 2013 19:26, Stanislav Sinyagin wrote: > > but what's the problem with exporting needed variables in public context > and transferring the call to the target context? > Then you can reuse the target context as much as you need > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Friday, October 18, 2013 4:34 PM > > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > This is a dialplan manipulation, does not help. > > Need to set channel variables before hitting any dial-plan. > > > On 13-10-18 02:08 AM, anand wrote: > > You can set dynamic data in dial plan as below > > > > > Anand > > On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: > > Hi, yes, I already use all of these. The profile_name variable and > putting different profiles into different contexts. > > But what I really need is to be able to set a couple custom variables per > profile. This would allow to use the exact same context with different > profiles....as well as couple other things. > > I guess it's not possible!!!! :-( > > On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: > > if you run info() on the incoming call, you can see the profile's IP > address and port and profile name in various variables. > Then you can build the matching rules in your dialplan and differentiate > between the profiles, and set variables that you need. > > But probably simply assigning different contexts to your profiles would do > the job :) > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, October 17, 2013 10:59 PM > *Subject:* Re: [Freeswitch-users] How to set channel variable per SIP > profile > > Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. > > Is it possible? > On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: > > you can set variables at the gateway level, and then they are set as > channel variables for inbound calls: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml > > > > > > > > > > > > > > > > > > > > > > > ------------------------------ > *From:* Victor Chukalovskiy > *To:* FreeSWITCH Users Help > *Sent:* Thursday, October 17, 2013 8:36 PM > *Subject:* [Freeswitch-users] How to set channel variable per SIP profile > > Hi, > > Subject says it all. Is there a way to set channel variable for all > calls passing through a SIP profile? > > For example, I want all calls authorized with ACL and hitting SIP > profile to have country=foo. This has to be done before hitting the > dialplan. > There is a bunch of "params" in the SIP profile, but these aren't > channel variables. > > Thanks! > -Victor > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/52738ecf/attachment-0001.html From ssinyagin at yahoo.com Mon Oct 21 01:34:22 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Sun, 20 Oct 2013 14:34:22 -0700 (PDT) Subject: [Freeswitch-users] How to set channel variable per SIP profile In-Reply-To: References: <52602E47.5050803@gmail.com> <1382043013.31446.YahooMailNeo@web126203.mail.ne1.yahoo.com> <52604FA0.1020202@gmail.com> <1382047176.4841.YahooMailNeo@web126205.mail.ne1.yahoo.com> <52609D74.20906@gmail.com> <5260D057.3060906@voxvalley.com> <526146F4.6070003@gmail.com> <1382120768.9154.YahooMailNeo@web126204.mail.ne1.yahoo.com> <5263FB96.6030807@gmail.com> <1382296669.39435.YahooMailNeo@web126204.mail.ne1.yahoo.com> Message-ID: <1382304862.20968.YahooMailNeo@web126203.mail.ne1.yahoo.com> hmm, ok, you're right :) someone should then open a feature request in jira ________________________________ From: Andrew Cassidy To: FreeSWITCH Users Help Sent: Sunday, October 20, 2013 10:29 PM Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile in sofia profile, translate is run before the xml dialplan is even considered. On 20 October 2013 20:17, Stanislav Sinyagin wrote: both Translate and Limit are parts of the dialplan, so nothing prevents you from setting all needed variables in the same dialplan in an extension with continue="true", and processing them in the following extensions. > >something like > > > > ????? ????? ??? > ????? ????? ??? > > >so, what's the real problem? > > > > > > > > > > >________________________________ > >From: Victor Chukalovskiy >To: freeswitch-users at lists.freeswitch.org >Sent: Sunday, October 20, 2013 5:49 PM > >Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile > > > >Andrew, that's right. Indicating "country" to hit the right mod_translate profile is one example. > >However, there are other cases when I want to set a variable per SIP profile. For example, I want several profiles to use the same dialplan context but use different limit values: > > > >In this example I need to set ${value_max_calls} on per SIP profile basis so that I don't have to do different dial-plan contexts for different profiles. > > On 13-10-20 03:56 AM, Andrew Cassidy wrote: > >Just referring to another thread related to this, the reason it needs to be set BEFORE the dialplan is he wants to use mod_translate as a dialplan module as I have. >> >> >> >>On 18 October 2013 19:26, Stanislav Sinyagin wrote: >> >>but what's the problem with exporting needed variables in public context and transferring the call to the target context? >>>Then you can reuse the target context as much as you need >>> >>> >>> >>> >>> >>> >>>________________________________ >>> >>>From: Victor Chukalovskiy >>>To: freeswitch-users at lists.freeswitch.org >>>Sent: Friday, October 18, 2013 4:34 PM >>> >>>Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile >>> >>> >>> >>>This is a dialplan manipulation, does not help. >>> >>>Need to set channel variables before hitting any dial-plan. >>> >>> >>> >>>On 13-10-18 02:08 AM, anand wrote: >>> >>>You can set dynamic data in dial plan as below >>> >>> >>> >>> >>>Anand >>> >>>On 10/18/2013 8:01 AM, Victor Chukalovskiy wrote: >>> >>>Hi, yes, I already use all of these. The profile_name variable and putting different profiles into different contexts. >>> >>>But what I really need is to be able to set a couple custom variables per profile. This would allow to use the exact same context with different profiles....as well as couple other things. >>> >>>I guess it's not possible!!!!? :-( >>> >>>On 13-10-17 05:59 PM, Stanislav Sinyagin wrote: >>> >>>if you run info() on the incoming call, you can see the profile's IP address and port and profile name in various variables. >>>>Then you can build the matching rules in your dialplan and differentiate between the profiles, and set variables that you need. >>>> >>>>But probably simply assigning different contexts to your profiles would do the job :) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>________________________________ >>>> From: Victor Chukalovskiy >>>>To: freeswitch-users at lists.freeswitch.org >>>>Sent: Thursday, October 17, 2013 10:59 PM >>>>Subject: Re: [Freeswitch-users] How to set channel variable per SIP profile >>>> >>>> >>>> >>>>Hmm, no gateways involved. Only SIP profile and apply-inbound-acl. >>>> >>>>Is it possible? >>>>On 13-10-17 04:50 PM, Stanislav Sinyagin wrote: >>>> >>>>you can set variables at the gateway level, and then they? are set as channel variables for inbound calls: >>>>> >>>>>http://wiki.freeswitch.org/wiki/Sofia.conf.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>________________________________ >>>>> From: Victor Chukalovskiy >>>>>To: FreeSWITCH Users Help >>>>>Sent: Thursday, October 17, 2013 8:36 PM >>>>>Subject: [Freeswitch-users] How to set channel variable per SIP profile >>>>> >>>>> >>>>>Hi, >>>>> >>>>>Subject says it all. Is there a way to set channel variable for all >>>>>calls passing through a SIP profile? >>>>> >>>>>For example, I want all calls authorized with ACL and hitting SIP >>>>>profile to have country=foo. This has to be done before hitting the >>>>>dialplan. >>>>>There is a bunch of "params" in the SIP profile, but these aren't >>>>>channel variables. >>>>> >>>>>Thanks! >>>>>-Victor >>>>> >>>>> >>>>> >>>>> >>>>>_________________________________________________________________________ >>>>>Professional FreeSWITCH Consulting Services: >>>>>consulting at freeswitch.org >>>>>http://www.freeswitchsolutions.com >>>>> >>>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> >>>>> >>>>>Official FreeSWITCH Sites >>>>>http://www.freeswitch.org >>>>>http://wiki.freeswitch.org >>>>>http://www.cluecon.com >>>>> >>>>>FreeSWITCH-users mailing list >>>>>FreeSWITCH-users at lists.freeswitch.org >>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting at freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> >>>> >>>>Official FreeSWITCH Sites >>>>http://www.freeswitch.org >>>>http://wiki.freeswitch.org >>>>http://www.cluecon.com >>>> >>>>FreeSWITCH-users mailing list >>>>FreeSWITCH-users at lists.freeswitch.org >>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> >>>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>> >>> >>> >>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>> >>> >>> >>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >> >> >> >> -- >>Andrew Cassidy BSc (Hons) MBCS SSCA >>Managing Director >> >> >> >> >>T?03300 100 960? F?03300 100 961 >>E?andrew at cassidywebservices.co.uk >>W?www.cassidywebservices.co.uk >> >> >>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director T?03300 100 960? F?03300 100 961 E?andrew at cassidywebservices.co.uk W?www.cassidywebservices.co.uk _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/3322a05a/attachment-0001.html From eidevm5 at gmail.com Mon Oct 21 03:58:42 2013 From: eidevm5 at gmail.com (Peter) Date: Mon, 21 Oct 2013 10:58:42 +1100 Subject: [Freeswitch-users] FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released! In-Reply-To: References: Message-ID: I hunted around for the release notes/changes to see what had been fixed, but couldn't find any. Closest I saw was http://wiki.freeswitch.org/wiki/Release_Notes but that hasn't been updated for a while. Is there an official place to view the release notes? On Fri, Oct 18, 2013 at 12:15 AM, Ken Rice wrote: > > FreeSWITCHers! > > Today we are proud to announce FreeSWITCH 1.2.14! > > Available today via git, > http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and > yum repos! > > This is a maintenance release to address several bugs that have been > identified since the last release. > > Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For > more information on how to join see: > http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/ca2101c4/attachment.html From krice at freeswitch.org Mon Oct 21 04:09:21 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 20 Oct 2013 19:09:21 -0500 Subject: [Freeswitch-users] FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released! In-Reply-To: Message-ID: Git log On 10/20/13 6:58 PM, "Peter" wrote: > I hunted around for the release notes/changes to see what had been fixed, but > couldn't find any. > > Closest I saw was http://wiki.freeswitch.org/wiki/Release_Notes but that > hasn't been updated for a while. > > Is there an official place to view the release notes? > > > On Fri, Oct 18, 2013 at 12:15 AM, Ken Rice wrote: >> >> FreeSWITCHers! >> >> Today we are proud to announce FreeSWITCH 1.2.14! >> >> Available today via git, >> http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum >> repos! >> >> This is a maintenance release to address several bugs that have been >> identified since the last release. >> ? >> Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For >> more information on how to join >> see:?http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructi >> ons >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://www.ss7.us/cluecon-gplus FB ClueCon : http://www.ss7.us/cluecon-fb G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus FB FreeSWITCH : http://www.ss7.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/d2ae8f49/attachment.html From krice at freeswitch.org Mon Oct 21 04:10:18 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 20 Oct 2013 19:10:18 -0500 Subject: [Freeswitch-users] FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released! In-Reply-To: Message-ID: Git log Or http://fisheye.freeswitch.org/browse/~br=v1.2.stable/freeswitch.git This are the official logs On 10/20/13 6:58 PM, "Peter" wrote: > I hunted around for the release notes/changes to see what had been fixed, but > couldn't find any. > > Closest I saw was http://wiki.freeswitch.org/wiki/Release_Notes but that > hasn't been updated for a while. > > Is there an official place to view the release notes? > > > On Fri, Oct 18, 2013 at 12:15 AM, Ken Rice wrote: >> >> FreeSWITCHers! >> >> Today we are proud to announce FreeSWITCH 1.2.14! >> >> Available today via git, >> http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum >> repos! >> >> This is a maintenance release to address several bugs that have been >> identified since the last release. >> ? >> Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For >> more information on how to join >> see:?http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructi >> ons >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://www.ss7.us/cluecon-gplus FB ClueCon : http://www.ss7.us/cluecon-fb G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus FB FreeSWITCH : http://www.ss7.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/2bef6a9e/attachment.html From yuri.ritvin at gmail.com Mon Oct 21 08:03:38 2013 From: yuri.ritvin at gmail.com (Yuri Ritvin) Date: Mon, 21 Oct 2013 00:03:38 -0400 Subject: [Freeswitch-users] Same timestamp 161 for all RFC2833 events generated by freeswitch Message-ID: In Freeswitch 1.2.12 all RFC2833 events generated by freeswitch bear the same timestamp (161). Is there a way to send incremental timestamp as it's sent for the rest of the RTP packets ? Thank you. Yuri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/02f9d40f/attachment.html From steveayre at gmail.com Mon Oct 21 13:30:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 21 Oct 2013 10:30:57 +0100 Subject: [Freeswitch-users] Same timestamp 161 for all RFC2833 events generated by freeswitch In-Reply-To: References: Message-ID: 1.2.12 is out of date. Do you see the same behaviour in newer versions (1.2.14), and if so do you also see it on master? If so file a Jira so that it can be corrected; if not then that upgrade will have solved your issue. On 21 October 2013 05:03, Yuri Ritvin wrote: > In Freeswitch 1.2.12 all RFC2833 events generated by freeswitch bear the > same timestamp (161). > Is there a way to send incremental timestamp as it's sent for the rest of > the RTP packets ? > > Thank you. > Yuri > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/c121af3c/attachment.html From yuri.ritvin at gmail.com Mon Oct 21 17:25:06 2013 From: yuri.ritvin at gmail.com (Yuri Ritvin) Date: Mon, 21 Oct 2013 09:25:06 -0400 Subject: [Freeswitch-users] Same timestamp 161 for all RFC2833 events generated by freeswitch In-Reply-To: References: Message-ID: It's the same with 1.2.14 - timestamp remains eq 161 for all DTMF packets generated by Freeswitch. I'll file Jira for this issue. Thanks. Yuri On Mon, Oct 21, 2013 at 5:30 AM, Steven Ayre wrote: > 1.2.12 is out of date. Do you see the same behaviour in newer versions > (1.2.14), and if so do you also see it on master? If so file a Jira so that > it can be corrected; if not then that upgrade will have solved your issue. > > > On 21 October 2013 05:03, Yuri Ritvin wrote: > >> In Freeswitch 1.2.12 all RFC2833 events generated by freeswitch bear the >> same timestamp (161). >> Is there a way to send incremental timestamp as it's sent for the rest of >> the RTP packets ? >> >> Thank you. >> Yuri >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/f398943a/attachment-0001.html From yuri.ritvin at gmail.com Mon Oct 21 17:40:48 2013 From: yuri.ritvin at gmail.com (Yuri Ritvin) Date: Mon, 21 Oct 2013 09:40:48 -0400 Subject: [Freeswitch-users] Same timestamp 161 for all RFC2833 events generated by freeswitch In-Reply-To: References: Message-ID: JIRA 5895 has been filed for this problem. On Mon, Oct 21, 2013 at 9:25 AM, Yuri Ritvin wrote: > It's the same with 1.2.14 - timestamp remains eq 161 for all DTMF packets > generated by Freeswitch. I'll file Jira for this issue. > > Thanks. > Yuri > > > On Mon, Oct 21, 2013 at 5:30 AM, Steven Ayre wrote: > >> 1.2.12 is out of date. Do you see the same behaviour in newer versions >> (1.2.14), and if so do you also see it on master? If so file a Jira so that >> it can be corrected; if not then that upgrade will have solved your issue. >> >> >> On 21 October 2013 05:03, Yuri Ritvin wrote: >> >>> In Freeswitch 1.2.12 all RFC2833 events generated by freeswitch bear the >>> same timestamp (161). >>> Is there a way to send incremental timestamp as it's sent for the rest >>> of the RTP packets ? >>> >>> Thank you. >>> Yuri >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/c83340fe/attachment.html From anthony.minessale at gmail.com Mon Oct 21 18:09:53 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Oct 2013 09:09:53 -0500 Subject: [Freeswitch-users] Same timestamp 161 for all RFC2833 events generated by freeswitch In-Reply-To: References: Message-ID: Did you attach the full debug log or provide details about exactly what you are doing to reproduce your problem. On Oct 21, 2013 9:43 AM, "Yuri Ritvin" wrote: > JIRA 5895 has been filed for this problem. > > > On Mon, Oct 21, 2013 at 9:25 AM, Yuri Ritvin wrote: > >> It's the same with 1.2.14 - timestamp remains eq 161 for all DTMF packets >> generated by Freeswitch. I'll file Jira for this issue. >> >> Thanks. >> Yuri >> >> >> On Mon, Oct 21, 2013 at 5:30 AM, Steven Ayre wrote: >> >>> 1.2.12 is out of date. Do you see the same behaviour in newer versions >>> (1.2.14), and if so do you also see it on master? If so file a Jira so that >>> it can be corrected; if not then that upgrade will have solved your issue. >>> >>> >>> On 21 October 2013 05:03, Yuri Ritvin wrote: >>> >>>> In Freeswitch 1.2.12 all RFC2833 events generated by freeswitch bear >>>> the same timestamp (161). >>>> Is there a way to send incremental timestamp as it's sent for the rest >>>> of the RTP packets ? >>>> >>>> Thank you. >>>> Yuri >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/5d0d75ca/attachment.html From lylepratt at gmail.com Mon Oct 21 01:30:18 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Sun, 20 Oct 2013 16:30:18 -0500 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? Message-ID: Hello! I'm trying to use the "group_confirm_file" functionality to prevent voicemail systems from answering my outbound calls to users. The problem I'm having is that there does not seem to be a way to prevent the "group_confirm_file" from looping/repeating. I only want the file to play ONE time then hang up if they do not press the "group_confirm_key". Currently, if someone ignores one of the calls (on their cell phone for example) and their voicemail system picks up, they will get a voicemail with the "group_confirm_file" audio. This is extremely annoying! How can I change this? Is it possible for me to write a module to change/correct this behavior? Thanks! Lyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131020/6311ec6d/attachment-0001.html From gggyniidt at gmail.com Mon Oct 21 14:44:50 2013 From: gggyniidt at gmail.com (=?GB2312?B?zfXTwMzO?=) Date: Mon, 21 Oct 2013 18:44:50 +0800 Subject: [Freeswitch-users] Can't receive CHANNEL_HANGUP_COMPLETE event when using mod_rtmp Message-ID: Hi I made some changes on testserver.c. Then I call a sipnumber from the flex client to a x-lite client. When I hang up from the flex side, can receive CHANNEL_HANGUP_COMPLETE event loop can be ended. When I hang up from the x-lite side, I can't receive CHANNEL_HANGUP_COMPLETE event and loop can not be ended, the variable status value is ESL_BREAK. Thanks. Code is as follows static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock, struct sockaddr_in *addr) { esl_handle_t handle = {{0}}; int done = 0; esl_status_t status; time_t exp = 0; esl_attach_handle(&handle, client_sock, addr); esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock); esl_filter(&handle, "unique-id", esl_event_get_header(handle.info_event, "caller-unique-id")); esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_HANGUP_COMPLETE DTMF CUSTOM"); esl_send_recv(&handle, "linger"); esl_execute(&handle, "set", "hangup_after_bridge=true", NULL); esl_execute(&handle, "bridge", "user/1001 at 192.168.1.128", NULL); int i=rand(); while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) { printf("Thread:%d\n", i); if (done) { if (time(NULL) >= exp) { break; } } else if (status == ESL_SUCCESS) { const char *type = esl_event_get_header(handle.last_event, "content-type"); if (type && !strcasecmp(type, "text/disconnect-notice")) { const char *dispo = esl_event_get_header(handle.last_event, "content-disposition"); esl_log(ESL_LOG_INFO, "Got a disconnection notice dispostion: [%s]\n", dispo ? dispo : ""); if (dispo && !strcmp(dispo, "linger")) { done = 1; esl_log(ESL_LOG_INFO, "Waiting 5 seconds for any remaining events.\n"); exp = time(NULL) + 5; } } } } esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock); esl_disconnect(&handle); } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/1a9d4b80/attachment-0001.html From lylepratt at gmail.com Mon Oct 21 18:11:49 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Mon, 21 Oct 2013 09:11:49 -0500 Subject: [Freeswitch-users] How do I stop group_confirm_file from looping? Message-ID: I'm trying to use the "group_confirm_file" functionality to prevent voicemail systems from answering my outbound bridge calls to users. The problem I'm having is that there does not seem to be a way to prevent the "group_confirm_file" from looping/repeating. I only want the file to play ONE time then hang up if they do not press the "group_confirm_key". Currently, if someone ignores one of the calls (on their cell phone for example) and their voicemail system picks up, they will get a voicemail with the "group_confirm_file" audio. This is extremely annoying! How can I change this? Is it possible for me to write a module to change/correct this behavior? Thanks! Lyle Pratt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/cf0ce131/attachment.html From regis.freeswitch.org at tornad.net Mon Oct 21 20:55:10 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Mon, 21 Oct 2013 18:55:10 +0200 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: Hi, Try to look at that : http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout Regards 2013/10/20 Lyle Pratt > Hello! I'm trying to use the "group_confirm_file" functionality to prevent > voicemail systems from answering my outbound calls to users. > > The problem I'm having is that there does not seem to be a way to prevent > the "group_confirm_file" from looping/repeating. I only want the file to > play ONE time then hang up if they do not press the "group_confirm_key". > Currently, if someone ignores one of the calls (on their cell phone for > example) and their voicemail system picks up, they will get a voicemail > with the "group_confirm_file" audio. This is extremely annoying! > > How can I change this? Is it possible for me to write a module to > change/correct this behavior? > > Thanks! > Lyle > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/37f6f622/attachment.html From lylepratt at gmail.com Mon Oct 21 21:04:41 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Mon, 21 Oct 2013 12:04:41 -0500 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: Hi Regis, Thanks for the tip, but unfortunately that does not help. I need to prevent the group_confirm_file from looping at all. It doesn't have anything to do with the leg timeout. In other words, my leg timeout could be 10 seconds or 100 seconds, but the group_confirm_file should still only play one time, wait for input, then hangup. -Lyle On Mon, Oct 21, 2013 at 11:55 AM, Regis M wrote: > Hi, > > Try to look at that : > http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout > > Regards > > > 2013/10/20 Lyle Pratt > >> Hello! I'm trying to use the "group_confirm_file" functionality to >> prevent voicemail systems from answering my outbound calls to users. >> >> The problem I'm having is that there does not seem to be a way to prevent >> the "group_confirm_file" from looping/repeating. I only want the file to >> play ONE time then hang up if they do not press the "group_confirm_key". >> Currently, if someone ignores one of the calls (on their cell phone for >> example) and their voicemail system picks up, they will get a voicemail >> with the "group_confirm_file" audio. This is extremely annoying! >> >> How can I change this? Is it possible for me to write a module to >> change/correct this behavior? >> >> Thanks! >> Lyle >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/a9d4bdd8/attachment.html From regis.freeswitch.org at tornad.net Mon Oct 21 21:20:37 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Mon, 21 Oct 2013 19:20:37 +0200 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: There's also some other undocumented variables : Look here : http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Channel_variables group_confirm_read_timeout will do the job maybe Anyway, you have to fix the timeout at the audio lenght of your group_confirm_file.. 2013/10/21 Lyle Pratt > Hi Regis, > > Thanks for the tip, but unfortunately that does not help. I need to > prevent the group_confirm_file from looping at all. It doesn't have > anything to do with the leg timeout. In other words, my leg timeout could > be 10 seconds or 100 seconds, but the group_confirm_file should still only > play one time, wait for input, then hangup. > > -Lyle > > > On Mon, Oct 21, 2013 at 11:55 AM, Regis M > wrote: > >> Hi, >> >> Try to look at that : >> http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout >> >> Regards >> >> >> 2013/10/20 Lyle Pratt >> >>> Hello! I'm trying to use the "group_confirm_file" functionality to >>> prevent voicemail systems from answering my outbound calls to users. >>> >>> The problem I'm having is that there does not seem to be a way to >>> prevent the "group_confirm_file" from looping/repeating. I only want the >>> file to play ONE time then hang up if they do not press the >>> "group_confirm_key". Currently, if someone ignores one of the calls (on >>> their cell phone for example) and their voicemail system picks up, they >>> will get a voicemail with the "group_confirm_file" audio. This is extremely >>> annoying! >>> >>> How can I change this? Is it possible for me to write a module to >>> change/correct this behavior? >>> >>> Thanks! >>> Lyle >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/e2c8063c/attachment-0001.html From lylepratt at gmail.com Mon Oct 21 21:27:48 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Mon, 21 Oct 2013 12:27:48 -0500 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: Hi Regis, Thanks for sticking with me on this issue. I've tried group_confirm_read_timeout, but its not what I'm looking for. I've also taken a look at the source for switch_ivr_read which is what the "group_confirm" functionality is using, and it doesn't seem to support a max retries, or hard timeout to wait for a "confirmation". I think that ( http://docs.freeswitch.org/group__switch__ivr__menu.html#g372a59da20a90e23f4fee8e36b829fee ) I'm not sure what you mean by "fix the timeout at the audio lenght of your group_confirm_file". Can you give me a few more details? Thanks! -Lyle On Mon, Oct 21, 2013 at 12:20 PM, Regis M wrote: > There's also some other undocumented variables : > Look here : > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Channel_variables > > group_confirm_read_timeout will do the job maybe > > Anyway, you have to fix the timeout at the audio lenght of your > group_confirm_file.. > > > > 2013/10/21 Lyle Pratt > >> Hi Regis, >> >> Thanks for the tip, but unfortunately that does not help. I need to >> prevent the group_confirm_file from looping at all. It doesn't have >> anything to do with the leg timeout. In other words, my leg timeout could >> be 10 seconds or 100 seconds, but the group_confirm_file should still only >> play one time, wait for input, then hangup. >> >> -Lyle >> >> >> On Mon, Oct 21, 2013 at 11:55 AM, Regis M < >> regis.freeswitch.org at tornad.net> wrote: >> >>> Hi, >>> >>> Try to look at that : >>> http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout >>> >>> Regards >>> >>> >>> 2013/10/20 Lyle Pratt >>> >>>> Hello! I'm trying to use the "group_confirm_file" functionality to >>>> prevent voicemail systems from answering my outbound calls to users. >>>> >>>> The problem I'm having is that there does not seem to be a way to >>>> prevent the "group_confirm_file" from looping/repeating. I only want the >>>> file to play ONE time then hang up if they do not press the >>>> "group_confirm_key". Currently, if someone ignores one of the calls (on >>>> their cell phone for example) and their voicemail system picks up, they >>>> will get a voicemail with the "group_confirm_file" audio. This is extremely >>>> annoying! >>>> >>>> How can I change this? Is it possible for me to write a module to >>>> change/correct this behavior? >>>> >>>> Thanks! >>>> Lyle >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/48cd9972/attachment.html From lylepratt at gmail.com Mon Oct 21 21:44:54 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Mon, 21 Oct 2013 12:44:54 -0500 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: Additional information: I thought that "group_confirm_read_timeout" would be what I needed as well, but I've tried it and it does not seem to have any affect on how long it waits for confirmation. Maybe it doesn't work correctly when using Enterprise Originate? -Lyle On Mon, Oct 21, 2013 at 12:27 PM, Lyle Pratt wrote: > Hi Regis, > > Thanks for sticking with me on this issue. I've tried group_confirm_read_timeout, > but its not what I'm looking for. I've also taken a look at the source for > switch_ivr_read which is what the "group_confirm" functionality is using, > and it doesn't seem to support a max retries, or hard timeout to wait for a > "confirmation". I think that ( > http://docs.freeswitch.org/group__switch__ivr__menu.html#g372a59da20a90e23f4fee8e36b829fee > ) > > I'm not sure what you mean by "fix the timeout at the audio lenght of > your group_confirm_file". Can you give me a few more details? > > Thanks! > -Lyle > > > > > On Mon, Oct 21, 2013 at 12:20 PM, Regis M > wrote: > >> There's also some other undocumented variables : >> Look here : >> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Channel_variables >> >> group_confirm_read_timeout will do the job maybe >> >> Anyway, you have to fix the timeout at the audio lenght of your >> group_confirm_file.. >> >> >> >> 2013/10/21 Lyle Pratt >> >>> Hi Regis, >>> >>> Thanks for the tip, but unfortunately that does not help. I need to >>> prevent the group_confirm_file from looping at all. It doesn't have >>> anything to do with the leg timeout. In other words, my leg timeout could >>> be 10 seconds or 100 seconds, but the group_confirm_file should still only >>> play one time, wait for input, then hangup. >>> >>> -Lyle >>> >>> >>> On Mon, Oct 21, 2013 at 11:55 AM, Regis M < >>> regis.freeswitch.org at tornad.net> wrote: >>> >>>> Hi, >>>> >>>> Try to look at that : >>>> http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout >>>> >>>> Regards >>>> >>>> >>>> 2013/10/20 Lyle Pratt >>>> >>>>> Hello! I'm trying to use the "group_confirm_file" functionality to >>>>> prevent voicemail systems from answering my outbound calls to users. >>>>> >>>>> The problem I'm having is that there does not seem to be a way to >>>>> prevent the "group_confirm_file" from looping/repeating. I only want the >>>>> file to play ONE time then hang up if they do not press the >>>>> "group_confirm_key". Currently, if someone ignores one of the calls (on >>>>> their cell phone for example) and their voicemail system picks up, they >>>>> will get a voicemail with the "group_confirm_file" audio. This is extremely >>>>> annoying! >>>>> >>>>> How can I change this? Is it possible for me to write a module to >>>>> change/correct this behavior? >>>>> >>>>> Thanks! >>>>> Lyle >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/3223b2e4/attachment-0001.html From anthony.minessale at gmail.com Mon Oct 21 21:45:17 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Oct 2013 12:45:17 -0500 Subject: [Freeswitch-users] How do I stop group_confirm_file from looping? In-Reply-To: References: Message-ID: Use the exec app mode and call a lua script that plays once, waits for input once and hangs up. On Oct 21, 2013 11:19 AM, "Lyle Pratt" wrote: > I'm trying to use the "group_confirm_file" functionality to prevent > voicemail systems from answering my outbound bridge calls to users. > > The problem I'm having is that there does not seem to be a way to prevent > the "group_confirm_file" from looping/repeating. I only want the file to > play ONE time then hang up if they do not press the "group_confirm_key". > Currently, if someone ignores one of the calls (on their cell phone for > example) and their voicemail system picks up, they will get a voicemail > with the "group_confirm_file" audio. This is extremely annoying! > > How can I change this? Is it possible for me to write a module to > change/correct this behavior? > > Thanks! > Lyle Pratt > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/85bd62bb/attachment.html From lylepratt at gmail.com Mon Oct 21 21:51:48 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Mon, 21 Oct 2013 12:51:48 -0500 Subject: [Freeswitch-users] How do I stop group_confirm_file from looping? In-Reply-To: References: Message-ID: Thanks Anthony. I will try to figure that out and report back in original thread. My sincere apologies to the group, but It looks like I double posted this topic. I posted yesterday but did not receive a confirmation and did not know that it went through. I will only reply to the original through now. -Lyle On Mon, Oct 21, 2013 at 12:45 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Use the exec app mode and call a lua script that plays once, waits for > input once and hangs up. > On Oct 21, 2013 11:19 AM, "Lyle Pratt" wrote: > >> I'm trying to use the "group_confirm_file" functionality to prevent >> voicemail systems from answering my outbound bridge calls to users. >> >> The problem I'm having is that there does not seem to be a way to prevent >> the "group_confirm_file" from looping/repeating. I only want the file to >> play ONE time then hang up if they do not press the "group_confirm_key". >> Currently, if someone ignores one of the calls (on their cell phone for >> example) and their voicemail system picks up, they will get a voicemail >> with the "group_confirm_file" audio. This is extremely annoying! >> >> How can I change this? Is it possible for me to write a module to >> change/correct this behavior? >> >> Thanks! >> Lyle Pratt >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/556d87c6/attachment.html From lylepratt at gmail.com Mon Oct 21 22:25:28 2013 From: lylepratt at gmail.com (Lyle Pratt) Date: Mon, 21 Oct 2013 13:25:28 -0500 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: Anthony suggested that I: "Use the exec app mode and call a lua script that plays once, waits for input once and hangs up." I guess I can do that as a last resort, however based on the documentation, it seems that "group_confirm_read_timeout" should work, however it does not have any affect for me. Can someone else try this or does anyone know why it is not working? -Lyle On Mon, Oct 21, 2013 at 12:44 PM, Lyle Pratt wrote: > Additional information: > > I thought that "group_confirm_read_timeout" would be what I needed as > well, but I've tried it and it does not seem to have any affect on how long > it waits for confirmation. Maybe it doesn't work correctly when using > Enterprise Originate? > > -Lyle > > > On Mon, Oct 21, 2013 at 12:27 PM, Lyle Pratt wrote: > >> Hi Regis, >> >> Thanks for sticking with me on this issue. I've tried group_confirm_read_timeout, >> but its not what I'm looking for. I've also taken a look at the source for >> switch_ivr_read which is what the "group_confirm" functionality is using, >> and it doesn't seem to support a max retries, or hard timeout to wait for a >> "confirmation". I think that ( >> http://docs.freeswitch.org/group__switch__ivr__menu.html#g372a59da20a90e23f4fee8e36b829fee >> ) >> >> I'm not sure what you mean by "fix the timeout at the audio lenght of >> your group_confirm_file". Can you give me a few more details? >> >> Thanks! >> -Lyle >> >> >> >> >> On Mon, Oct 21, 2013 at 12:20 PM, Regis M < >> regis.freeswitch.org at tornad.net> wrote: >> >>> There's also some other undocumented variables : >>> Look here : >>> >>> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Channel_variables >>> >>> group_confirm_read_timeout will do the job maybe >>> >>> Anyway, you have to fix the timeout at the audio lenght of your >>> group_confirm_file.. >>> >>> >>> >>> 2013/10/21 Lyle Pratt >>> >>>> Hi Regis, >>>> >>>> Thanks for the tip, but unfortunately that does not help. I need to >>>> prevent the group_confirm_file from looping at all. It doesn't have >>>> anything to do with the leg timeout. In other words, my leg timeout could >>>> be 10 seconds or 100 seconds, but the group_confirm_file should still only >>>> play one time, wait for input, then hangup. >>>> >>>> -Lyle >>>> >>>> >>>> On Mon, Oct 21, 2013 at 11:55 AM, Regis M < >>>> regis.freeswitch.org at tornad.net> wrote: >>>> >>>>> Hi, >>>>> >>>>> Try to look at that : >>>>> http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout >>>>> >>>>> Regards >>>>> >>>>> >>>>> 2013/10/20 Lyle Pratt >>>>> >>>>>> Hello! I'm trying to use the "group_confirm_file" functionality to >>>>>> prevent voicemail systems from answering my outbound calls to users. >>>>>> >>>>>> The problem I'm having is that there does not seem to be a way to >>>>>> prevent the "group_confirm_file" from looping/repeating. I only want the >>>>>> file to play ONE time then hang up if they do not press the >>>>>> "group_confirm_key". Currently, if someone ignores one of the calls (on >>>>>> their cell phone for example) and their voicemail system picks up, they >>>>>> will get a voicemail with the "group_confirm_file" audio. This is extremely >>>>>> annoying! >>>>>> >>>>>> How can I change this? Is it possible for me to write a module to >>>>>> change/correct this behavior? >>>>>> >>>>>> Thanks! >>>>>> Lyle >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/d801d7dc/attachment-0001.html From regis.freeswitch.org at tornad.net Mon Oct 21 23:26:16 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Mon, 21 Oct 2013 21:26:16 +0200 Subject: [Freeswitch-users] How to prevent group_confirm_file from looping? In-Reply-To: References: Message-ID: Hi, beware... long post :) In my memory, when I used group_confirm, I noticed that the read timeout start AFTER the file prompt was played, not during or at the answer starts... In fact, I think you have "Sound prompt lenght" time +"read timeout" time to press a key or the confirm will be canceled/rejected... So you're could also try with group_confirm_read_timeout = 0 or 100. >I'm not sure what you mean by "fix the timeout at the audio lenght of your group_confirm_file". Can you give me a few more details? I mean that when I tried to do that, I had to mix with all the timeout to get my wanted features... This means : On this string : originate [leg_timeout=TIMEOUT1,group_confirm_read_timeout=TIMEOUT2...]sofia/endpoint &park() - with TIMEOUT1= 10000, you will have (ring allowed time) + (sound prompt_lenght) + (TIMEOUT2 ) <- will result with hangup at TIMEOUT1 but could hangup during prompt if it takes time to answer (various ring time) - with group_confirm_cancel_timeout=1 .... The ring time is not care,, and the leg_timeout if canceled on legB answer, so you can choose : timeout for ring (leg_timeout with less value) and timeout for the read of digit.(but this timeout start after prompt), so if you just want your prompt, you have to put read_timeout to 0 One thing which is strange for me is that you said the sound loops... for me, the sound doesn't loop if no confirm is done within the timeout... Anyway, you must have a look at the LUA script which is a better solution if you want to manage too much complex behavior in dialplan.. Hope it helps a little and sorry for my poor english... Regards, 2013/10/21 Lyle Pratt > Anthony suggested that I: "Use the exec app mode and call a lua script > that plays once, waits for input once and hangs up." > > I guess I can do that as a last resort, however based on the > documentation, it seems that "group_confirm_read_timeout" should work, > however it does not have any affect for me. Can someone else try this or > does anyone know why it is not working? > > -Lyle > > > On Mon, Oct 21, 2013 at 12:44 PM, Lyle Pratt wrote: > >> Additional information: >> >> I thought that "group_confirm_read_timeout" would be what I needed as >> well, but I've tried it and it does not seem to have any affect on how long >> it waits for confirmation. Maybe it doesn't work correctly when using >> Enterprise Originate? >> >> -Lyle >> >> >> On Mon, Oct 21, 2013 at 12:27 PM, Lyle Pratt wrote: >> >>> Hi Regis, >>> >>> Thanks for sticking with me on this issue. I've tried group_confirm_read_timeout, >>> but its not what I'm looking for. I've also taken a look at the source for >>> switch_ivr_read which is what the "group_confirm" functionality is using, >>> and it doesn't seem to support a max retries, or hard timeout to wait for a >>> "confirmation". I think that ( >>> http://docs.freeswitch.org/group__switch__ivr__menu.html#g372a59da20a90e23f4fee8e36b829fee >>> ) >>> >>> I'm not sure what you mean by "fix the timeout at the audio lenght of >>> your group_confirm_file". Can you give me a few more details? >>> >>> Thanks! >>> -Lyle >>> >>> >>> >>> >>> On Mon, Oct 21, 2013 at 12:20 PM, Regis M < >>> regis.freeswitch.org at tornad.net> wrote: >>> >>>> There's also some other undocumented variables : >>>> Look here : >>>> >>>> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Channel_variables >>>> >>>> group_confirm_read_timeout will do the job maybe >>>> >>>> Anyway, you have to fix the timeout at the audio lenght of your >>>> group_confirm_file.. >>>> >>>> >>>> >>>> 2013/10/21 Lyle Pratt >>>> >>>>> Hi Regis, >>>>> >>>>> Thanks for the tip, but unfortunately that does not help. I need to >>>>> prevent the group_confirm_file from looping at all. It doesn't have >>>>> anything to do with the leg timeout. In other words, my leg timeout could >>>>> be 10 seconds or 100 seconds, but the group_confirm_file should still only >>>>> play one time, wait for input, then hangup. >>>>> >>>>> -Lyle >>>>> >>>>> >>>>> On Mon, Oct 21, 2013 at 11:55 AM, Regis M < >>>>> regis.freeswitch.org at tornad.net> wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Try to look at that : >>>>>> http://wiki.freeswitch.org/wiki/Variable_group_confirm_cancel_timeout >>>>>> >>>>>> Regards >>>>>> >>>>>> >>>>>> 2013/10/20 Lyle Pratt >>>>>> >>>>>>> Hello! I'm trying to use the "group_confirm_file" functionality to >>>>>>> prevent voicemail systems from answering my outbound calls to users. >>>>>>> >>>>>>> The problem I'm having is that there does not seem to be a way to >>>>>>> prevent the "group_confirm_file" from looping/repeating. I only want the >>>>>>> file to play ONE time then hang up if they do not press the >>>>>>> "group_confirm_key". Currently, if someone ignores one of the calls (on >>>>>>> their cell phone for example) and their voicemail system picks up, they >>>>>>> will get a voicemail with the "group_confirm_file" audio. This is extremely >>>>>>> annoying! >>>>>>> >>>>>>> How can I change this? Is it possible for me to write a module to >>>>>>> change/correct this behavior? >>>>>>> >>>>>>> Thanks! >>>>>>> Lyle >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/bcefb083/attachment-0001.html From ben at langfeld.co.uk Mon Oct 21 23:33:35 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Mon, 21 Oct 2013 17:33:35 -0200 Subject: [Freeswitch-users] AdhearsionConf 2013 Message-ID: Apologies for interrupting Ladies and Gents, but if you would spare a moment of your time for a brief suggestion... Old School Telephony meets the New School Open Web: Adhearsion ( http://adhearsion.com) is one of the premiere choices of framework for building rich Voice and Real-Time Communications applications. The project is six years old and has a history of successful deployments of very powerful and sophisticated voice integrations. At AdhearsionConf (Dec 4-5, Atlanta GA, http://adhearsionconf.com) you can hear from experts in the space and learn how to integrate new tools like WebRTC, as well as advancements in hot topics like speech recognition and text-to-speech, and how to use Adhearsion in conjunction with FreeSWITCH to vastly expand the capabilities of your network both internally and shared with third-party developers to create applications which leave a lasting impression on your users. Tickets for the conference are available at http://adhearsionconf.com/tickets at the early-bird rate of $110 until Nov 11th. The FreeSWITCH community's very own Chris Rienzo will be there to talk about mod_rayo, along with a flock of other excellent speakers including Tim Wenhold (PHRG) and Matt Jordan (Digium). I'd personally love to see FreeSWITCH people making a strong attendance and showing Matt what for! The first round is also on me! Ben Langfeld Adhearsion Foundation Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/7d112706/attachment.html From alipey at gmail.com Tue Oct 22 01:21:23 2013 From: alipey at gmail.com (Ali Pey) Date: Mon, 21 Oct 2013 17:21:23 -0400 Subject: [Freeswitch-users] Voicemail notification Message-ID: Hello, I need to send a customized voicemail notification based on certain condition once a voicemail is recorded. 1- How can I retrieve the voicemail filename? 2- Is api_hangup_hook a proper way to detect if a voicemail was left? 3- How can I assign custom variables to the channel so at api_hangup_hook I know what I needed to do? Please help. Thanks, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/4c354e2c/attachment.html From msc at freeswitch.org Tue Oct 22 02:21:56 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Oct 2013 15:21:56 -0700 Subject: [Freeswitch-users] newbie: bridge control In-Reply-To: References: Message-ID: On Sun, Oct 20, 2013 at 8:50 AM, ik wrote: > Hello List, > > I'm a newbie to Freeswitch. > Welcome! > > I need to bridge an inbound call to another call, and detect pattren of > DTMF. > If the the pattern happen, I need to capture it to a variable, and hangup > leg B (the bridged one), and continue doing stuff with Leg A. > Does the B leg already exist? If so, you'll need something like uuid_bridge API. If it does not already exist and you need to create it then use the bridge dialplan app. For matching a pattern your best bet is the bind_digit_action dialplan app. For "doing stuff" on the A you'll need to be more specific. Do you need to interact with the caller? If so, use the transfer dialplan application to send the call elsewhere in the dialplan to do more "stuff." If by "stuff" you mean run an external script then checkout the api_hangup_hook variable. > > If Leg B was disconnected, I need hangup leg A as well. > This happens naturally unless you have the chan var hangup_after_bridge set to false. > > I wish to better understand how to look such thing on the wiki, prior for > direct answer on how to do it, in order to better learn FS and it's usage :) > I like to use the navigation buttons on the left side. You can look up dialplan applications ("apps"), FreeSWITCH API commands, ("APIs"), or channel variables, ("chan vars"). Don't forget to check out the IRC channel as well: #freeswitch on irc.freenode.net. > > So any help on this matter is more then welcome. > > Thanks > Ido > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131021/403adff9/attachment.html From dujinfang at gmail.com Tue Oct 22 07:43:34 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 22 Oct 2013 11:43:34 +0800 Subject: [Freeswitch-users] Can't receive CHANNEL_HANGUP_COMPLETE event when using mod_rtmp In-Reply-To: References: Message-ID: <321D373A98EA497B817ABBE8BA36A471@gmail.com> chances are the calling side didn't hangup. what "show channels" tell you? What's your dialplan look like? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, October 21, 2013 at 6:44 PM, ??? wrote: > Hi > I made some changes on testserver.c. Then I call a sipnumber from the flex client to a x-lite client. When I hang up from the flex side, can receive CHANNEL_HANGUP_COMPLETE event loop can be ended. When I hang up from the x-lite side, I can't receive CHANNEL_HANGUP_COMPLETE event and loop can not be ended, the variable status value is ESL_BREAK. > > Thanks. > > Code is as follows > > static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock, > struct sockaddr_in *addr) { > esl_handle_t handle = {{0}}; > int done = 0; > esl_status_t status; > time_t exp = 0; > > esl_attach_handle(&handle, client_sock, addr); > > esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock); > > esl_filter(&handle, "unique-id", esl_event_get_header(handle.info_event, "caller-unique-id")); > esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_HANGUP_COMPLETE DTMF CUSTOM"); > > esl_send_recv(&handle, "linger"); > > > esl_execute(&handle, "set", "hangup_after_bridge=true", NULL); > esl_execute(&handle, "bridge", "user/1001 at 192.168.1.128 (mailto:1001 at 192.168.1.128)", NULL); > > int i=rand(); > while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) { > printf("Thread:%d\n", i); > if (done) { > if (time(NULL) >= exp) { > break; > } > } else if (status == ESL_SUCCESS) { > const char *type = esl_event_get_header(handle.last_event, "content-type"); > if (type && !strcasecmp(type, "text/disconnect-notice")) { > const char *dispo = esl_event_get_header(handle.last_event, "content-disposition"); > esl_log(ESL_LOG_INFO, "Got a disconnection notice dispostion: [%s]\n", dispo ? dispo : ""); > if (dispo && !strcmp(dispo, "linger")) { > done = 1; > esl_log(ESL_LOG_INFO, "Waiting 5 seconds for any remaining events.\n"); > exp = time(NULL) + 5; > } > } > } > } > > esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock); > esl_disconnect(&handle); > } > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/d54ca834/attachment-0001.html From gggyniidt at gmail.com Tue Oct 22 08:02:54 2013 From: gggyniidt at gmail.com (=?GB2312?B?zfXTwMzO?=) Date: Tue, 22 Oct 2013 12:02:54 +0800 Subject: [Freeswitch-users] Can't receive CHANNEL_HANGUP_COMPLETE event when using mod_rtmp In-Reply-To: <321D373A98EA497B817ABBE8BA36A471@gmail.com> References: <321D373A98EA497B817ABBE8BA36A471@gmail.com> Message-ID: freeswitch at wtosdev> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 56eb412a-3ace-11e3-b62f-956876d28f28,inbound,2013-10-22 11:59:24,1382414364,sofia/internal/1000 at 192.168.1.128 ,CS_EXECUTE,1000,1000,192.168.1.6,1001,bridge,user/1001 at 192.168.1.128 ,XML,default,PCMU,8000,64000,PCMU,8000,64000,,wtosdev,1000 at 192.168.1.128,,ACTIVE,Outbound Call,1001,SEND,56eb412a-3ace-11e3-b62f-956876d28f28,Outbound Call,1001 freeswitch at wtosdev> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 56eb412a-3ace-11e3-b62f-956876d28f28,inbound,2013-10-22 11:59:24,1382414364,sofia/internal/1000 at 192.168.1.128 ,CS_EXECUTE,1000,1000,192.168.1.6,1001,1000 at 192.168.1.128,,ACTIVE,Outbound Call,1001,SEND,56eb412a-3ace-11e3-b62f-956876d28f28,wtosdev,Outbound Call,1001,,,,,,,,,,,,,,,,,,, My Dialplan Like This 2013/10/22 Seven Du > chances are the calling side didn't hangup. what "show channels" tell > you? What's your dialplan look like? > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Monday, October 21, 2013 at 6:44 PM, ??? wrote: > > Hi > I made some changes on testserver.c. Then I call a sipnumber from the flex > client to a x-lite client. When I hang up from the flex side, can receive > CHANNEL_HANGUP_COMPLETE event loop can be ended. When I hang up from the > x-lite side, I can't receive CHANNEL_HANGUP_COMPLETE event and loop can > not be ended, the variable status value is ESL_BREAK. > > Thanks. > > Code is as follows > > static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock, > struct sockaddr_in *addr) { > esl_handle_t handle = {{0}}; > int done = 0; > esl_status_t status; > time_t exp = 0; > > esl_attach_handle(&handle, client_sock, addr); > > esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock); > > esl_filter(&handle, "unique-id", esl_event_get_header(handle.info_event, > "caller-unique-id")); > esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_HANGUP_COMPLETE DTMF > CUSTOM"); > > esl_send_recv(&handle, "linger"); > > > esl_execute(&handle, "set", "hangup_after_bridge=true", NULL); > esl_execute(&handle, "bridge", "user/1001 at 192.168.1.128", NULL); > > int i=rand(); > while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) { > printf("Thread:%d\n", i); > if (done) { > if (time(NULL) >= exp) { > break; > } > } else if (status == ESL_SUCCESS) { > const char *type = esl_event_get_header(handle.last_event, "content-type"); > if (type && !strcasecmp(type, "text/disconnect-notice")) { > const char *dispo = esl_event_get_header(handle.last_event, > "content-disposition"); > esl_log(ESL_LOG_INFO, "Got a disconnection notice dispostion: [%s]\n", > dispo ? dispo : ""); > if (dispo && !strcmp(dispo, "linger")) { > done = 1; > esl_log(ESL_LOG_INFO, "Waiting 5 seconds for any remaining events.\n"); > exp = time(NULL) + 5; > } > } > } > } > > esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock); > esl_disconnect(&handle); > } > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/bcdb66cc/attachment.html From hcoin at quietfountain.com Tue Oct 22 10:01:58 2013 From: hcoin at quietfountain.com (hcoin) Date: Tue, 22 Oct 2013 01:01:58 -0500 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling Message-ID: <526614D6.3020401@quietfountain.com> This has been a really frustrating problem, I'm sure the answer is simple but I just can't see it. I had several extensions registered to the internal profile, sending calls out the external profile to a sip-pstn gateway, all seemed fine. Then created another internal profile, using a different sip port on the same lan address, because of 'no device left behind' and NAT issues. All seemed well, all the phones register normally. Looking at the databases in FS they all show the proper ports, the proper domains, etc. However, every single call gets picked up as a new call via sophia/external/... and it hits the public dialplan normally -- except that's the wrong plan, it should hit the default plan and be identified as sofia/internal/.... and so forth. 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel sofia/external/hcoin at pbx.foobar.com [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] What could possibly be the reason phones registered on the internal profile have their new calls identified as sophia/external and don't hit the correct plan? Both the phones and the freeswitch are on the same subnet. This should be so vanilla. What am I missing? From idokan at gmail.com Tue Oct 22 11:26:08 2013 From: idokan at gmail.com (ik) Date: Tue, 22 Oct 2013 09:26:08 +0200 Subject: [Freeswitch-users] newbie: bridge control In-Reply-To: References: Message-ID: Thanks Michael, I'ved created the following dialplan (fs v1.2.14) : bind_digit_action constantly reporting Syntax Error, USAGE ,,[,][,][,] The original aim of the program is the capture the numbers after 200*, and bridge leg-a to the new number, but first disconnecting current leg-b. When using execute_extension with "exec_after_bridge_app=execute_extension "exec_after_bridge_arg=digits ${bridging_capture}" It makes leg-b constantly ring after the call hangup. So does the transfer action that is commented out. So what am I doing wrong here ? Thanks, Ido On Tue, Oct 22, 2013 at 1:21 AM, Michael Collins wrote: > > > > On Sun, Oct 20, 2013 at 8:50 AM, ik wrote: > >> Hello List, >> >> I'm a newbie to Freeswitch. >> > Welcome! > > >> >> I need to bridge an inbound call to another call, and detect pattren of >> DTMF. >> If the the pattern happen, I need to capture it to a variable, and hangup >> leg B (the bridged one), and continue doing stuff with Leg A. >> > Does the B leg already exist? If so, you'll need something like > uuid_bridge API. If it does not already exist and you need to create it > then use the bridge dialplan app. > > For matching a pattern your best bet is the bind_digit_action dialplan app. > > For "doing stuff" on the A you'll need to be more specific. Do you need to > interact with the caller? If so, use the transfer dialplan application to > send the call elsewhere in the dialplan to do more "stuff." If by "stuff" > you mean run an external script then checkout the api_hangup_hook variable. > > >> >> If Leg B was disconnected, I need hangup leg A as well. >> > This happens naturally unless you have the chan var hangup_after_bridge > set to false. > > >> >> I wish to better understand how to look such thing on the wiki, prior for >> direct answer on how to do it, in order to better learn FS and it's usage :) >> > I like to use the navigation buttons on the left side. You can look up > dialplan applications ("apps"), FreeSWITCH API commands, ("APIs"), or > channel variables, ("chan vars"). Don't forget to check out the IRC channel > as well: #freeswitch on irc.freenode.net. > >> >> So any help on this matter is more then welcome. >> >> Thanks >> Ido >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/6820214e/attachment-0001.html From GB at cm.nl Tue Oct 22 12:14:31 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 22 Oct 2013 10:14:31 +0200 Subject: [Freeswitch-users] Audio quality issues Message-ID: Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue's. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren't doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn't echo the RTP back. These numbers get lower once Sipp echo's the RTP. I've tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn't that much. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/9ff282fb/attachment.html From ssinyagin at yahoo.com Tue Oct 22 13:10:49 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 22 Oct 2013 02:10:49 -0700 (PDT) Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: Message-ID: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load?? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, ? I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue?s. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren?t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn?t echo the RTP back. These numbers get lower once Sipp echo?s the RTP. ? I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn?t that much. ? I disabled all the modules we don?t need, like CDR?s, conferencing, etc. ? Are there any parameters(config files)/modules that can affect the quality of the audio stream? ? Regards, ? Grant ? ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/ba88ab8d/attachment.html From GB at cm.nl Tue Oct 22 14:01:29 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 22 Oct 2013 12:01:29 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: The network shouldn?t be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn?t that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian > To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" > Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue?s. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren?t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn?t echo the RTP back. These numbers get lower once Sipp echo?s the RTP. I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn?t that much. I disabled all the modules we don?t need, like CDR?s, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/c6e1f659/attachment-0001.html From xmppser at gmail.com Tue Oct 22 14:05:46 2013 From: xmppser at gmail.com (xmppser) Date: Tue, 22 Oct 2013 03:05:46 -0700 (PDT) Subject: [Freeswitch-users] FreeTDM tones.conf Disconnect Supervision In-Reply-To: References: Message-ID: <1382436346467-7595879.post@n2.nabble.com> Hi? I have this issue also, I use sangoma A400, use sip phone call pstn mobile, when mobile hangup, the sip phone can not hangup, i have add tone_detect app in dialplan, it seems sangoma A400 and freetdm can not detect busy tones or disconct srperversion from pstm. but asterisk & zaptel have this detect. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeTDM-tones-conf-Disconnect-Supervision-tp7595763p7595879.html Sent from the freeswitch-users mailing list archive at Nabble.com. From hardyanto.donny at gmail.com Tue Oct 22 15:13:46 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Tue, 22 Oct 2013 18:13:46 +0700 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling In-Reply-To: <526614D6.3020401@quietfountain.com> References: <526614D6.3020401@quietfountain.com> Message-ID: Is your client in the internet or the lan? Donny On Tue, Oct 22, 2013 at 1:01 PM, hcoin wrote: > > This has been a really frustrating problem, I'm sure the answer is > simple but I just can't see it. > > I had several extensions registered to the internal profile, sending > calls out the external profile to a sip-pstn gateway, all seemed fine. > > Then created another internal profile, using a different sip port on the > same lan address, because of 'no device left behind' and NAT issues. > > All seemed well, all the phones register normally. Looking at the > databases in FS they all show the proper ports, the proper domains, etc. > > However, every single call gets picked up as a new call via > sophia/external/... and it hits the public dialplan normally -- except > that's the wrong plan, it should hit the default plan and be identified > as sofia/internal/.... and so forth. > 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel > sofia/external/hcoin at pbx.foobar.com [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] > > What could possibly be the reason phones registered on the internal > profile have their new calls identified as sophia/external and don't hit > the correct plan? Both the phones and the freeswitch are on the same > subnet. This should be so vanilla. What am I missing? > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/cda58a97/attachment.html From vermeulen.deon at gmail.com Tue Oct 22 15:51:50 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Tue, 22 Oct 2013 13:51:50 +0200 Subject: [Freeswitch-users] SDReporter Integration with FS Message-ID: <526666D6.6010901@gmail.com> Hi I've successfully setup SDReporter, FreeSWITCH to export CDRs using cdr_csv with the Asterisk template. I've setup a cron to "automatically" push the csv file(s) to SDReporter $HOME_DIR/SDReporter/CDRConverter-4.0.8/data/INPUT/Asterisk/1/ I've tried different options in the format of the CDRs ,but no matter what I do I keep getting an " CDRFieldChannelAsterisk(String): wrong Asterisk channel. " ERROR. I've removed the "${channel_name}","${bridge_channel}" variables, but still I get the above error. I currently run FreeSWITCH Version 1.5.6b+git~20131018T052734Z~e0054af96f. Has anyone successfully got SDReporter working with FS? Thanks for any assistance. -- Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/b5d2ea67/attachment.html From neeraj.p at directi.com Tue Oct 22 11:31:40 2013 From: neeraj.p at directi.com (neeraj.p) Date: Tue, 22 Oct 2013 13:01:40 +0530 Subject: [Freeswitch-users] codec transcodation not working Message-ID: Hey, I tried to establish audio call between two clients having different codec supports . I expected that transcoding would take place. But I am getting a codec negotiation error . Here is the information about client I am using leg A : suppoerted codec : speex leg B: supported codec : G722,PCMU,PCMA,GSM,OPUS Here is some lines from my freeswitch logs 2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR. SDP: v=0 o=Zoiper 0 0 IN IP4 120.63.38.94 s=Zoiper c=IN IP4 120.63.38.94 t=0 0 m=audio 11525 RTP/AVP 110 a=rtpmap:110 speex/8000 Here is some info about my freeswitch version - 1.4 beta codecs supported - codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,B64 (STANDARD),mod_b64 codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,VP8 Video (passthru),mod_vp8 My sip_profile configurations I also tried with inbound-late-negotiaion=false . But still getting codec negotiation error. Please help . Regards, Neeraj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/40f2307c/attachment.html From alipey at gmail.com Tue Oct 22 16:25:26 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 22 Oct 2013 08:25:26 -0400 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: I think the problem here is that you are playing a file for every call for the duration of the call. The bottleneck seems to be the disk access. If there were to be two way audio path, FS would only proxy the media which would be quite faster as there would be no file reading and playing involved. Attempt a test case with fewer or no file play and only media proxy and test again. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian wrote: > The network shouldn?t be an issue, since we have at least 1Gbps lines. The > tests stay within the network.**** > > ** ** > > I forgot to mention the calls are being distributed across two machines by > a Kamailio instance.**** > > So for a total of 800 concurrent calls generated by Sipp, each machine has > 400 active calls.**** > > CPU load reaches about 70% per machine.**** > > ** ** > > At this point both FS machines are virtualized, since the performance gain > wasn?t that much compared to physical. **** > > The VM host shows it is using ~3/4 of its CPU resources.**** > > ** ** > > Htop shows that the normal priority threads(green) and the kernel > threads(red) are about the same length. **** > > ** ** > > Also, FS is running on Ubuntu Server 12.04 x64.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav > Sinyagin > *Sent:* Tuesday, October 22, 2013 11:11 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues**** > > ** ** > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this. > > **** > > ** ** > > ** ** > ------------------------------ > > *From:* Grant Bagdasarian > *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" < > freeswitch-users at lists.freeswitch.org> > *Sent:* Tuesday, October 22, 2013 10:14 AM > *Subject:* [Freeswitch-users] Audio quality issues**** > > ** ** > > Hello,**** > > **** > > I was wondering what the maximum concurrent calls for FS before audio > quality becomes an issue? I assume the specs of the machine would also > affect this.**** > > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU > and get about 800 concurrent calls at 10-20 CPS. The audio quality at these > rates is still fair, but we do notice some quality issue?s. **** > > Going above these numbers screws up the audio quality: choppy sound, audio > drops etc. We aren?t doing any heavy media processing, just simply playing > a file (G711-Alaw) which lasts about 2 minutes during the load test.**** > > These numbers are for one way audio, where Sipp doesn?t echo the RTP back. > These numbers get lower once Sipp echo?s the RTP.**** > > **** > > I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but > the performance gain on physical vs virtual isn?t that much. **** > > **** > > I disabled all the modules we don?t need, like CDR?s, conferencing, etc.** > ** > > **** > > Are there any parameters(config files)/modules that can affect the quality > of the audio stream?**** > > **** > > Regards,**** > > **** > > Grant**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/d9c312e7/attachment-0001.html From GB at cm.nl Tue Oct 22 17:22:50 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 22 Oct 2013 15:22:50 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests. How do I simulate two way audio? I know I can make Sipp send an RTP stream using a pcap file, but how do I make FS sent RTP back which is not read from disk? Does FS have an echo application? Or is it enough for Sipp to send the media? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues I think the problem here is that you are playing a file for every call for the duration of the call. The bottleneck seems to be the disk access. If there were to be two way audio path, FS would only proxy the media which would be quite faster as there would be no file reading and playing involved. Attempt a test case with fewer or no file play and only media proxy and test again. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian > wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian > To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" > Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue's. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren't doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn't echo the RTP back. These numbers get lower once Sipp echo's the RTP. I've tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn't that much. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [cid:~WRD000.jpg] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/a64428e3/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/a64428e3/attachment.jpg From mike at jerris.com Tue Oct 22 17:24:48 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Oct 2013 09:24:48 -0400 Subject: [Freeswitch-users] codec transcodation not working In-Reply-To: References: Message-ID: can you post a pastebin link to a full log of this call? On Oct 22, 2013, at 3:31 AM, "neeraj.p" wrote: > Hey, > > I tried to establish audio call between two clients having different codec supports . I expected that transcoding would take place. > But I am getting a codec negotiation error . > > Here is the information about client I am using > > leg A : > suppoerted codec : speex > > leg B: > supported codec : G722,PCMU,PCMA,GSM,OPUS > > > Here is some lines from my freeswitch logs > > 2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP. Disable 2833 dtmf and switch to INFO > 2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR. SDP: > v=0 > o=Zoiper 0 0 IN IP4 120.63.38.94 > s=Zoiper > c=IN IP4 120.63.38.94 > t=0 0 > m=audio 11525 RTP/AVP 110 > a=rtpmap:110 speex/8000 > > Here is some info about my freeswitch > > version - 1.4 beta > > codecs supported - codec,ADPCM (IMA),mod_spandsp > codec,AMR,mod_amr > codec,B64 (STANDARD),mod_b64 > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_spandsp > codec,G.723.1 6.3k,mod_g723_1 > codec,G.726 16k,mod_spandsp > codec,G.726 16k (AAL2),mod_spandsp > codec,G.726 24k,mod_spandsp > codec,G.726 24k (AAL2),mod_spandsp > codec,G.726 32k,mod_spandsp > codec,G.726 32k (AAL2),mod_spandsp > codec,G.726 40k,mod_spandsp > codec,G.726 40k (AAL2),mod_spandsp > codec,G.729,mod_g729 > codec,GSM,mod_spandsp > codec,H.261 Video (passthru),mod_h26x > codec,H.263 Video (passthru),mod_h26x > codec,H.263+ Video (passthru),mod_h26x > codec,H.263++ Video (passthru),mod_h26x > codec,H.264 Video (passthru),mod_h26x > codec,LPC-10,mod_spandsp > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE > codec,Speex,mod_speex > codec,VP8 Video (passthru),mod_vp8 > > My sip_profile configurations > > > > I also tried with inbound-late-negotiaion=false . But still getting codec negotiation error. > > Please help . > > > > Regards, > Neeraj > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/960a0b26/attachment-0001.html From mike at jerris.com Tue Oct 22 17:28:28 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Oct 2013 09:28:28 -0400 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: <9158E8F4-E00D-478C-8AC8-ED5DE9336920@jerris.com> I wouldn't be shocked if the virtual nic's are bottlenecking on pps. I've seen this before. Also possible you have a crap physical nic. What kind of nic is it and what virtualization technology? On Oct 22, 2013, at 6:01 AM, Grant Bagdasarian wrote: > The network shouldn?t be an issue, since we have at least 1Gbps lines. The tests stay within the network. > > I forgot to mention the calls are being distributed across two machines by a Kamailio instance. > So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. > CPU load reaches about 70% per machine. > > At this point both FS machines are virtualized, since the performance gain wasn?t that much compared to physical. > The VM host shows it is using ~3/4 of its CPU resources. > > Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. > > Also, FS is running on Ubuntu Server 12.04 x64. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin > Sent: Tuesday, October 22, 2013 11:11 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quality issues > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this. > > > > > From: Grant Bagdasarian > To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" > Sent: Tuesday, October 22, 2013 10:14 AM > Subject: [Freeswitch-users] Audio quality issues > > Hello, > > I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue?s. > Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren?t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. > These numbers are for one way audio, where Sipp doesn?t echo the RTP back. These numbers get lower once Sipp echo?s the RTP. > > I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn?t that much. > > I disabled all the modules we don?t need, like CDR?s, conferencing, etc. > > Are there any parameters(config files)/modules that can affect the quality of the audio stream? > > Regards, > > Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/3abf91ed/attachment.html From steveu at coppice.org Tue Oct 22 17:32:38 2013 From: steveu at coppice.org (Steve Underwood) Date: Tue, 22 Oct 2013 21:32:38 +0800 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: <52667E76.8010904@coppice.org> Hi Grant, Two possibilities spring to mind: - If your audio is coming from a disk, can that disk keep up? - How good is your ethernet switch? Notice in the second point I said how good, not how expensive. Many switches choke on a large number of small media packets, including some expensive big name products. Regards, Steve On 10/22/2013 06:01 PM, Grant Bagdasarian wrote: > > The network shouldn?t be an issue, since we have at least 1Gbps lines. > The tests stay within the network. > > I forgot to mention the calls are being distributed across two > machines by a Kamailio instance. > > So for a total of 800 concurrent calls generated by Sipp, each machine > has 400 active calls. > > CPU load reaches about 70% per machine. > > At this point both FS machines are virtualized, since the performance > gain wasn?t that much compared to physical. > > The VM host shows it is using ~3/4 of its CPU resources. > > Htop shows that the normal priority threads(green) and the kernel > threads(red) are about the same length. > > Also, FS is running on Ubuntu Server 12.04 x64. > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Stanislav Sinyagin > *Sent:* Tuesday, October 22, 2013 11:11 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this. > > ------------------------------------------------------------------------ > > *From:*Grant Bagdasarian > > *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org > )" > > > *Sent:* Tuesday, October 22, 2013 10:14 AM > *Subject:* [Freeswitch-users] Audio quality issues > > Hello, > > I was wondering what the maximum concurrent calls for FS before audio > quality becomes an issue? I assume the specs of the machine would also > affect this. > > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 > CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality > at these rates is still fair, but we do notice some quality issue?s. > > Going above these numbers screws up the audio quality: choppy sound, > audio drops etc. We aren?t doing any heavy media processing, just > simply playing a file (G711-Alaw) which lasts about 2 minutes during > the load test. > > These numbers are for one way audio, where Sipp doesn?t echo the RTP > back. These numbers get lower once Sipp echo?s the RTP. > > I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), > but the performance gain on physical vs virtual isn?t that much. > > I disabled all the modules we don?t need, like CDR?s, conferencing, etc. > > Are there any parameters(config files)/modules that can affect the > quality of the audio stream? > > Regards, > > Grant > > From hcoin at quietfountain.com Tue Oct 22 17:33:47 2013 From: hcoin at quietfountain.com (hcoin) Date: Tue, 22 Oct 2013 08:33:47 -0500 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling References: <526614D6.3020401@quietfountain.com> Message-ID: <52667EBB.2010204@quietfountain.com> Lan. Lan registered phones can't even get a tone stream. Phones register promptly, everything seems in order. It's just that every call coming in on the lan interface and 'internal' port gets identified as 'sofia/external/ and so never hits the default dial plan. Here's a bit of the internal profile. The site has some vpns so anything in the rfc1918 space has no nat. Shouldn't matter anyway as all extensions on the same lan subnet as the fs box get identified as 'external' as well. No , and The external profile has alias=false in the domains, no aliases and parse=true On 10/22/2013 06:13 AM, Donny Hardyanto wrote: > Is your client in the internet or the lan? > > Donny > > > On Tue, Oct 22, 2013 at 1:01 PM, hcoin > wrote: > > > This has been a really frustrating problem, I'm sure the answer is > simple but I just can't see it. > > I had several extensions registered to the internal profile, sending > calls out the external profile to a sip-pstn gateway, all seemed fine. > > Then created another internal profile, using a different sip port > on the > same lan address, because of 'no device left behind' and NAT issues.. > > All seemed well, all the phones register normally. Looking at the > databases in FS they all show the proper ports, the proper > domains, etc. > > However, every single call gets picked up as a new call via > sophia/external/... and it hits the public dialplan normally -- except > that's the wrong plan, it should hit the default plan and be > identified > as sofia/internal/.... and so forth. > 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel > sofia/external/hcoin at pbx.foobar.com > [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] > > What could possibly be the reason phones registered on the internal > profile have their new calls identified as sophia/external and > don't hit > the correct plan? Both the phones and the freeswitch are on the same > subnet. This should be so vanilla. What am I missing? > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www..freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/0ac4e028/attachment-0001.html From GB at cm.nl Tue Oct 22 17:41:52 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 22 Oct 2013 15:41:52 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: <9158E8F4-E00D-478C-8AC8-ED5DE9336920@jerris.com> References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> <9158E8F4-E00D-478C-8AC8-ED5DE9336920@jerris.com> Message-ID: We're using VMWare ESXi 5.1 (Free Version). NIC version: Intel I350 Gigabit Network. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, October 22, 2013 3:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues I wouldn't be shocked if the virtual nic's are bottlenecking on pps. I've seen this before. Also possible you have a crap physical nic. What kind of nic is it and what virtualization technology? On Oct 22, 2013, at 6:01 AM, Grant Bagdasarian > wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian > To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" > Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue's. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren't doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn't echo the RTP back. These numbers get lower once Sipp echo's the RTP. I've tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn't that much. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/0a54d436/attachment.html From ashwinrath at gmail.com Tue Oct 22 17:47:24 2013 From: ashwinrath at gmail.com (Ashwin Rath) Date: Tue, 22 Oct 2013 19:17:24 +0530 Subject: [Freeswitch-users] codec transcodation not working In-Reply-To: References: Message-ID: Hi Did your try forcing a codec(s) by setting the absolute_codec_string var ? try something like this and disable late negotiation. On Tue, Oct 22, 2013 at 6:54 PM, Michael Jerris wrote: > can you post a pastebin link to a full log of this call? > > On Oct 22, 2013, at 3:31 AM, "neeraj.p" wrote: > > Hey, > > I tried to establish audio call between two clients having different codec > supports . I expected that transcoding would take place. > But I am getting a codec negotiation error . > > Here is the information about client I am using > > leg A : > suppoerted codec : speex > > leg B: > supported codec : G722,PCMU,PCMA,GSM,OPUS > > > Here is some lines from my freeswitch logs > > 2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending > early media > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in > SDP. Disable 2833 dtmf and switch to INFO > 2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR. > SDP: > v=0 > o=Zoiper 0 0 IN IP4 120.63.38.94 > s=Zoiper > c=IN IP4 120.63.38.94 > t=0 0 > m=audio 11525 RTP/AVP 110 > a=rtpmap:110 speex/8000 > > Here is some info about my freeswitch > > version - 1.4 beta > > codecs supported - codec,ADPCM (IMA),mod_spandsp > codec,AMR,mod_amr > codec,B64 (STANDARD),mod_b64 > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_spandsp > codec,G.723.1 6.3k,mod_g723_1 > codec,G.726 16k,mod_spandsp > codec,G.726 16k (AAL2),mod_spandsp > codec,G.726 24k,mod_spandsp > codec,G.726 24k (AAL2),mod_spandsp > codec,G.726 32k,mod_spandsp > codec,G.726 32k (AAL2),mod_spandsp > codec,G.726 40k,mod_spandsp > codec,G.726 40k (AAL2),mod_spandsp > codec,G.729,mod_g729 > codec,GSM,mod_spandsp > codec,H.261 Video (passthru),mod_h26x > codec,H.263 Video (passthru),mod_h26x > codec,H.263+ Video (passthru),mod_h26x > codec,H.263++ Video (passthru),mod_h26x > codec,H.264 Video (passthru),mod_h26x > codec,LPC-10,mod_spandsp > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE > codec,Speex,mod_speex > codec,VP8 Video (passthru),mod_vp8 > > My sip_profile configurations > > > > > > > I also tried with inbound-late-negotiaion=false . But still getting codec > negotiation error. > > Please help . > > > > Regards, > Neeraj > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/c4c0c388/attachment-0001.html From alipey at gmail.com Tue Oct 22 17:50:03 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 22 Oct 2013 09:50:03 -0400 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: You can make calls from sipp that also terminates on sipp and then play a wave file in sipp. Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls. Does this make sense? On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian wrote: > A while back I used iotop to measure the disk access, and FS was hardly > using any io during tests.**** > > ** ** > > How do I simulate two way audio? **** > > I know I can make Sipp send an RTP stream using a pcap file, but how do I > make FS sent RTP back which is not read from disk? Does FS have an echo > application?**** > > Or is it enough for Sipp to send the media?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Tuesday, October 22, 2013 2:25 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues**** > > ** ** > > I think the problem here is that you are playing a file for every call for > the duration of the call. The bottleneck seems to be the disk access. If > there were to be two way audio path, FS would only proxy the media which > would be quite faster as there would be no file reading and playing > involved. Attempt a test case with fewer or no file play and only media > proxy and test again.**** > > ** ** > > ** ** > > On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian wrote:**** > > The network shouldn?t be an issue, since we have at least 1Gbps lines. The > tests stay within the network.**** > > **** > > I forgot to mention the calls are being distributed across two machines by > a Kamailio instance.**** > > So for a total of 800 concurrent calls generated by Sipp, each machine has > 400 active calls.**** > > CPU load reaches about 70% per machine.**** > > **** > > At this point both FS machines are virtualized, since the performance gain > wasn?t that much compared to physical. **** > > The VM host shows it is using ~3/4 of its CPU resources.**** > > **** > > Htop shows that the normal priority threads(green) and the kernel > threads(red) are about the same length. **** > > **** > > Also, FS is running on Ubuntu Server 12.04 x64.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav > Sinyagin > *Sent:* Tuesday, October 22, 2013 11:11 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues**** > > **** > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this.**** > > **** > > **** > ------------------------------ > > *From:* Grant Bagdasarian > *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" < > freeswitch-users at lists.freeswitch.org> > *Sent:* Tuesday, October 22, 2013 10:14 AM > *Subject:* [Freeswitch-users] Audio quality issues**** > > **** > > Hello,**** > > **** > > I was wondering what the maximum concurrent calls for FS before audio > quality becomes an issue? I assume the specs of the machine would also > affect this.**** > > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU > and get about 800 concurrent calls at 10-20 CPS. The audio quality at these > rates is still fair, but we do notice some quality issue?s. **** > > Going above these numbers screws up the audio quality: choppy sound, audio > drops etc. We aren?t doing any heavy media processing, just simply playing > a file (G711-Alaw) which lasts about 2 minutes during the load test.**** > > These numbers are for one way audio, where Sipp doesn?t echo the RTP back. > These numbers get lower once Sipp echo?s the RTP.**** > > **** > > I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but > the performance gain on physical vs virtual isn?t that much. **** > > **** > > I disabled all the modules we don?t need, like CDR?s, conferencing, etc.** > ** > > **** > > Are there any parameters(config files)/modules that can affect the quality > of the audio stream?**** > > **** > > Regards,**** > > **** > > Grant**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > [image: Image removed by sender.]**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/6ff24d13/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 823 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/6ff24d13/attachment.jpe From mehroz.ashraf85 at gmail.com Tue Oct 22 17:56:31 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 22 Oct 2013 06:56:31 -0700 (PDT) Subject: [Freeswitch-users] Video switching In-Reply-To: <1375333454405-7593468.post@n2.nabble.com> References: <1375267219620-7593439.post@n2.nabble.com> <1375333454405-7593468.post@n2.nabble.com> Message-ID: <1382450191344-7595891.post@n2.nabble.com> Upon further investigation, I have observed that FS is always changing the Video port while keeping the audio port same in SDP updates. Probably this is the reason the video is lost after the 2nd switch from audio to video. and Eventually, the new offered video ports does not opens at system level (netstate -ntulp) and older ports can bee seen bound by Freeswitch, but a tcpdump on those new ports does show rtp data flowing. Is that the normal behaviour? if not, what could be the reason FS changing video ports in each new session. I am on Proxy mode. Looking forward. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-switching-tp7593439p7595891.html Sent from the freeswitch-users mailing list archive at Nabble.com. From GB at cm.nl Tue Oct 22 18:03:29 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 22 Oct 2013 16:03:29 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: Yes, it does! I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo Echo application for FS. Going to see if that works first. If not, I'll setup a Sipp in server mode. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues You can make calls from sipp that also terminates on sipp and then play a wave file in sipp. Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls. Does this make sense? On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian > wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests. How do I simulate two way audio? I know I can make Sipp send an RTP stream using a pcap file, but how do I make FS sent RTP back which is not read from disk? Does FS have an echo application? Or is it enough for Sipp to send the media? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues I think the problem here is that you are playing a file for every call for the duration of the call. The bottleneck seems to be the disk access. If there were to be two way audio path, FS would only proxy the media which would be quite faster as there would be no file reading and playing involved. Attempt a test case with fewer or no file play and only media proxy and test again. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian > wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian > To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" > Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue's. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren't doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn't echo the RTP back. These numbers get lower once Sipp echo's the RTP. I've tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn't that much. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [cid:~WRD000.jpg] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [cid:~WRD000.jpg] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/6ebdb9d3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD000.jpg Type: image/jpeg Size: 823 bytes Desc: ~WRD000.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/6ebdb9d3/attachment-0001.jpg From mike at jerris.com Tue Oct 22 18:04:15 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Oct 2013 10:04:15 -0400 Subject: [Freeswitch-users] Video switching In-Reply-To: <1382450191344-7595891.post@n2.nabble.com> References: <1375267219620-7593439.post@n2.nabble.com> <1375333454405-7593468.post@n2.nabble.com> <1382450191344-7595891.post@n2.nabble.com> Message-ID: <12DFDB1C-89F1-40B6-B8CD-BF02AFED9EB0@jerris.com> Please file a bug at jira.freeswitch.org On Oct 22, 2013, at 9:56 AM, mehroz wrote: > Upon further investigation, I have observed that FS is always changing the > Video port while keeping the audio port same in SDP updates. > Probably this is the reason the video is lost after the 2nd switch from > audio to video. and Eventually, the new offered video ports does not opens > at system level (netstate -ntulp) and older ports can bee seen bound by > Freeswitch, but a tcpdump on those new ports does show rtp data flowing. > > Is that the normal behaviour? if not, what could be the reason FS changing > video ports in each new session. I am on Proxy mode. > > Looking forward. > Thanks > > From alipey at gmail.com Tue Oct 22 19:28:43 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 22 Oct 2013 11:28:43 -0400 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: I would try both and see if there is any difference. Echo should not create an overhead but you never konow until you test it :) Please do post your results here. These would be some valuable info for the community. On Tue, Oct 22, 2013 at 10:03 AM, Grant Bagdasarian wrote: > Yes, it does! **** > > ** ** > > I also found this: > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo**** > > ** ** > > Echo application for FS. **** > > Going to see if that works first. If not, I?ll setup a Sipp in server mode. > **** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Tuesday, October 22, 2013 3:50 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues**** > > ** ** > > You can make calls from sipp that also terminates on sipp and then play a > wave file in sipp.**** > > ** ** > > Change your dial plan in FS to route the calls to an instant of sipp that > can terminate the calls.**** > > ** ** > > Does this make sense?**** > > ** ** > > On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian wrote:**** > > A while back I used iotop to measure the disk access, and FS was hardly > using any io during tests.**** > > **** > > How do I simulate two way audio? **** > > I know I can make Sipp send an RTP stream using a pcap file, but how do I > make FS sent RTP back which is not read from disk? Does FS have an echo > application?**** > > Or is it enough for Sipp to send the media?**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey > *Sent:* Tuesday, October 22, 2013 2:25 PM**** > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues**** > > **** > > I think the problem here is that you are playing a file for every call for > the duration of the call. The bottleneck seems to be the disk access. If > there were to be two way audio path, FS would only proxy the media which > would be quite faster as there would be no file reading and playing > involved. Attempt a test case with fewer or no file play and only media > proxy and test again.**** > > **** > > **** > > On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian wrote:**** > > The network shouldn?t be an issue, since we have at least 1Gbps lines. The > tests stay within the network.**** > > **** > > I forgot to mention the calls are being distributed across two machines by > a Kamailio instance.**** > > So for a total of 800 concurrent calls generated by Sipp, each machine has > 400 active calls.**** > > CPU load reaches about 70% per machine.**** > > **** > > At this point both FS machines are virtualized, since the performance gain > wasn?t that much compared to physical. **** > > The VM host shows it is using ~3/4 of its CPU resources.**** > > **** > > Htop shows that the normal priority threads(green) and the kernel > threads(red) are about the same length. **** > > **** > > Also, FS is running on Ubuntu Server 12.04 x64.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav > Sinyagin > *Sent:* Tuesday, October 22, 2013 11:11 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues**** > > **** > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this.**** > > **** > > **** > ------------------------------ > > *From:* Grant Bagdasarian > *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" < > freeswitch-users at lists.freeswitch.org> > *Sent:* Tuesday, October 22, 2013 10:14 AM > *Subject:* [Freeswitch-users] Audio quality issues**** > > **** > > Hello,**** > > **** > > I was wondering what the maximum concurrent calls for FS before audio > quality becomes an issue? I assume the specs of the machine would also > affect this.**** > > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU > and get about 800 concurrent calls at 10-20 CPS. The audio quality at these > rates is still fair, but we do notice some quality issue?s. **** > > Going above these numbers screws up the audio quality: choppy sound, audio > drops etc. We aren?t doing any heavy media processing, just simply playing > a file (G711-Alaw) which lasts about 2 minutes during the load test.**** > > These numbers are for one way audio, where Sipp doesn?t echo the RTP back. > These numbers get lower once Sipp echo?s the RTP.**** > > **** > > I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but > the performance gain on physical vs virtual isn?t that much. **** > > **** > > I disabled all the modules we don?t need, like CDR?s, conferencing, etc.** > ** > > **** > > Are there any parameters(config files)/modules that can affect the quality > of the audio stream?**** > > **** > > Regards,**** > > **** > > Grant**** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > [image: Image removed by sender.]**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > [image: Image removed by sender.]**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/3c25e78e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 823 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/3c25e78e/attachment-0001.jpe From nandy1925 at gmail.com Tue Oct 22 19:34:08 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Tue, 22 Oct 2013 23:34:08 +0800 Subject: [Freeswitch-users] FreeTDM tones.conf Disconnect Supervision In-Reply-To: <1382436346467-7595879.post@n2.nabble.com> References: <1382436346467-7595879.post@n2.nabble.com> Message-ID: [Solved] I also used tone_detect to hangup the PSTN line. FreeTDM can detect call progress tone (busy, disconnect) BUT ... as mentioned above it detects TONES ONLY NOT CADENCE. It works if 2 or more tones are used even w/o detecting the cadence. It's becomes a problem when only one tone is used. To detect cadence, mod_spandsp is recommended http://wiki.freeswitch.org/wiki/Mod_spandsp#Call_Progress It fires Events so a script is needed to catch them and fire the hangup signal. /Nandy On Tue, Oct 22, 2013 at 6:05 PM, xmppser wrote: > Hi? > I have this issue also, I use sangoma A400, use sip phone call pstn mobile, > when mobile hangup, > the sip phone can not hangup, i have add tone_detect app in dialplan, it > seems sangoma A400 and freetdm can not detect busy tones or disconct > srperversion from pstm. but asterisk & zaptel have this detect. > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeTDM-tones-conf-Disconnect-Supervision-tp7595763p7595879.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/08de7e20/attachment.html From andretodd at verizon.net Tue Oct 22 19:40:00 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 22 Oct 2013 11:40:00 -0400 Subject: [Freeswitch-users] Limit help Message-ID: <0MV2006VAU7PQG50@vms173019.mailsrvcs.net> Hi, how do I set the CPS and Port limits on a bridge Can I set both on one application or do I need to set 2 limits? 5 ports 5 cps Can I do this for both 5 ports and 5 cps? Also how do I do the same on a gateway? I need to make sure I don't send too many calls to the provider. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/13171bd6/attachment.html From mytemike72 at gmail.com Tue Oct 22 19:51:30 2013 From: mytemike72 at gmail.com (Michael Lutz) Date: Tue, 22 Oct 2013 17:51:30 +0200 Subject: [Freeswitch-users] Controlling the conference via event socket Message-ID: Hi, Is there anybody with experience or knows a way of controlling a session in a conference via event socket? I would like to to be able to make a session 'leave' a conference, and set the 'mute' setting via the ESL. I did not find any documentation on this, other than being able to subscribe and read the maintenance events. Anyone got idea's on this one? Regards, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/17f21a8d/attachment.html From anthony.minessale at gmail.com Tue Oct 22 20:00:43 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Oct 2013 11:00:43 -0500 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: Putting the sound file on a ramdisk usually counts out disk IO. You've made a mistake somewhere that you should look for if you saw no gain from using physical machines over virtuals. I'm always squeamish about helping on load test questions because we tend to get caught up in it and eventually use up a lot of time just steering people into things that fall outside the scope of the project. Baseline: I would try one of your servers on a modern kernel (Debian 7 or equiv) using latest HEAD build from master or stable branch and put your test extension high in your dialplan to avoid the extra stuff that goes on in the demo pbx config. Also disable presence on sip with manage-presence and manage-shared-appearance both commented out or set to false in the sofia profile. On Tue, Oct 22, 2013 at 10:28 AM, Ali Pey wrote: > I would try both and see if there is any difference. Echo should not > create an overhead but you never konow until you test it :) > > Please do post your results here. These would be some valuable info for > the community. > > > On Tue, Oct 22, 2013 at 10:03 AM, Grant Bagdasarian wrote: > >> Yes, it does! **** >> >> ** ** >> >> I also found this: >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo**** >> >> ** ** >> >> Echo application for FS. **** >> >> Going to see if that works first. If not, I?ll setup a Sipp in server >> mode.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey >> *Sent:* Tuesday, October 22, 2013 3:50 PM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Audio quality issues**** >> >> ** ** >> >> You can make calls from sipp that also terminates on sipp and then play a >> wave file in sipp.**** >> >> ** ** >> >> Change your dial plan in FS to route the calls to an instant of sipp that >> can terminate the calls.**** >> >> ** ** >> >> Does this make sense?**** >> >> ** ** >> >> On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian wrote:**** >> >> A while back I used iotop to measure the disk access, and FS was hardly >> using any io during tests.**** >> >> **** >> >> How do I simulate two way audio? **** >> >> I know I can make Sipp send an RTP stream using a pcap file, but how do I >> make FS sent RTP back which is not read from disk? Does FS have an echo >> application?**** >> >> Or is it enough for Sipp to send the media?**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ali Pey >> *Sent:* Tuesday, October 22, 2013 2:25 PM**** >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Audio quality issues**** >> >> **** >> >> I think the problem here is that you are playing a file for every call >> for the duration of the call. The bottleneck seems to be the disk access. >> If there were to be two way audio path, FS would only proxy the media which >> would be quite faster as there would be no file reading and playing >> involved. Attempt a test case with fewer or no file play and only media >> proxy and test again.**** >> >> **** >> >> **** >> >> On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian wrote:**** >> >> The network shouldn?t be an issue, since we have at least 1Gbps lines. >> The tests stay within the network.**** >> >> **** >> >> I forgot to mention the calls are being distributed across two machines >> by a Kamailio instance.**** >> >> So for a total of 800 concurrent calls generated by Sipp, each machine >> has 400 active calls.**** >> >> CPU load reaches about 70% per machine.**** >> >> **** >> >> At this point both FS machines are virtualized, since the performance >> gain wasn?t that much compared to physical. **** >> >> The VM host shows it is using ~3/4 of its CPU resources.**** >> >> **** >> >> Htop shows that the normal priority threads(green) and the kernel >> threads(red) are about the same length. **** >> >> **** >> >> Also, FS is running on Ubuntu Server 12.04 x64.**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav >> Sinyagin >> *Sent:* Tuesday, October 22, 2013 11:11 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Audio quality issues**** >> >> **** >> >> 800 calls at 64kbps is 51Mbps. >> Could there be a network issue, like a 100Mbps line between the endpoints? >> >> How heavy is your CPU load? "htop" command would be helpful in this.**** >> >> **** >> >> **** >> ------------------------------ >> >> *From:* Grant Bagdasarian >> *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" < >> freeswitch-users at lists.freeswitch.org> >> *Sent:* Tuesday, October 22, 2013 10:14 AM >> *Subject:* [Freeswitch-users] Audio quality issues**** >> >> **** >> >> Hello,**** >> >> **** >> >> I was wondering what the maximum concurrent calls for FS before audio >> quality becomes an issue? I assume the specs of the machine would also >> affect this.**** >> >> We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU >> and get about 800 concurrent calls at 10-20 CPS. The audio quality at these >> rates is still fair, but we do notice some quality issue?s. **** >> >> Going above these numbers screws up the audio quality: choppy sound, >> audio drops etc. We aren?t doing any heavy media processing, just simply >> playing a file (G711-Alaw) which lasts about 2 minutes during the load test. >> **** >> >> These numbers are for one way audio, where Sipp doesn?t echo the RTP >> back. These numbers get lower once Sipp echo?s the RTP.**** >> >> **** >> >> I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but >> the performance gain on physical vs virtual isn?t that much. **** >> >> **** >> >> I disabled all the modules we don?t need, like CDR?s, conferencing, etc.* >> *** >> >> **** >> >> Are there any parameters(config files)/modules that can affect the >> quality of the audio stream?**** >> >> **** >> >> Regards,**** >> >> **** >> >> Grant**** >> >> **** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> **** >> >> [image: Image removed by sender.]**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> ** ** >> >> [image: Image removed by sender.]**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/ba11d6d4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 823 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/ba11d6d4/attachment-0001.jpe From matt at williamsmjw.com Tue Oct 22 20:01:30 2013 From: matt at williamsmjw.com (Matt Williams) Date: Tue, 22 Oct 2013 11:01:30 -0500 Subject: [Freeswitch-users] Limit help In-Reply-To: <0MV2006VAU7PQG50@vms173019.mailsrvcs.net> References: <0MV2006VAU7PQG50@vms173019.mailsrvcs.net> Message-ID: You need to set your CPS and port limits separately. You can set limits per gateway the same way you would limit any application. It is not bound to any specific application. The values you use are completely arbitrary. That said here is an example from the Wiki http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits On Tue, Oct 22, 2013 at 10:40 AM, Andre Demattia wrote: > Hi, how do I set the CPS and Port limits on a bridge > > Can I set both on one application or do I need to set 2 limits? > > 5 ports > > > 5 cps > > > Can I do this for both 5 ports and 5 cps? > > > > Also how do I do the same on a gateway? I need to make sure I don't send > too many calls to the provider. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/2e093b8b/attachment.html From andretodd at verizon.net Tue Oct 22 20:21:59 2013 From: andretodd at verizon.net (Andre Demattia) Date: Tue, 22 Oct 2013 12:21:59 -0400 Subject: [Freeswitch-users] Limit help Message-ID: <0MV20027DW5OHY90@vms173005.mailsrvcs.net> Can I use limit_execute for a outbound gateway in place of the loop back option? -----Original Message----- From: "Matt Williams" Sent: ?10/?22/?2013 12:01 PM To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] Limit help You need to set your CPS and port limits separately. You can set limits per gateway the same way you would limit any application. It is not bound to any specific application. The values you use are completely arbitrary. That said here is an example from the Wiki http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits On Tue, Oct 22, 2013 at 10:40 AM, Andre Demattia wrote: Hi, how do I set the CPS and Port limits on a bridge Can I set both on one application or do I need to set 2 limits? 5 ports 5 cps Can I do this for both 5 ports and 5 cps? Also how do I do the same on a gateway? I need to make sure I don't send too many calls to the provider. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/413dd8a8/attachment.html From hcoin at quietfountain.com Tue Oct 22 21:23:25 2013 From: hcoin at quietfountain.com (hcoin) Date: Tue, 22 Oct 2013 12:23:25 -0500 Subject: [Freeswitch-users] How to turn on debug output just before 'New Channel sofia/...' ? References: <0MV2006VAU7PQG50@vms173019.mailsrvcs.net> Message-ID: <5266B48D.80402@quietfountain.com> What turns on debugging info that shows where a new call from a local extension to FS gets assigned to one of FS's sip profiles? The bit that comes after the sip traces but before "switch_channel.c:1034 New Channel sofia/...." I need to get more debugging info that connects an ip and port with an 'invite' to a FS sip profile. Because in my case the profile FS comes up with (external) has nothing to do with the port the invite from the registered phone actually came in on (internal). Thanks! From miha at softnet.si Tue Oct 22 21:28:16 2013 From: miha at softnet.si (Miha) Date: Tue, 22 Oct 2013 19:28:16 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: Hi, I would be also interested in this as i am experiancing same issue, poor audio quality and media dropping. Br, Miha On Tue, 22 Oct 2013 16:03:29 +0200 Grant Bagdasarian wrote: > Yes, it does! > > I also found this: > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo > > Echo application for FS. > Going to see if that works first. If not, I'll setup a > Sipp in server mode. > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On > Behalf Of Ali Pey > Sent: Tuesday, October 22, 2013 3:50 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quality issues > > You can make calls from sipp that also terminates on sipp > and then play a wave file in sipp. > > Change your dial plan in FS to route the calls to an > instant of sipp that can terminate the calls. > > Does this make sense? > > On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian > > wrote: > A while back I used iotop to measure the disk access, and > FS was hardly using any io during tests. > > How do I simulate two way audio? > I know I can make Sipp send an RTP stream using a pcap > file, but how do I make FS sent RTP back which is not > read from disk? Does FS have an echo application? > Or is it enough for Sipp to send the media? > > From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] > On Behalf Of Ali Pey > Sent: Tuesday, October 22, 2013 2:25 PM > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quality issues > > I think the problem here is that you are playing a file > for every call for the duration of the call. The > bottleneck seems to be the disk access. If there were to > be two way audio path, FS would only proxy the media > which would be quite faster as there would be no file > reading and playing involved. Attempt a test case with > fewer or no file play and only media proxy and test > again. > > > On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian > > wrote: > The network shouldn't be an issue, since we have at least > 1Gbps lines. The tests stay within the network. > > I forgot to mention the calls are being distributed > across two machines by a Kamailio instance. > So for a total of 800 concurrent calls generated by Sipp, > each machine has 400 active calls. > CPU load reaches about 70% per machine. > > At this point both FS machines are virtualized, since the > performance gain wasn't that much compared to physical. > The VM host shows it is using ~3/4 of its CPU resources. > > Htop shows that the normal priority threads(green) and > the kernel threads(red) are about the same length. > > Also, FS is running on Ubuntu Server 12.04 x64. > > From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] > On Behalf Of Stanislav Sinyagin > Sent: Tuesday, October 22, 2013 11:11 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Audio quality issues > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line > between the endpoints? > > How heavy is your CPU load? "htop" command would be > helpful in this. > > > ________________________________ > From: Grant Bagdasarian > > To: "FreeSWITCH Users Help > (freeswitch-users at lists.freeswitch.org)" > > > Sent: Tuesday, October 22, 2013 10:14 AM > Subject: [Freeswitch-users] Audio quality issues > > Hello, > > I was wondering what the maximum concurrent calls for FS > before audio quality becomes an issue? I assume the specs > of the machine would also affect this. > We are currently running FS on a Six Core (12 Threads) > Intel E5-2430 CPU and get about 800 concurrent calls at > 10-20 CPS. The audio quality at these rates is still > fair, but we do notice some quality issue's. > Going above these numbers screws up the audio quality: > choppy sound, audio drops etc. We aren't doing any heavy > media processing, just simply playing a file (G711-Alaw) > which lasts about 2 minutes during the load test. > These numbers are for one way audio, where Sipp doesn't > echo the RTP back. These numbers get lower once Sipp > echo's the RTP. > > I've tried FS on a physical box and also on a virtual box > (ESXi 5.1), but the performance gain on physical vs > virtual isn't that much. > > I disabled all the modules we don't need, like CDR's, > conferencing, etc. > > Are there any parameters(config files)/modules that can > affect the quality of the audio stream? > > Regards, > > Grant > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > [cid:~WRD000.jpg] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > [cid:~WRD000.jpg] From krice at freeswitch.org Tue Oct 22 22:36:23 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 22 Oct 2013 13:36:23 -0500 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern Message-ID: ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM Eastern Live! Cal Lemming, Will be making a 0day disclosure! Don?t miss this exclusive presentation for the FreeSwitch community! This is one you will not want to miss! Join us on the FreeSWITCH Bridge for this Special Presentation! For conference call access information visit http://ss7.us/call888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://www.ss7.us/cluecon-gplus FB ClueCon : http://www.ss7.us/cluecon-fb G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus FB FreeSWITCH : http://www.ss7.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/bf760ab4/attachment-0001.html From peter at olssononline.se Tue Oct 22 22:38:21 2013 From: peter at olssononline.se (Peter Olsson) Date: Tue, 22 Oct 2013 20:38:21 +0200 Subject: [Freeswitch-users] Controlling the conference via event socket In-Reply-To: References: Message-ID: <4420468540895867213@unknownmsgid> Using ESL gives you access to all API commands. Look into the API commands for conference, and you should find you answer there. /Peter > 22 okt 2013 kl. 17:55 skrev Michael Lutz : > > Hi, > > Is there anybody with experience or knows a way of controlling a session in a conference via event socket? > > I would like to to be able to make a session 'leave' a conference, and set the 'mute' setting via the ESL. > > I did not find any documentation on this, other than being able to subscribe and read the maintenance events. > > Anyone got idea's on this one? > > Regards, > Mike. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vermeulen.deon at gmail.com Tue Oct 22 22:41:19 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Tue, 22 Oct 2013 20:41:19 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: My 2cents is to move to bare metal. We ran a Guest on Commercial Licensed ESX and without real load (>50%) started seeing bad behavior. We moved to Bare Metal just over 3weeks ago and since then no more strange behavior. I've setup OpenVZ for LAB environment and so far no issues with almost the same load when using Commercial Licensed ESX. Kind Regards Sent from my iPhone > On Oct 22, 2013, at 19:28, "Miha" wrote: > > Hi, > > > I would be also interested in this as i am experiancing > same issue, poor audio quality and media dropping. > > Br, > > Miha > > On Tue, 22 Oct 2013 16:03:29 +0200 > Grant Bagdasarian wrote: >> Yes, it does! >> >> I also found this: > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo >> >> Echo application for FS. >> Going to see if that works first. If not, I'll setup a >> Sipp in server mode. >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On >> Behalf Of Ali Pey >> Sent: Tuesday, October 22, 2013 3:50 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Audio quality issues >> >> You can make calls from sipp that also terminates on sipp >> and then play a wave file in sipp. >> >> Change your dial plan in FS to route the calls to an >> instant of sipp that can terminate the calls. >> >> Does this make sense? >> >> On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian >> > wrote: >> A while back I used iotop to measure the disk access, and >> FS was hardly using any io during tests. >> >> How do I simulate two way audio? >> I know I can make Sipp send an RTP stream using a pcap >> file, but how do I make FS sent RTP back which is not >> read from disk? Does FS have an echo application? >> Or is it enough for Sipp to send the media? >> >> From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] >> On Behalf Of Ali Pey >> Sent: Tuesday, October 22, 2013 2:25 PM >> >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Audio quality issues >> >> I think the problem here is that you are playing a file >> for every call for the duration of the call. The >> bottleneck seems to be the disk access. If there were to >> be two way audio path, FS would only proxy the media >> which would be quite faster as there would be no file >> reading and playing involved. Attempt a test case with >> fewer or no file play and only media proxy and test >> again. >> >> >> On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian >> > wrote: >> The network shouldn't be an issue, since we have at least >> 1Gbps lines. The tests stay within the network. >> >> I forgot to mention the calls are being distributed >> across two machines by a Kamailio instance. >> So for a total of 800 concurrent calls generated by Sipp, >> each machine has 400 active calls. >> CPU load reaches about 70% per machine. >> >> At this point both FS machines are virtualized, since the >> performance gain wasn't that much compared to physical. >> The VM host shows it is using ~3/4 of its CPU resources. >> >> Htop shows that the normal priority threads(green) and >> the kernel threads(red) are about the same length. >> >> Also, FS is running on Ubuntu Server 12.04 x64. >> >> From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] >> On Behalf Of Stanislav Sinyagin >> Sent: Tuesday, October 22, 2013 11:11 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Audio quality issues >> >> 800 calls at 64kbps is 51Mbps. >> Could there be a network issue, like a 100Mbps line >> between the endpoints? >> >> How heavy is your CPU load? "htop" command would be >> helpful in this. >> >> >> ________________________________ >> From: Grant Bagdasarian > >> To: "FreeSWITCH Users Help > (freeswitch-users at lists.freeswitch.org)" > > >> Sent: Tuesday, October 22, 2013 10:14 AM >> Subject: [Freeswitch-users] Audio quality issues >> >> Hello, >> >> I was wondering what the maximum concurrent calls for FS >> before audio quality becomes an issue? I assume the specs >> of the machine would also affect this. >> We are currently running FS on a Six Core (12 Threads) >> Intel E5-2430 CPU and get about 800 concurrent calls at >> 10-20 CPS. The audio quality at these rates is still >> fair, but we do notice some quality issue's. >> Going above these numbers screws up the audio quality: >> choppy sound, audio drops etc. We aren't doing any heavy >> media processing, just simply playing a file (G711-Alaw) >> which lasts about 2 minutes during the load test. >> These numbers are for one way audio, where Sipp doesn't >> echo the RTP back. These numbers get lower once Sipp >> echo's the RTP. >> >> I've tried FS on a physical box and also on a virtual box >> (ESXi 5.1), but the performance gain on physical vs >> virtual isn't that much. >> >> I disabled all the modules we don't need, like CDR's, >> conferencing, etc. >> >> Are there any parameters(config files)/modules that can >> affect the quality of the audio stream? >> >> Regards, >> >> Grant > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication >> Server >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication >> Server >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> [cid:~WRD000.jpg] > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication >> Server >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> [cid:~WRD000.jpg] > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From schoch+freeswitch.org at xwin32.com Wed Oct 23 00:07:13 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 22 Oct 2013 13:07:13 -0700 Subject: [Freeswitch-users] Voicemail notification In-Reply-To: References: Message-ID: On Mon, Oct 21, 2013 at 2:21 PM, Ali Pey wrote: > I need to send a customized voicemail notification based on certain > condition once a voicemail is recorded. > I did this externally to FreeSWITCH by sending the vm-notify to a perl script. We run FreeSWITCH on CentOS, so it was easy to set up an email address that I called "vm-page". For this notification, I didn't need the audio file, so I used vm-notify instead of vm-mailto. In my directory file, I used this: In /etc/aliases, I have this: # Paging notify from phone system vm-page: "|/etc/smrsh/vm-page" And /etc/smrsh/vm-page is a perl script that parses the message, looking for the 'From' line (which contains the caller-ID info), the 'Return-Path' header (which contains the voice mailbox number), the 'X-Priority' line (which can identify an urgent message), and the 'X-Voicemail-Length' header. In our system, this script parses this stuff, reformats it, and then sends it through an email-to-SMS gateway at vtext.com. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/9ceefb67/attachment.html From lists at telefaks.de Wed Oct 23 00:25:51 2013 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 22 Oct 2013 22:25:51 +0200 Subject: [Freeswitch-users] loopback_bowout Message-ID: <5266DF4F.2000002@telefaks.de> Hello, I have a strange behaviour with forwarding a call from an external gateway to another number on the same gateway. * when I forward a call from an internal number to an external number via loopback, I firstly get 4 channels. After answering the call, I have 2 channels remaining and audio is fine * when I forward a call from an*_*external_ number to an external number via loopback, I firstly get 4 channels. After answering the call, I still have 4 channels remaining and audio is _not_ there. Freeswitch is dialling via "external" profile via port 5080. I also tried to set "loopback_bowout=true", but this did not change anything. The dialstring is: The dialplan for both scenarios is the same. Anybody had the same issue and knows how to overcome this? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/a61d1b3f/attachment.html From brian at freeswitch.org Wed Oct 23 00:36:09 2013 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Oct 2013 15:36:09 -0500 Subject: [Freeswitch-users] loopback_bowout In-Reply-To: <5266DF4F.2000002@telefaks.de> References: <5266DF4F.2000002@telefaks.de> Message-ID: <93392EFB-3CC9-4B00-B09A-A7A1449E00B8@freeswitch.org> What git rev are you on because we just fixed this last week! -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 22, 2013, at 3:25 PM, Peter Steinbach wrote: > Hello, > > I have a strange behaviour with forwarding a call from an external gateway to another number on the same gateway. > ? when I forward a call from an internal number to an external number via loopback, I firstly get 4 channels. After answering the call, I have 2 channels remaining and audio is fine > ? when I forward a call from an _external_ number to an external number via loopback, I firstly get 4 channels. After answering the call, I still have 4 channels remaining and audio is _not_ there. Freeswitch is dialling via "external" profile via port 5080. I also tried to set "loopback_bowout=true", but this did not change anything. > The dialstring is: > The dialplan for both scenarios is the same. > Anybody had the same issue and knows how to overcome this? > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > > mailto:lists > (att) telefaks.de > Internet: > www.telefaks.de > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From grcamauer at gmail.com Wed Oct 23 01:39:50 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 22 Oct 2013 19:39:50 -0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: You have turned off CDR, but what about logging? Logging at certain levels gets pretty taxing at these call rates. Have you implemented all the hints from https://wiki.freeswitch.org/wiki/Performance_Tuning and https://wiki.freeswitch.org/wiki/Performance_testing_and_configurations ? Guillermo On Tue, Oct 22, 2013 at 3:41 PM, Deon Vermeulen wrote: > My 2cents is to move to bare metal. > We ran a Guest on Commercial Licensed ESX and without real load (>50%) > started seeing bad behavior. > We moved to Bare Metal just over 3weeks ago and since then no more strange > behavior. > > I've setup OpenVZ for LAB environment and so far no issues with almost the > same load when using Commercial Licensed ESX. > > > Kind Regards > > Sent from my iPhone > > > On Oct 22, 2013, at 19:28, "Miha" wrote: > > > > Hi, > > > > > > I would be also interested in this as i am experiancing > > same issue, poor audio quality and media dropping. > > > > Br, > > > > Miha > > > > On Tue, 22 Oct 2013 16:03:29 +0200 > > Grant Bagdasarian wrote: > >> Yes, it does! > >> > >> I also found this: > > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo > >> > >> Echo application for FS. > >> Going to see if that works first. If not, I'll setup a > >> Sipp in server mode. > >> > >> From: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On > >> Behalf Of Ali Pey > >> Sent: Tuesday, October 22, 2013 3:50 PM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] Audio quality issues > >> > >> You can make calls from sipp that also terminates on sipp > >> and then play a wave file in sipp. > >> > >> Change your dial plan in FS to route the calls to an > >> instant of sipp that can terminate the calls. > >> > >> Does this make sense? > >> > >> On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian > >> > wrote: > >> A while back I used iotop to measure the disk access, and > >> FS was hardly using any io during tests. > >> > >> How do I simulate two way audio? > >> I know I can make Sipp send an RTP stream using a pcap > >> file, but how do I make FS sent RTP back which is not > >> read from disk? Does FS have an echo application? > >> Or is it enough for Sipp to send the media? > >> > >> From: > > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> > > [mailto:freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] > >> On Behalf Of Ali Pey > >> Sent: Tuesday, October 22, 2013 2:25 PM > >> > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] Audio quality issues > >> > >> I think the problem here is that you are playing a file > >> for every call for the duration of the call. The > >> bottleneck seems to be the disk access. If there were to > >> be two way audio path, FS would only proxy the media > >> which would be quite faster as there would be no file > >> reading and playing involved. Attempt a test case with > >> fewer or no file play and only media proxy and test > >> again. > >> > >> > >> On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian > >> > wrote: > >> The network shouldn't be an issue, since we have at least > >> 1Gbps lines. The tests stay within the network. > >> > >> I forgot to mention the calls are being distributed > >> across two machines by a Kamailio instance. > >> So for a total of 800 concurrent calls generated by Sipp, > >> each machine has 400 active calls. > >> CPU load reaches about 70% per machine. > >> > >> At this point both FS machines are virtualized, since the > >> performance gain wasn't that much compared to physical. > >> The VM host shows it is using ~3/4 of its CPU resources. > >> > >> Htop shows that the normal priority threads(green) and > >> the kernel threads(red) are about the same length. > >> > >> Also, FS is running on Ubuntu Server 12.04 x64. > >> > >> From: > > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> > > [mailto:freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] > >> On Behalf Of Stanislav Sinyagin > >> Sent: Tuesday, October 22, 2013 11:11 AM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] Audio quality issues > >> > >> 800 calls at 64kbps is 51Mbps. > >> Could there be a network issue, like a 100Mbps line > >> between the endpoints? > >> > >> How heavy is your CPU load? "htop" command would be > >> helpful in this. > >> > >> > >> ________________________________ > >> From: Grant Bagdasarian > > >> To: "FreeSWITCH Users Help > > (freeswitch-users at lists.freeswitch.org freeswitch-users at lists.freeswitch.org>)" > > freeswitch-users at lists.freeswitch.org>> > >> Sent: Tuesday, October 22, 2013 10:14 AM > >> Subject: [Freeswitch-users] Audio quality issues > >> > >> Hello, > >> > >> I was wondering what the maximum concurrent calls for FS > >> before audio quality becomes an issue? I assume the specs > >> of the machine would also affect this. > >> We are currently running FS on a Six Core (12 Threads) > >> Intel E5-2430 CPU and get about 800 concurrent calls at > >> 10-20 CPS. The audio quality at these rates is still > >> fair, but we do notice some quality issue's. > >> Going above these numbers screws up the audio quality: > >> choppy sound, audio drops etc. We aren't doing any heavy > >> media processing, just simply playing a file (G711-Alaw) > >> which lasts about 2 minutes during the load test. > >> These numbers are for one way audio, where Sipp doesn't > >> echo the RTP back. These numbers get lower once Sipp > >> echo's the RTP. > >> > >> I've tried FS on a physical box and also on a virtual box > >> (ESXi 5.1), but the performance gain on physical vs > >> virtual isn't that much. > >> > >> I disabled all the modules we don't need, like CDR's, > >> conferencing, etc. > >> > >> Are there any parameters(config files)/modules that can > >> affect the quality of the audio stream? > >> > >> Regards, > >> > >> Grant > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > >> Server > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > >> Server > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> [cid:~WRD000.jpg] > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel Communication > >> Server > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> [cid:~WRD000.jpg] > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/a3ce1e62/attachment-0001.html From matt at williamsmjw.com Wed Oct 23 02:05:45 2013 From: matt at williamsmjw.com (Matt Williams) Date: Tue, 22 Oct 2013 17:05:45 -0500 Subject: [Freeswitch-users] Limit help In-Reply-To: <0MV20027DW5OHY90@vms173005.mailsrvcs.net> References: <0MV20027DW5OHY90@vms173005.mailsrvcs.net> Message-ID: You can use limit_execute to execute any application you wish. The best way to wrap your head around how simple and useful the limit application is to use is to play with it a little while. Everything should click into place. Thank You, Matthew Williams IKN Network Operations On Tue, Oct 22, 2013 at 11:21 AM, Andre Demattia wrote: > Can I use limit_execute for a outbound gateway in place of the loop back > option? > ------------------------------ > From: Matt Williams > Sent: 10/22/2013 12:01 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Limit help > > You need to set your CPS and port limits separately. You can set limits > per gateway the same way you would limit any application. It is not bound > to any specific application. The values you use are completely arbitrary. > That said here is an example from the Wiki > http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits > > > > > On Tue, Oct 22, 2013 at 10:40 AM, Andre Demattia wrote: > >> Hi, how do I set the CPS and Port limits on a bridge >> >> Can I set both on one application or do I need to set 2 limits? >> >> 5 ports >> >> >> 5 cps >> >> >> Can I do this for both 5 ports and 5 cps? >> >> >> >> Also how do I do the same on a gateway? I need to make sure I don't send >> too many calls to the provider. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/6f49a877/attachment.html From mytemike72 at gmail.com Wed Oct 23 02:15:17 2013 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 23 Oct 2013 00:15:17 +0200 Subject: [Freeswitch-users] Controlling the conference via event socket In-Reply-To: <4420468540895867213@unknownmsgid> References: <4420468540895867213@unknownmsgid> Message-ID: Peter, Thanks for your reply, of course I know ESL can call all API commands :)) Unfortunately I was little too naive to only check the wiki and not the help dump of fs_cli itself. Too bad nothing was mentioned about controlling the conference in wiki as appears the options available are very rich! Anyway, thanks for your response and pointing me to the (obvious) right direction! Regards, Mike 2013/10/22 Peter Olsson > Using ESL gives you access to all API commands. Look into the API > commands for conference, and you should find you answer there. > > /Peter > > > 22 okt 2013 kl. 17:55 skrev Michael Lutz : > > > > Hi, > > > > Is there anybody with experience or knows a way of controlling a session > in a conference via event socket? > > > > I would like to to be able to make a session 'leave' a conference, and > set the 'mute' setting via the ESL. > > > > I did not find any documentation on this, other than being able to > subscribe and read the maintenance events. > > > > Anyone got idea's on this one? > > > > Regards, > > Mike. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/ef6b9629/attachment.html From alipey at gmail.com Wed Oct 23 02:20:20 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 22 Oct 2013 18:20:20 -0400 Subject: [Freeswitch-users] Voicemail notification In-Reply-To: References: Message-ID: Hello Steve, Thank you for the response. My problem is that I can't find the file name in anyway. I use api_hangup_hook and can see all the relevant information but don't know how to find the filename. I am also able to set custom variables to the channel so I can see them in api_hangup_hook, but I can't find the voicemail filename anywhere. Regards, Ali Pey On Tue, Oct 22, 2013 at 4:07 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Mon, Oct 21, 2013 at 2:21 PM, Ali Pey wrote: > >> I need to send a customized voicemail notification based on certain >> condition once a voicemail is recorded. >> > > I did this externally to FreeSWITCH by sending the vm-notify to a perl > script. > > We run FreeSWITCH on CentOS, so it was easy to set up an email address > that I called "vm-page". For this notification, I didn't need the audio > file, so I used vm-notify instead of vm-mailto. In my directory file, I > used this: > > > > > In /etc/aliases, I have this: > > # Paging notify from phone system > vm-page: "|/etc/smrsh/vm-page" > > And /etc/smrsh/vm-page is a perl script that parses the message, looking > for the 'From' line (which contains the caller-ID info), the 'Return-Path' > header (which contains the voice mailbox number), the 'X-Priority' line > (which can identify an urgent message), and the 'X-Voicemail-Length' header. > > In our system, this script parses this stuff, reformats it, and then sends > it through an email-to-SMS gateway at vtext.com. > > -- > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/c0aa0521/attachment.html From ssinyagin at yahoo.com Wed Oct 23 02:44:18 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 22 Oct 2013 15:44:18 -0700 (PDT) Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: <1382481858.78813.YahooMailNeo@web126201.mail.ne1.yahoo.com> "delay_echo" is even more convenient for testing: then you can clearly hear your own voice without distortions from echo cancellation. ________________________________ From: Grant Bagdasarian To: FreeSWITCH Users Help Sent: Tuesday, October 22, 2013 4:03 PM Subject: Re: [Freeswitch-users] Audio quality issues Yes, it does! ? I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo ? Echo application for FS. Going to see if that works first. If not, I?ll setup a Sipp in server mode. ? From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues ? You can make calls from sipp that also terminates on sipp and then play a wave file in sipp. ? Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls. ? Does this make sense? ? On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests. ? How do I simulate two way audio? I know I can make Sipp send an RTP stream using a pcap file, but how do I make FS sent RTP back which is not read from disk? Does FS have an echo application? Or is it enough for Sipp to send the media? ? From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues ? I think the problem here is that you are playing a file for every call for the duration of the call. The bottleneck seems to be the disk access. If there were to be two way audio path, FS would only proxy the media which would be quite faster as there would be no file reading and playing involved. Attempt a test case with fewer or no file play and only media proxy and test again. ? ? On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian wrote: The network shouldn?t be an issue, since we have at least 1Gbps lines. The tests stay within the network. ? I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. ? At this point both FS machines are virtualized, since the performance gain wasn?t that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. ? Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. ? Also, FS is running on Ubuntu Server 12.04 x64. ? From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues ? 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load?? "htop" command would be helpful in this. ? ? ________________________________ From:Grant Bagdasarian To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues ? Hello, ? I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue?s. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren?t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn?t echo the RTP back. These numbers get lower once Sipp echo?s the RTP. ? I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn?t that much. ? I disabled all the modules we don?t need, like CDR?s, conferencing, etc. ? Are there any parameters(config files)/modules that can affect the quality of the audio stream? ? Regards, ? Grant ? ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/e4457961/attachment-0001.html From hardyanto.donny at gmail.com Wed Oct 23 02:58:10 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Wed, 23 Oct 2013 05:58:10 +0700 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: Actually G711 will consumes abaout 167kbps for a call (64 kbps for each side, and the other is IP header, RTP header, Ethernet header) so 800 calls of this will be 133.8 Mbps. If your line is 100 Mbps then you have saturated the line. Donny ---------- Forwarded message ---------- From: "Stanislav Sinyagin" Date: Oct 22, 2013 4:12 PM Subject: Re: [Freeswitch-users] Audio quality issues To: "FreeSWITCH Users Help" Cc: 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ------------------------------ *From:* Grant Bagdasarian *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" < freeswitch-users at lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue?s. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren?t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn?t echo the RTP back. These numbers get lower once Sipp echo?s the RTP. I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn?t that much. I disabled all the modules we don?t need, like CDR?s, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/0a68356f/attachment.html From andretodd at verizon.net Wed Oct 23 03:09:52 2013 From: andretodd at verizon.net (Andre) Date: Tue, 22 Oct 2013 19:09:52 -0400 Subject: [Freeswitch-users] Limit help In-Reply-To: References: <0MV20027DW5OHY90@vms173005.mailsrvcs.net> Message-ID: <026701cecf7b$d1225be0$736713a0$@verizon.net> Ok, I think I got it. Can you verify I got it :) I can use this code to limit my gateways And this code for my customers 5 ports 5 cps How do I add CPS to the gateways using the above example? Is this right? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matt Williams Sent: Tuesday, October 22, 2013 6:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Limit help You can use limit_execute to execute any application you wish. The best way to wrap your head around how simple and useful the limit application is to use is to play with it a little while. Everything should click into place. Thank You, Matthew Williams IKN Network Operations On Tue, Oct 22, 2013 at 11:21 AM, Andre Demattia > wrote: Can I use limit_execute for a outbound gateway in place of the loop back option? _____ From: Matt Williams Sent: 10/22/2013 12:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Limit help You need to set your CPS and port limits separately. You can set limits per gateway the same way you would limit any application. It is not bound to any specific application. The values you use are completely arbitrary. That said here is an example from the Wiki http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-us er_channel_limits On Tue, Oct 22, 2013 at 10:40 AM, Andre Demattia > wrote: Hi, how do I set the CPS and Port limits on a bridge Can I set both on one application or do I need to set 2 limits? 5 ports 5 cps Can I do this for both 5 ports and 5 cps? Also how do I do the same on a gateway? I need to make sure I don't send too many calls to the provider. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/b6fd85fe/attachment-0001.html From anthony.minessale at gmail.com Wed Oct 23 03:11:39 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Oct 2013 18:11:39 -0500 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: That was a memorable milestone in FS development history the first time we broke that barrier and got confused why it was not working.... =D Before that it was unheard of to saturate 100mbps with existing voip software..... On Tue, Oct 22, 2013 at 5:58 PM, Donny Hardyanto wrote: > Actually G711 will consumes abaout 167kbps for a call (64 kbps for each > side, and the other is IP header, RTP header, Ethernet header) so 800 calls > of this will be 133.8 Mbps. If your line is 100 Mbps then you have > saturated the line. > > Donny > > ---------- Forwarded message ---------- > From: "Stanislav Sinyagin" > Date: Oct 22, 2013 4:12 PM > Subject: Re: [Freeswitch-users] Audio quality issues > To: "FreeSWITCH Users Help" > Cc: > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this. > > > > > ------------------------------ > *From:* Grant Bagdasarian > *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" < > freeswitch-users at lists.freeswitch.org> > *Sent:* Tuesday, October 22, 2013 10:14 AM > *Subject:* [Freeswitch-users] Audio quality issues > > Hello, > > I was wondering what the maximum concurrent calls for FS before audio > quality becomes an issue? I assume the specs of the machine would also > affect this. > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU > and get about 800 concurrent calls at 10-20 CPS. The audio quality at these > rates is still fair, but we do notice some quality issue?s. > Going above these numbers screws up the audio quality: choppy sound, audio > drops etc. We aren?t doing any heavy media processing, just simply playing > a file (G711-Alaw) which lasts about 2 minutes during the load test. > These numbers are for one way audio, where Sipp doesn?t echo the RTP back. > These numbers get lower once Sipp echo?s the RTP. > > I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but > the performance gain on physical vs virtual isn?t that much. > > I disabled all the modules we don?t need, like CDR?s, conferencing, etc. > > Are there any parameters(config files)/modules that can affect the quality > of the audio stream? > > Regards, > > Grant > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/e7d50eb7/attachment.html From anthony.minessale at gmail.com Wed Oct 23 03:38:31 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Oct 2013 18:38:31 -0500 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling In-Reply-To: <526614D6.3020401@quietfountain.com> References: <526614D6.3020401@quietfountain.com> Message-ID: Did you change all the fields in the new profile you duplicated that were relevant to the name like name... I usually cp internal.xml new.xml then edit new.xml and global replace internal with new right off the bat. You might find your mistake faster if you backup and revert to default sip profiles from sample and slowly make changes again. On Oct 22, 2013 1:04 AM, "hcoin" wrote: > > This has been a really frustrating problem, I'm sure the answer is > simple but I just can't see it. > > I had several extensions registered to the internal profile, sending > calls out the external profile to a sip-pstn gateway, all seemed fine. > > Then created another internal profile, using a different sip port on the > same lan address, because of 'no device left behind' and NAT issues. > > All seemed well, all the phones register normally. Looking at the > databases in FS they all show the proper ports, the proper domains, etc. > > However, every single call gets picked up as a new call via > sophia/external/... and it hits the public dialplan normally -- except > that's the wrong plan, it should hit the default plan and be identified > as sofia/internal/.... and so forth. > 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel > sofia/external/hcoin at pbx.foobar.com [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] > > What could possibly be the reason phones registered on the internal > profile have their new calls identified as sophia/external and don't hit > the correct plan? Both the phones and the freeswitch are on the same > subnet. This should be so vanilla. What am I missing? > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/373bc5f8/attachment.html From mishehu at freeswitch.org Wed Oct 23 04:40:45 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Tue, 22 Oct 2013 19:40:45 -0500 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> Message-ID: <52671B0D.70604@freeswitch.org> Your network may be rated for 1 gbps, but that may assume packets at or near the MTU size, and that MTU used in the rating might be around 9000 bytes. RTP packets are typically around 200 or less bytes each, sent every ptime (default is 20ms). You may saturate your network infrastructure long before you saturate your FreeSWITCH. -Yossi On 10/22/2013 05:01 AM, Grant Bagdasarian wrote: > > The network shouldn't be an issue, since we have at least 1Gbps lines. > The tests stay within the network. > > I forgot to mention the calls are being distributed across two > machines by a Kamailio instance. > > So for a total of 800 concurrent calls generated by Sipp, each machine > has 400 active calls. > > CPU load reaches about 70% per machine. > > At this point both FS machines are virtualized, since the performance > gain wasn't that much compared to physical. > > The VM host shows it is using ~3/4 of its CPU resources. > > Htop shows that the normal priority threads(green) and the kernel > threads(red) are about the same length. > > Also, FS is running on Ubuntu Server 12.04 x64. > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Stanislav Sinyagin > *Sent:* Tuesday, October 22, 2013 11:11 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Audio quality issues > > 800 calls at 64kbps is 51Mbps. > Could there be a network issue, like a 100Mbps line between the endpoints? > > How heavy is your CPU load? "htop" command would be helpful in this. > > ------------------------------------------------------------------------ > > *From:*Grant Bagdasarian > > *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org > )" > > > *Sent:* Tuesday, October 22, 2013 10:14 AM > *Subject:* [Freeswitch-users] Audio quality issues > > Hello, > > I was wondering what the maximum concurrent calls for FS before audio > quality becomes an issue? I assume the specs of the machine would also > affect this. > > We are currently running FS on a Six Core (12 Threads) Intel E5-2430 > CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality > at these rates is still fair, but we do notice some quality issue's. > > Going above these numbers screws up the audio quality: choppy sound, > audio drops etc. We aren't doing any heavy media processing, just > simply playing a file (G711-Alaw) which lasts about 2 minutes during > the load test. > > These numbers are for one way audio, where Sipp doesn't echo the RTP > back. These numbers get lower once Sipp echo's the RTP. > > I've tried FS on a physical box and also on a virtual box (ESXi 5.1), > but the performance gain on physical vs virtual isn't that much. > > I disabled all the modules we don't need, like CDR's, conferencing, etc. > > Are there any parameters(config files)/modules that can affect the > quality of the audio stream? > > Regards, > > Grant > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/da99af0f/attachment-0001.html From hardyanto.donny at gmail.com Wed Oct 23 04:55:49 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Wed, 23 Oct 2013 07:55:49 +0700 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling In-Reply-To: <52667EBB.2010204@quietfountain.com> References: <526614D6.3020401@quietfountain.com> <52667EBB.2010204@quietfountain.com> Message-ID: I think that the culprit if this internal.xml Donny On Oct 22, 2013 8:36 PM, "hcoin" wrote: > Lan. Lan registered phones can't even get a tone stream. Phones > register promptly, everything seems in order. It's just that every call > coming in on the lan interface and 'internal' port gets identified as > 'sofia/external/ and so never hits the default dial plan. Here's a bit of > the internal profile. The site has some vpns so anything in the rfc1918 > space has no nat. Shouldn't matter anyway as all extensions on the same > lan subnet as the fs box get identified as 'external' as well. > > No , and parse="false"/> > > > > > > > > > > > > > > > > > > > > > value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > The external profile has alias=false in the domains, no aliases and > parse=true > > > > On 10/22/2013 06:13 AM, Donny Hardyanto wrote: > > Is your client in the internet or the lan? > > Donny > > > On Tue, Oct 22, 2013 at 1:01 PM, hcoin wrote: > >> >> This has been a really frustrating problem, I'm sure the answer is >> simple but I just can't see it. >> >> I had several extensions registered to the internal profile, sending >> calls out the external profile to a sip-pstn gateway, all seemed fine. >> >> Then created another internal profile, using a different sip port on the >> same lan address, because of 'no device left behind' and NAT issues.. >> >> All seemed well, all the phones register normally. Looking at the >> databases in FS they all show the proper ports, the proper domains, etc. >> >> However, every single call gets picked up as a new call via >> sophia/external/... and it hits the public dialplan normally -- except >> that's the wrong plan, it should hit the default plan and be identified >> as sofia/internal/.... and so forth. >> 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel >> sofia/external/hcoin at pbx.foobar.com[28ed125a-3adb-11e3-9cc1-cbb8efb09b83] >> >> What could possibly be the reason phones registered on the internal >> profile have their new calls identified as sophia/external and don't hit >> the correct plan? Both the phones and the freeswitch are on the same >> subnet. This should be so vanilla. What am I missing? >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/1e083869/attachment.html From neeraj.p at directi.com Tue Oct 22 18:23:43 2013 From: neeraj.p at directi.com (neeraj.p) Date: Tue, 22 Oct 2013 19:53:43 +0530 Subject: [Freeswitch-users] codec transcodation not working Message-ID: Here is the full log when leg A calls leg B http://pastebin.com/0JAZQpe7 Here is the log when leg B calls leg A http://pastebin.com/8Q5abKEC I can see different errors in these two cases . Regards, Neeraj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131022/4f930ccf/attachment.html From hardyanto.donny at gmail.com Wed Oct 23 05:21:07 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Wed, 23 Oct 2013 08:21:07 +0700 Subject: [Freeswitch-users] codec transcodation not working In-Reply-To: References: Message-ID: >From what I am understand that in freeswitch leg A dont call leg B. It bridges them. So it suppose like this client 1 -(leg a)-> FS ---(bridge)-- FS -(leg b)-> client 2. So when client 1 call FS, the codec from FS dan client 1 must match (speex and telephony event). This will be leg A. then FS call client 2, the codec from FS and client 2 must match (OPUS for example and telephony event). This will be leg B. Then you bridge them leg A to leg B. Then you have transcoding. May be if you can show the corresponding dialplan we will know better. Donny On Tue, Oct 22, 2013 at 9:23 PM, neeraj.p wrote: > Here is the full log when leg A calls leg B > http://pastebin.com/0JAZQpe7 > > Here is the log when leg B calls leg A > http://pastebin.com/8Q5abKEC > > I can see different errors in these two cases . > > Regards, > Neeraj > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/36acba9f/attachment-0001.html From fvillarroel at yahoo.com Wed Oct 23 07:19:09 2013 From: fvillarroel at yahoo.com (Fernando Villarroel) Date: Wed, 23 Oct 2013 00:19:09 -0300 Subject: [Freeswitch-users] [OT] 4kconference Message-ID: Dear All. Sorry for off topic. Does anyone know what happen with 4kconference Chile 2013? Regards Enviado desde mi iPhone From xmppser at gmail.com Wed Oct 23 08:17:05 2013 From: xmppser at gmail.com (xmppser) Date: Tue, 22 Oct 2013 21:17:05 -0700 (PDT) Subject: [Freeswitch-users] FreeTDM tones.conf Disconnect Supervision In-Reply-To: References: <1382436346467-7595879.post@n2.nabble.com> Message-ID: <1382501825929-7595923.post@n2.nabble.com> Hi, thanks, so, how did you resolve this problem? can you give more detail? thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeTDM-tones-conf-Disconnect-Supervision-tp7595763p7595923.html Sent from the freeswitch-users mailing list archive at Nabble.com. From hcoin at quietfountain.com Wed Oct 23 10:14:34 2013 From: hcoin at quietfountain.com (hcoin) Date: Wed, 23 Oct 2013 01:14:34 -0500 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling References: <526614D6.3020401@quietfountain.com> Message-ID: <5267694A.7090107@quietfountain.com> Anthony and Donny, thanks for replying. Putting a packet capture on the line revealed the problem to be a combination of quirks in both linphone (windows version ignores fs nonstandard destination port) and dns-forwarder (override of foo.bar.com fails if foo.bar.com is a cname on the public internet, not an A record). The call was coming in on the external profile because the dns forwarder was letting the resolution go to the public internet and so the local systems were sending out to the router, which sent it back in to... the external interface. However, I do now know how to watch calls pass through freeswitch and have read most of the source code in the sofia endpoint, nta, nua, etc. etc... and had lots of fun with gdb stepping around watching the packets flow. The main lession I think is worth sharing is this: Use 5060 for sip. If you are thinking of various profiles using the same address but different ports on the one hand, or on the other hand using ip aliases so each profile uses the 'standard' ports but a different ip--- go with the ip alias approach. So in /etc/network/interfaces , supposing your main nic is eth0: iface eth0 inet dhcp <-- or whatnot on your system> .. post-up ifup eth0:1 pre-down ifdown eth0:1 .. iface eth0:1 inet static address netmask Problems all melted away as if they never were. Thanks again for trying to help! I even bought the freeswitch book. Ka-Ching for someone on this list... On 10/22/2013 06:38 PM, Anthony Minessale wrote: > > Did you change all the fields in the new profile you duplicated that > were relevant to the name like name... > > I usually cp internal.xml new.xml then edit new.xml and global replace > internal with new right off the bat. > > You might find your mistake faster if you backup and revert to default > sip profiles from sample and slowly make changes again. > > On Oct 22, 2013 1:04 AM, "hcoin" > wrote: > > > This has been a really frustrating problem, I'm sure the answer is > simple but I just can't see it. > > I had several extensions registered to the internal profile, sending > calls out the external profile to a sip-pstn gateway, all seemed fine. > > Then created another internal profile, using a different sip port > on the > same lan address, because of 'no device left behind' and NAT issues.. > > All seemed well, all the phones register normally. Looking at the > databases in FS they all show the proper ports, the proper > domains, etc. > > However, every single call gets picked up as a new call via > sophia/external/... and it hits the public dialplan normally -- except > that's the wrong plan, it should hit the default plan and be > identified > as sofia/internal/.... and so forth. > 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel > sofia/external/hcoin at pbx.foobar.com > [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] > > What could possibly be the reason phones registered on the internal > profile have their new calls identified as sophia/external and > don't hit > the correct plan? Both the phones and the freeswitch are on the same > subnet. This should be so vanilla. What am I missing? > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www..freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/90455adf/attachment.html From hardyanto.donny at gmail.com Wed Oct 23 10:42:53 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Wed, 23 Oct 2013 13:42:53 +0700 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling In-Reply-To: <5267694A.7090107@quietfountain.com> References: <526614D6.3020401@quietfountain.com> <5267694A.7090107@quietfountain.com> Message-ID: I am now practicing not using standard port because some hacks couple month ago. It was quite bad, it cost thousand of dollars and we cannot find the culprit IP address because the router ALG rewrites them and there is no accessible log on the router. Donny On Oct 23, 2013 1:17 PM, "hcoin" wrote: > Anthony and Donny, thanks for replying. > > Putting a packet capture on the line revealed the problem to be a > combination of quirks in both linphone (windows version ignores fs > nonstandard destination port) and dns-forwarder (override of foo.bar.comfails if > foo.bar.com is a cname on the public internet, not an A record). The > call was coming in on the external profile because the dns forwarder was > letting the resolution go to the public internet and so the local systems > were sending out to the router, which sent it back in to... the external > interface. However, I do now know how to watch calls pass through > freeswitch and have read most of the source code in the sofia endpoint, > nta, nua, etc. etc... and had lots of fun with gdb stepping around > watching the packets flow. > > The main lession I think is worth sharing is this: Use 5060 for sip. If > you are thinking of various profiles using the same address but different > ports on the one hand, or on the other hand using ip aliases so each > profile uses the 'standard' ports but a different ip--- go with the ip > alias approach. So in /etc/network/interfaces , supposing your main nic > is eth0: > > iface eth0 inet dhcp <-- or whatnot on your system> > .. > post-up ifup eth0:1 > pre-down ifdown eth0:1 > .. > > > iface eth0:1 inet static > address > netmask > > Problems all melted away as if they never were. > > Thanks again for trying to help! I even bought the freeswitch book. > Ka-Ching for someone on this list... > > > > On 10/22/2013 06:38 PM, Anthony Minessale wrote: > > Did you change all the fields in the new profile you duplicated that were > relevant to the name like name... > > I usually cp internal.xml new.xml then edit new.xml and global replace > internal with new right off the bat. > > You might find your mistake faster if you backup and revert to default sip > profiles from sample and slowly make changes again. > On Oct 22, 2013 1:04 AM, "hcoin" wrote: > >> >> This has been a really frustrating problem, I'm sure the answer is >> simple but I just can't see it. >> >> I had several extensions registered to the internal profile, sending >> calls out the external profile to a sip-pstn gateway, all seemed fine. >> >> Then created another internal profile, using a different sip port on the >> same lan address, because of 'no device left behind' and NAT issues.. >> >> All seemed well, all the phones register normally. Looking at the >> databases in FS they all show the proper ports, the proper domains, etc. >> >> However, every single call gets picked up as a new call via >> sophia/external/... and it hits the public dialplan normally -- except >> that's the wrong plan, it should hit the default plan and be identified >> as sofia/internal/.... and so forth. >> 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel >> sofia/external/hcoin at pbx.foobar.com[28ed125a-3adb-11e3-9cc1-cbb8efb09b83] >> >> What could possibly be the reason phones registered on the internal >> profile have their new calls identified as sophia/external and don't hit >> the correct plan? Both the phones and the freeswitch are on the same >> subnet. This should be so vanilla. What am I missing? >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/ca6cbe5c/attachment-0001.html From peter at olssononline.se Wed Oct 23 10:45:53 2013 From: peter at olssononline.se (Peter Olsson) Date: Wed, 23 Oct 2013 08:45:53 +0200 Subject: [Freeswitch-users] Controlling the conference via event socket In-Reply-To: References: <4420468540895867213@unknownmsgid> Message-ID: <2393052659321050380@unknownmsgid> Mike, It would be great if you could add some more info about this on the Wiki, to help other people in the same situation. Good to hear that this resolved your problem though! /Peter 23 okt 2013 kl. 00:21 skrev Michael Lutz : Peter, Thanks for your reply, of course I know ESL can call all API commands :)) Unfortunately I was little too naive to only check the wiki and not the help dump of fs_cli itself. Too bad nothing was mentioned about controlling the conference in wiki as appears the options available are very rich! Anyway, thanks for your response and pointing me to the (obvious) right direction! Regards, Mike 2013/10/22 Peter Olsson > Using ESL gives you access to all API commands. Look into the API > commands for conference, and you should find you answer there. > > /Peter > > > 22 okt 2013 kl. 17:55 skrev Michael Lutz : > > > > Hi, > > > > Is there anybody with experience or knows a way of controlling a session > in a conference via event socket? > > > > I would like to to be able to make a session 'leave' a conference, and > set the 'mute' setting via the ESL. > > > > I did not find any documentation on this, other than being able to > subscribe and read the maintenance events. > > > > Anyone got idea's on this one? > > > > Regards, > > Mike. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/9c5810bc/attachment.html From harsimran2201 at gmail.com Wed Oct 23 10:21:18 2013 From: harsimran2201 at gmail.com (Harsimran singh) Date: Wed, 23 Oct 2013 11:51:18 +0530 Subject: [Freeswitch-users] Configurable Hold Music while conference in freeswitch Message-ID: Hi, I want play configurable hold sound while conference using freeswitch. I am using "moh-sound" variable in conference.conf.xml as follows. The uuid is the variable "variable_uuid" in the events comes from freeswitch,but the freeswitch is unable to expand this variable and error comes as follows : [ERR] mod_shout.c:683 Error opening /srv/sounds/${uuid}/knowlus_hold_music.mp3 [ERR] mod_shout.c:862 Error from mpg123: File access error. (code 22) Freeswitch is not able to expand the variable as variable exists and the file which i want to play also exists. Can anybody help me out in this ? With Regards Harsimran Singh +91-9711271158 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/30861c5a/attachment.html From miha at softnet.si Wed Oct 23 12:07:36 2013 From: miha at softnet.si (Miha) Date: Wed, 23 Oct 2013 10:07:36 +0200 Subject: [Freeswitch-users] Parsing from sip header Message-ID: <526783C8.7070408@softnet.si> Hi, I am sending to FS some attribute in header which I have added on proxy side. How can I parse this header, to get this data and send it further? I can not see it in varibles which are printed with info application. I added: Moved: 38618108751. BR, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/8e7a3b25/attachment.html From mehroz.ashraf85 at gmail.com Wed Oct 23 12:13:11 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 23 Oct 2013 01:13:11 -0700 (PDT) Subject: [Freeswitch-users] Video switching In-Reply-To: <12DFDB1C-89F1-40B6-B8CD-BF02AFED9EB0@jerris.com> References: <1375267219620-7593439.post@n2.nabble.com> <1375333454405-7593468.post@n2.nabble.com> <1382450191344-7595891.post@n2.nabble.com> <12DFDB1C-89F1-40B6-B8CD-BF02AFED9EB0@jerris.com> Message-ID: <1382515991968-7595930.post@n2.nabble.com> Here you go! http://jira.freeswitch.org/browse/FS-5902 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-switching-tp7593439p7595930.html Sent from the freeswitch-users mailing list archive at Nabble.com. From miha at softnet.si Wed Oct 23 12:33:15 2013 From: miha at softnet.si (Miha) Date: Wed, 23 Oct 2013 10:33:15 +0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: <52667E76.8010904@coppice.org> References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> <52667E76.8010904@coppice.org> Message-ID: <526789CB.80905@softnet.si> Hi Steve, do you have any good preposal which switch has good performance for voip trafic? miha Dne 10/22/2013 3:32 PM, pi?e Steve Underwood: > Hi Grant, > > Two possibilities spring to mind: > > - If your audio is coming from a disk, can that disk keep up? > - How good is your ethernet switch? > > Notice in the second point I said how good, not how expensive. Many > switches choke on a large number of small media packets, including some > expensive big name products. > > Regards, > Steve > > On 10/22/2013 06:01 PM, Grant Bagdasarian wrote: >> The network shouldn?t be an issue, since we have at least 1Gbps lines. >> The tests stay within the network. >> >> I forgot to mention the calls are being distributed across two >> machines by a Kamailio instance. >> >> So for a total of 800 concurrent calls generated by Sipp, each machine >> has 400 active calls. >> >> CPU load reaches about 70% per machine. >> >> At this point both FS machines are virtualized, since the performance >> gain wasn?t that much compared to physical. >> >> The VM host shows it is using ~3/4 of its CPU resources. >> >> Htop shows that the normal priority threads(green) and the kernel >> threads(red) are about the same length. >> >> Also, FS is running on Ubuntu Server 12.04 x64. >> >> *From:*freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >> *Stanislav Sinyagin >> *Sent:* Tuesday, October 22, 2013 11:11 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Audio quality issues >> >> 800 calls at 64kbps is 51Mbps. >> Could there be a network issue, like a 100Mbps line between the endpoints? >> >> How heavy is your CPU load? "htop" command would be helpful in this. >> >> ------------------------------------------------------------------------ >> >> *From:*Grant Bagdasarian > >> *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org >> )" >> > > >> *Sent:* Tuesday, October 22, 2013 10:14 AM >> *Subject:* [Freeswitch-users] Audio quality issues >> >> Hello, >> >> I was wondering what the maximum concurrent calls for FS before audio >> quality becomes an issue? I assume the specs of the machine would also >> affect this. >> >> We are currently running FS on a Six Core (12 Threads) Intel E5-2430 >> CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality >> at these rates is still fair, but we do notice some quality issue?s. >> >> Going above these numbers screws up the audio quality: choppy sound, >> audio drops etc. We aren?t doing any heavy media processing, just >> simply playing a file (G711-Alaw) which lasts about 2 minutes during >> the load test. >> >> These numbers are for one way audio, where Sipp doesn?t echo the RTP >> back. These numbers get lower once Sipp echo?s the RTP. >> >> I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), >> but the performance gain on physical vs virtual isn?t that much. >> >> I disabled all the modules we don?t need, like CDR?s, conferencing, etc. >> >> Are there any parameters(config files)/modules that can affect the >> quality of the audio stream? >> >> Regards, >> >> Grant >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From david.villasmil.work at gmail.com Wed Oct 23 13:05:16 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 23 Oct 2013 11:05:16 +0200 Subject: [Freeswitch-users] GIT Down? Message-ID: Hello guys, I'm getting this since last night: root at llwi079:/usr/local/src# git clone git:// git.freeswitch.org/freeswitch.git Cloning into 'freeswitch'... fatal: unable to connect to git.freeswitch.org: git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out root at llwi079:/usr/local/src# git clone git:// git.freeswitch.org/freeswitch.git Cloning into 'freeswitch'... ^@fatal: unable to connect to git.freeswitch.org: git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out Are we down? Thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/c78fd6a9/attachment-0001.html From steveayre at gmail.com Wed Oct 23 13:40:25 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 23 Oct 2013 10:40:25 +0100 Subject: [Freeswitch-users] GIT Down? In-Reply-To: References: Message-ID: Looks ok from here (UK) On 23 October 2013 10:05, David Villasmil wrote: > Hello guys, > > I'm getting this since last night: > > root at llwi079:/usr/local/src# git clone git:// > git.freeswitch.org/freeswitch.git > Cloning into 'freeswitch'... > fatal: unable to connect to git.freeswitch.org: > git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out > > root at llwi079:/usr/local/src# git clone git:// > git.freeswitch.org/freeswitch.git > Cloning into 'freeswitch'... > ^@fatal: unable to connect to git.freeswitch.org: > git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out > > Are we down? > > Thanks! > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/fe53f68f/attachment.html From david.villasmil.work at gmail.com Wed Oct 23 14:28:55 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 23 Oct 2013 12:28:55 +0200 Subject: [Freeswitch-users] GIT Down? In-Reply-To: References: Message-ID: I'm in Spain, not working from here... I had to download the bz2 file... David On Wed, Oct 23, 2013 at 11:40 AM, Steven Ayre wrote: > Looks ok from here (UK) > > > On 23 October 2013 10:05, David Villasmil wrote: > >> Hello guys, >> >> I'm getting this since last night: >> >> root at llwi079:/usr/local/src# git clone git:// >> git.freeswitch.org/freeswitch.git >> Cloning into 'freeswitch'... >> fatal: unable to connect to git.freeswitch.org: >> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out >> >> root at llwi079:/usr/local/src# git clone git:// >> git.freeswitch.org/freeswitch.git >> Cloning into 'freeswitch'... >> ^@fatal: unable to connect to git.freeswitch.org: >> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out >> >> Are we down? >> >> Thanks! >> >> David >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/6807587d/attachment.html From lists at telefaks.de Wed Oct 23 14:32:04 2013 From: lists at telefaks.de (Peter Steinbach) Date: Wed, 23 Oct 2013 12:32:04 +0200 Subject: [Freeswitch-users] loopback_bowout In-Reply-To: <93392EFB-3CC9-4B00-B09A-A7A1449E00B8@freeswitch.org> References: <5266DF4F.2000002@telefaks.de> <93392EFB-3CC9-4B00-B09A-A7A1449E00B8@freeswitch.org> Message-ID: <5267A5A4.80000@telefaks.de> Hello Brian, ok, we will update then, as our rev is older. Best regards Peter On 10/22/13 22:36, Brian West wrote: > What git rev are you on because we just fixed this last week! > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > On Oct 22, 2013, at 3:25 PM, Peter Steinbach wrote: > >> Hello, >> >> I have a strange behaviour with forwarding a call from an external gateway to another number on the same gateway. >> ? when I forward a call from an internal number to an external number via loopback, I firstly get 4 channels. After answering the call, I have 2 channels remaining and audio is fine >> ? when I forward a call from an _external_ number to an external number via loopback, I firstly get 4 channels. After answering the call, I still have 4 channels remaining and audio is _not_ there. Freeswitch is dialling via "external" profile via port 5080. I also tried to set "loopback_bowout=true", but this did not change anything. >> The dialstring is: >> The dialplan for both scenarios is the same. >> Anybody had the same issue and knows how to overcome this? >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> >> mailto:lists >> (att) telefaks.de >> Internet: >> www.telefaks.de >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From shahzad.bhatti at g-r-v.com Wed Oct 23 14:32:57 2013 From: shahzad.bhatti at g-r-v.com (Shahzad_Sab) Date: Wed, 23 Oct 2013 03:32:57 -0700 (PDT) Subject: [Freeswitch-users] Calling stored procedure in freeswitch database or using luasql In-Reply-To: <1369129093912-7590895.post@n2.nabble.com> References: <1369129093912-7590895.post@n2.nabble.com> Message-ID: <1382524377956-7595937.post@n2.nabble.com> i am also trying to do and if u got any idea do reply me too regards shahzad bhatti -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calling-stored-procedure-in-freeswitch-database-or-using-luasql-tp7590895p7595937.html Sent from the freeswitch-users mailing list archive at Nabble.com. From regis.freeswitch.org at tornad.net Wed Oct 23 15:28:35 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Wed, 23 Oct 2013 13:28:35 +0200 Subject: [Freeswitch-users] Problem with password resend on the wiki (for french profile only?) Message-ID: Hi, Can someone fix the wiki retreive password system *? I forgot my password and the system bugs when I tried to get it back.. The error is : No cache directory configured Backtrace: #0 /var/www/extensions/LocalisationUpdate/LocalisationUpdate.class.php(553): LocalisationUpdate::filename('fr') #1 /var/www/extensions/LocalisationUpdate/LocalisationUpdate.class.php(36): LocalisationUpdate::readFile('fr') #2 [internal function]: LocalisationUpdate::onRecache(Object(LocalisationCache), 'fr', Array) #3 /var/www/includes/Hooks.php(255): call_user_func_array('LocalisationUpd...', Array) #4 /var/www/includes/GlobalFunctions.php(3883): Hooks::run('LocalisationCac...', Array) #5 /var/www/includes/cache/LocalisationCache.php(796): wfRunHooks('LocalisationCac...', Array) #6 /var/www/includes/cache/LocalisationCache.php(426): LocalisationCache->recache('fr') #7 /var/www/includes/cache/LocalisationCache.php(310): LocalisationCache->initLanguage('fr') #8 /var/www/includes/cache/LocalisationCache.php(245): LocalisationCache->loadItem('fr', 'fallback') #9 /var/www/languages/Language.php(3978): LocalisationCache->getItem('fr', 'fallback') #10 /var/www/languages/Language.php(230): Language::getFallbacksFor('fr') #11 /var/www/languages/Language.php(189): Language::newFromCode('fr') #12 /var/www/includes/Message.php(381): Language::factory('fr') #13 /var/www/includes/specials/SpecialPasswordReset.php(243): Message->inLanguage('fr') #14 [internal function]: SpecialPasswordReset->onSubmit(Array, Object(HTMLForm)) #15 /var/www/includes/HTMLForm.php(412): call_user_func(Array, Array, Object(HTMLForm)) #16 /var/www/includes/HTMLForm.php(356): HTMLForm->trySubmit() #17 /var/www/includes/HTMLForm.php(371): HTMLForm->tryAuthorizedSubmit() #18 /var/www/includes/SpecialPage.php(996): HTMLForm->show() #19 /var/www/includes/SpecialPage.php(613): FormSpecialPage->execute(NULL) #20 /var/www/includes/SpecialPageFactory.php(487): SpecialPage->run(NULL) #21 /var/www/includes/Wiki.php(291): SpecialPageFactory::executePath(Object(Title), Object(RequestContext)) #22 /var/www/includes/Wiki.php(565): MediaWiki->performRequest() #23 /var/www/includes/Wiki.php(458): MediaWiki->main() #24 /var/www/index.php(59): MediaWiki->run() #25 {main} Maybe because I'm set to french...:) ? ar/www/extensions/LocalisationUpdate/LocalisationUpdate.class.php(553): LocalisationUpdate::*filename('fr') * *Thank you. * * * * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/c5adcdc3/attachment-0001.html From mike at jerris.com Wed Oct 23 15:59:52 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 07:59:52 -0400 Subject: [Freeswitch-users] Configurable Hold Music while conference in freeswitch In-Reply-To: References: Message-ID: the conference param moh_sound is per conference, not per user. It currently will not use channel variables to attempt to expand this variable. It does appear it will use globals if they are set. On Oct 23, 2013, at 2:21 AM, Harsimran singh wrote: > Hi, > I want play configurable hold sound while conference using freeswitch. > I am using "moh-sound" variable in conference.conf.xml as follows. > > > > The uuid is the variable "variable_uuid" in the events comes from freeswitch,but the freeswitch is unable to expand this variable and error comes as follows : > > [ERR] mod_shout.c:683 Error opening /srv/sounds/${uuid}/knowlus_hold_music.mp3 > [ERR] mod_shout.c:862 Error from mpg123: File access error. (code 22) > > Freeswitch is not able to expand the variable as variable exists and the file which i want to play also exists. > > Can anybody help me out in this ? > > With Regards > > Harsimran Singh > +91-9711271158 From lists at telefaks.de Wed Oct 23 16:01:41 2013 From: lists at telefaks.de (Peter Steinbach) Date: Wed, 23 Oct 2013 14:01:41 +0200 Subject: [Freeswitch-users] GIT Down? In-Reply-To: References: Message-ID: <5267BAA5.1040608@telefaks.de> Works from here (Germany) Peter On 10/23/13 12:28, David Villasmil wrote: > I'm in Spain, not working from here... I had to download the bz2 file... > > David > > > On Wed, Oct 23, 2013 at 11:40 AM, Steven Ayre > wrote: > > Looks ok from here (UK) > > > On 23 October 2013 10:05, David Villasmil > > wrote: > > Hello guys, > > I'm getting this since last night: > > root at llwi079:/usr/local/src# git clone > git://git.freeswitch.org/freeswitch.git > > Cloning into 'freeswitch'... > fatal: unable to connect to git.freeswitch.org > : > git.freeswitch.org [0: > 198.22.64.222]: errno=Connection timed out > > root at llwi079:/usr/local/src# git clone > git://git.freeswitch.org/freeswitch.git > > Cloning into 'freeswitch'... > ^@fatal: unable to connect to git.freeswitch.org > : > git.freeswitch.org [0: > 198.22.64.222]: errno=Connection timed out > > Are we down? > > Thanks! > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/8d9aecc5/attachment.html From mike at jerris.com Wed Oct 23 16:04:13 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 08:04:13 -0400 Subject: [Freeswitch-users] Parsing from sip header In-Reply-To: <526783C8.7070408@softnet.si> References: <526783C8.7070408@softnet.si> Message-ID: <1809852E-41E5-4E15-BC30-D36703331C88@jerris.com> I don't think we currently parse the "Moved" Header (I've never actually seen that header before, is it standard?). We do parse x-headers On Oct 23, 2013, at 4:07 AM, Miha wrote: > Hi, > > I am sending to FS some attribute in header which I have added on proxy side. How can I parse this header, to get this data and send it further? > > I can not see it in varibles which are printed with info application. > > I added: > Moved: 38618108751. > From mike at jerris.com Wed Oct 23 16:07:22 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 08:07:22 -0400 Subject: [Freeswitch-users] GIT Down? In-Reply-To: References: Message-ID: <521EC9A1-BE09-4613-84B2-E045618C55D0@jerris.com> Are you able to ping git.freeswitch.org? All services appear to be up and running for me. If you are unable to ping.. check out a traceroute and see where its failing. On Oct 23, 2013, at 6:28 AM, David Villasmil wrote: > I'm in Spain, not working from here... I had to download the bz2 file... > > David > > > On Wed, Oct 23, 2013 at 11:40 AM, Steven Ayre wrote: > Looks ok from here (UK) > > > On 23 October 2013 10:05, David Villasmil wrote: > Hello guys, > > I'm getting this since last night: > > root at llwi079:/usr/local/src# git clone git://git.freeswitch.org/freeswitch.git > Cloning into 'freeswitch'... > fatal: unable to connect to git.freeswitch.org: > git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out > > root at llwi079:/usr/local/src# git clone git://git.freeswitch.org/freeswitch.git > Cloning into 'freeswitch'... > ^@fatal: unable to connect to git.freeswitch.org: > git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out > > Are we down? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/15c9e7ec/attachment-0001.html From harsimran2201 at gmail.com Wed Oct 23 16:41:29 2013 From: harsimran2201 at gmail.com (Harsimran singh) Date: Wed, 23 Oct 2013 18:11:29 +0530 Subject: [Freeswitch-users] Configurable Hold Music while conference in freeswitch In-Reply-To: References: Message-ID: Hi Michael , *Thanks for the reply. * It is agreed that moh-sound variable is per conference not per user. So How can i set the global values for moh-sound so that i can use different moh-sound for each conference ? Thanks With Regards Harsimran Singh +91-9711271158 On Wed, Oct 23, 2013 at 5:29 PM, Michael Jerris wrote: > the conference param moh_sound is per conference, not per user. It > currently will not use channel variables to attempt to expand this > variable. It does appear it will use globals if they are set. > > On Oct 23, 2013, at 2:21 AM, Harsimran singh > wrote: > > > Hi, > > I want play configurable hold sound while conference using freeswitch. > > I am using "moh-sound" variable in conference.conf.xml as follows. > > > > value="/srv/sounds/${uuid}/knowlus_hold_music.mp3"/> > > > > The uuid is the variable "variable_uuid" in the events comes from > freeswitch,but the freeswitch is unable to expand this variable and error > comes as follows : > > > > [ERR] mod_shout.c:683 Error opening > /srv/sounds/${uuid}/knowlus_hold_music.mp3 > > [ERR] mod_shout.c:862 Error from mpg123: File access error. (code 22) > > > > Freeswitch is not able to expand the variable as variable exists and the > file which i want to play also exists. > > > > Can anybody help me out in this ? > > > > With Regards > > > > Harsimran Singh > > +91-9711271158 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/8c8b6c4b/attachment.html From mbodbg at gmx.net Wed Oct 23 17:09:55 2013 From: mbodbg at gmx.net (mbo) Date: Wed, 23 Oct 2013 15:09:55 +0200 Subject: [Freeswitch-users] No ringback tone on bridge Message-ID: Hello, we receive calls from gateway A and bridge them using gateway B to the destination number . The problem is, that we do not hear a ringtone at all. If we put a "ring_ready" before the bridge, we can hear a ringtone immediately, but then the callers hear a ringtone even if the bridged destination is not ringing yet. Is there a solution that the callers hears a ring tone as soon as the destination is ringing, but not before ? Thanks Markus From david.villasmil.work at gmail.com Wed Oct 23 17:17:01 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 23 Oct 2013 15:17:01 +0200 Subject: [Freeswitch-users] GIT Down? In-Reply-To: <521EC9A1-BE09-4613-84B2-E045618C55D0@jerris.com> References: <521EC9A1-BE09-4613-84B2-E045618C55D0@jerris.com> Message-ID: Hello, No ping response: /home/david# ping git.freeswitch.org PING vc-01.ptk.freeswitch.org (198.22.64.222) 56(84) bytes of data. Must be a problem in my network... I can't even get out to freeswitch: /home/david# traceroute git.freeswitch.org -n traceroute to git.freeswitch.org (198.22.64.222), 30 hops max, 60 byte packets 1 1.2.3.4 1.445 ms 1.436 ms 1.449 ms 2 * * * 3 * * * 4 * * * 5 * * * 6 * * * 7 * * * 8 * * * 9 * * * 10 * * * 11 * * * 12 * *^C /home/david# route Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface default 1.2.3.1 0.0.0.0 UG 0 0 0 eth0 localnet * 255.255.255.0 U 0 0 0 eth0 /home/david# On Wed, Oct 23, 2013 at 2:07 PM, Michael Jerris wrote: > Are you able to ping git.freeswitch.org? All services appear to be up > and running for me. If you are unable to ping.. check out a traceroute and > see where its failing. > > > On Oct 23, 2013, at 6:28 AM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > I'm in Spain, not working from here... I had to download the bz2 file... > > David > > > On Wed, Oct 23, 2013 at 11:40 AM, Steven Ayre wrote: > >> Looks ok from here (UK) >> >> >> On 23 October 2013 10:05, David Villasmil > > wrote: >> >>> Hello guys, >>> >>> I'm getting this since last night: >>> >>> root at llwi079:/usr/local/src# git clone git:// >>> git.freeswitch.org/freeswitch.git >>> Cloning into 'freeswitch'... >>> fatal: unable to connect to git.freeswitch.org: >>> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out >>> >>> root at llwi079:/usr/local/src# git clone git:// >>> git.freeswitch.org/freeswitch.git >>> Cloning into 'freeswitch'... >>> ^@fatal: unable to connect to git.freeswitch.org: >>> git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out >>> >>> Are we down? >>> >>> Thanks! >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/fae67e0f/attachment.html From david.villasmil.work at gmail.com Wed Oct 23 17:19:32 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 23 Oct 2013 15:19:32 +0200 Subject: [Freeswitch-users] How to configure a Gateway that registers on my FS Message-ID: Hello, There are many examples for gateways on which FS registers... but how about the other way around? I have a gateway that needs to register on my FS... I know I can configure it as a client, but then how do I use it as a gateway? Thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/8bceb404/attachment.html From alex at digitalmail.com Wed Oct 23 17:25:07 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 23 Oct 2013 14:25:07 +0100 Subject: [Freeswitch-users] Forwarding SIP sessions Message-ID: <5267CE33.3030700@digitalmail.com> Is it possible to get Freeswitch to send a re-invite message in order to do the equivalent of "I'm not accepting this call, try xx.xx.xx.xx"? From david.villasmil.work at gmail.com Wed Oct 23 17:40:05 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 23 Oct 2013 15:40:05 +0200 Subject: [Freeswitch-users] Calling stored procedure in freeswitch database or using luasql In-Reply-To: <1382524377956-7595937.post@n2.nabble.com> References: <1369129093912-7590895.post@n2.nabble.com> <1382524377956-7595937.post@n2.nabble.com> Message-ID: Did you try calling it normally? like: assert( conn:query("call givme_my_data();", function(qrow) for key, val in pairs(qrow) do myrow[key] = val found = 1 end end ) ) On Wed, Oct 23, 2013 at 12:32 PM, Shahzad_Sab wrote: > i am also trying to do and if u got any idea do reply me too > > regards > > shahzad bhatti > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Calling-stored-procedure-in-freeswitch-database-or-using-luasql-tp7590895p7595937.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/bfb5d2c0/attachment-0001.html From jleung at v10networks.ca Wed Oct 23 17:41:27 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 23 Oct 2013 06:41:27 -0700 Subject: [Freeswitch-users] GIT Down? Message-ID: <56r9ryvlmlda98cp82ewbiju.1382535609752@email.android.com> Try github if all else fails. There's a mirror there under the official freeswitch account.? -------- Original message -------- From: David Villasmil Date: 10-23-2013 6:18 AM (GMT-08:00) To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] GIT Down? Hello, No ping response: /home/david# ping git.freeswitch.org PING vc-01.ptk.freeswitch.org (198.22.64.222) 56(84) bytes of data. Must be a problem in my network... I can't even get out to freeswitch: /home/david# traceroute git.freeswitch.org -n traceroute to git.freeswitch.org (198.22.64.222), 30 hops max, 60 byte packets ?1 ?1.2.3.4 ?1.445 ms ?1.436 ms ?1.449 ms ?2 ?* * * ?3 ?* * * ?4 ?* * * ?5 ?* * * ?6 ?* * * ?7 ?* * * ?8 ?* * * ?9 ?* * * 10 ?* * * 11 ?* * * 12 ?* *^C /home/david# route Kernel IP routing table Destination ? ? Gateway ? ? ? ? Genmask ? ? ? ? Flags Metric Ref ? ?Use Iface default ? ? ? ? 1.2.3.1 ? ?0.0.0.0 ? ? ? ? UG ? ?0 ? ? ?0 ? ? ? ?0 eth0 localnet ? ? ? ?* ? ? ? ? ? ? ? 255.255.255.0 ? U ? ? 0 ? ? ?0 ? ? ? ?0 eth0 /home/david# On Wed, Oct 23, 2013 at 2:07 PM, Michael Jerris wrote: Are you able to ping git.freeswitch.org? ?All services appear to be up and running for me. ?If you are unable to ping.. check out a traceroute and see where its failing. On Oct 23, 2013, at 6:28 AM, David Villasmil wrote: I'm in Spain, not working from here... I had to download the bz2 file... David On Wed, Oct 23, 2013 at 11:40 AM, Steven Ayre wrote: Looks ok from here (UK) On 23 October 2013 10:05, David Villasmil wrote: Hello guys, I'm getting this since last night: root at llwi079:/usr/local/src# git clone git://git.freeswitch.org/freeswitch.git Cloning into 'freeswitch'... fatal: unable to connect to git.freeswitch.org: git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out root at llwi079:/usr/local/src# git clone git://git.freeswitch.org/freeswitch.git Cloning into 'freeswitch'... ^@fatal: unable to connect to git.freeswitch.org: git.freeswitch.org[0: 198.22.64.222]: errno=Connection timed out Are we down? Thanks! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/feac192e/attachment.html From regis.freeswitch.org at tornad.net Wed Oct 23 17:51:43 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Wed, 23 Oct 2013 15:51:43 +0200 Subject: [Freeswitch-users] Forwarding SIP sessions In-Reply-To: <5267CE33.3030700@digitalmail.com> References: <5267CE33.3030700@digitalmail.com> Message-ID: +1 I'm very interested by this feature. 2013/10/23 Alex Lake > Is it possible to get Freeswitch to send a re-invite message in order to > do the equivalent of "I'm not accepting this call, try xx.xx.xx.xx"? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/effe6703/attachment.html From mike at jerris.com Wed Oct 23 17:52:21 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 09:52:21 -0400 Subject: [Freeswitch-users] Configurable Hold Music while conference in freeswitch In-Reply-To: References: Message-ID: <5051ED65-0D0E-4CBB-AD90-3CDDDEB45F73@jerris.com> you can't use a variable to do that, you can make different conference profiles for each one you want and references the right profile in the dialplan. On Oct 23, 2013, at 8:41 AM, Harsimran singh wrote: > Hi Michael , > Thanks for the reply. > It is agreed that moh-sound variable is per conference not per user. > So How can i set the global values for moh-sound so that i can use different moh-sound for each conference ? > > Thanks > > With Regards > > Harsimran Singh > +91-9711271158 > > > On Wed, Oct 23, 2013 at 5:29 PM, Michael Jerris wrote: > the conference param moh_sound is per conference, not per user. It currently will not use channel variables to attempt to expand this variable. It does appear it will use globals if they are set. > > On Oct 23, 2013, at 2:21 AM, Harsimran singh wrote: > > > Hi, > > I want play configurable hold sound while conference using freeswitch. > > I am using "moh-sound" variable in conference.conf.xml as follows. > > > > > > > > The uuid is the variable "variable_uuid" in the events comes from freeswitch,but the freeswitch is unable to expand this variable and error comes as follows : > > > > [ERR] mod_shout.c:683 Error opening /srv/sounds/${uuid}/knowlus_hold_music.mp3 > > [ERR] mod_shout.c:862 Error from mpg123: File access error. (code 22) > > > > Freeswitch is not able to expand the variable as variable exists and the file which i want to play also exists. > > > > Can anybody help me out in this ? > > > > With Regards > > > > Harsimran Singh > > +91-9711271158 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/6fc713e6/attachment.html From mike at jerris.com Wed Oct 23 17:55:26 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 09:55:26 -0400 Subject: [Freeswitch-users] Forwarding SIP sessions In-Reply-To: <5267CE33.3030700@digitalmail.com> References: <5267CE33.3030700@digitalmail.com> Message-ID: <9A49B57C-56AE-4C80-8C1A-0E02DF84FE91@jerris.com> that would be a 302 redirect not an re-invite typically: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect if the call is already answered, you can https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect On Oct 23, 2013, at 9:25 AM, Alex Lake wrote: > Is it possible to get Freeswitch to send a re-invite message in order to > do the equivalent of "I'm not accepting this call, try xx.xx.xx.xx"? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/b38d46f0/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Oct 23 17:59:04 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 23 Oct 2013 14:59:04 +0100 Subject: [Freeswitch-users] Forwarding SIP sessions In-Reply-To: References: <5267CE33.3030700@digitalmail.com> Message-ID: You mean like this? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect On 23 October 2013 14:51, Regis M wrote: > +1 > I'm very interested by this feature. > > > > 2013/10/23 Alex Lake > >> Is it possible to get Freeswitch to send a re-invite message in order to >> do the equivalent of "I'm not accepting this call, try xx.xx.xx.xx"? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/b4c9f94e/attachment.html From regis.freeswitch.org at tornad.net Wed Oct 23 18:00:36 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Wed, 23 Oct 2013 16:00:36 +0200 Subject: [Freeswitch-users] How to configure a Gateway that registers on my FS In-Reply-To: References: Message-ID: Hum... For me it's the same things, once your gateway is register on FS (as a normal directory/user module like a phone), you have to call the user to route call throw your gateway. By bridging on "user/gateway_name" Then, you can use ^ to specify an extension on your gateway side. I don't find the associate wiki page, sorry. Another solution is to open your gateway for public access for you freeswitch and send call directly by IP to sofia/external/@: like here : https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes Regards, 2013/10/23 David Villasmil > Hello, > > There are many examples for gateways on which FS registers... but how > about the other way around? > > I have a gateway that needs to register on my FS... > I know I can configure it as a client, but then how do I use it as a > gateway? > > Thanks > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/c5d58018/attachment.html From yehavi.bourvine at gmail.com Wed Oct 23 18:03:51 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 23 Oct 2013 17:03:51 +0300 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: HI, A question: Is it solved on 1.2.14 or only on HEAD? I tried today 1.2.14 and failed with the same sympthoms... Thanks, __Yehavi: 2013/10/14 Anthony Minessale > It is for many. It depends on the general goals etc. > The problem you describe was only on stable and all the symptoms, not > shutting down, channels count being wrong, zombies etc are all the same 1 > problem now fixed. > > > > On Mon, Oct 14, 2013 at 11:33 AM, Stephen Wilde wrote: > >> I have tried only branch v1.2.stable not master. Is the master good for a >> production environment? >> >> >> On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> That's all the same thing...... >>> >>> Do you have this problem on master branch as well? >>> >>> >>> >>> On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde wrote: >>> >>>> Thank you for your advice. >>>> I have tried the 1.2.stable but I have due to abandon this branch due >>>> to a some "zombie sessions" in FreeSwitch visible also during shutdown with >>>> a "Waiting x sessions ..." messages. >>>> I have seen that this issue (FS-5848) has already been solved so I can >>>> try with 1.2.stable but I don't see any fix that can affect my issue. >>>> >>>> >>>> On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> Try latest HEAD for a probable fix. >>>>> >>>>> >>>>> >>>>> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde wrote: >>>>> >>>>>> Hi all, >>>>>> we have an issue with our FreeSwitch box that to us seems to be >>>>>> performance related. >>>>>> The effect is that in FreeSwitch the number of session reaches the >>>>>> limit we set in config as: >>>>>> >>>>>> >>>>>> >>>>>> The limit is reached independently of its value because when >>>>>> FreeSwitch is in this state, the number of sessions grows indefinitely. >>>>>> >>>>>> We have tried to upgrade the hardware of the box moving from a Xeon 2 >>>>>> CPU E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but >>>>>> with this more powerful hardware it happens that the limit is reached with >>>>>> less sessions. >>>>>> It seems that performance are related to the speed of single core >>>>>> instead of speed of the box. >>>>>> Make sense? >>>>>> >>>>>> A example of "status" issued before the crash is: >>>>>> >>>>>> 14969852 session(s) since startup >>>>>> 13765 session(s) - 538 out of max 1000 per sec >>>>>> 30000 session(s) max >>>>>> min idle cpu 0.00/36.00 >>>>>> Current Stack Size/Max 240K/8192K >>>>>> >>>>>> >>>>>> Any advice? >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/71534455/attachment-0001.html From neeraj.p at directi.com Tue Oct 22 18:32:16 2013 From: neeraj.p at directi.com (neeraj.p) Date: Tue, 22 Oct 2013 20:02:16 +0530 Subject: [Freeswitch-users] codec transcodation not working Message-ID: <32DF6D7A-A5CE-455C-9EF1-5E25349BF459@directi.com> Hey Ashwinrath, Yes , I tried doing so. It is not working. Regards, Neeraj From neeraj.p at directi.com Wed Oct 23 11:10:59 2013 From: neeraj.p at directi.com (neeraj.p) Date: Wed, 23 Oct 2013 12:40:59 +0530 Subject: [Freeswitch-users] codec transcodation not working Message-ID: <013A65C0-D9E7-41D5-8350-2B9BCF6DE85F@directi.com> Hey Donny Here is my Dialplan http://pastebin.com/TAx1La4w --Neeraj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/a2788f4c/attachment.html From krice at freeswitch.org Wed Oct 23 18:10:03 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 23 Oct 2013 09:10:03 -0500 Subject: [Freeswitch-users] Performance issue In-Reply-To: Message-ID: The problem was only in the stable branch, (that would be the 1.2 line) and was fixed prior to the 1.2.14 release... On 10/23/13 9:03 AM, "Yehavi Bourvine" wrote: > HI, > ? > A question: Is it solved on 1.2.14 or only on HEAD? I tried today 1.2.14 and > failed with the same sympthoms... > ? > ?????????????????? Thanks, __Yehavi: > > > 2013/10/14 Anthony Minessale >> It is for many. ?It depends on the general goals etc. >> The problem you describe was only on stable and all the symptoms, not >> shutting down, channels count being wrong, zombies etc are all the same 1 >> problem now fixed. >> >> >> >> On Mon, Oct 14, 2013 at 11:33 AM, Stephen Wilde wrote: >>> I have tried only branch v1.2.stable not master. Is the master good for a >>> production environment? >>> >>> >>> On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale >>> wrote: >>>> That's all the same thing...... >>>> >>>> Do you have this problem on master branch as well? >>>> >>>> >>>> >>>> On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde >>>> wrote: >>>>> Thank you for your advice. >>>>> I have tried the 1.2.stable but I have due to abandon this branch due to a >>>>> some "zombie sessions" in FreeSwitch visible also during shutdown with a >>>>> "Waiting x sessions ..." messages. >>>>> I have seen that this issue (FS-5848)?has already been solved so I can try >>>>> with 1.2.stable but I don't see any fix that can affect my issue. >>>>> >>>>> >>>>> On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale >>>>> wrote: >>>>>> Try latest HEAD for a probable fix. >>>>>> >>>>>> >>>>>> >>>>>> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde >>>>>> wrote: >>>>>>> Hi all, >>>>>>> we have an issue with our FreeSwitch box that to us seems to be >>>>>>> performance related. >>>>>>> The effect is that in FreeSwitch the number of session reaches the limit >>>>>>> we set in config as: >>>>>>> >>>>>>> >>>>>>> >>>>>>> The limit is reached independently of its value because when FreeSwitch >>>>>>> is in this state, the number of sessions grows indefinitely. >>>>>>> >>>>>>> We have tried to upgrade the hardware of the box moving from a Xeon 2 >>>>>>> CPU E5649 (12 core?2.53GHz) to a Xeon 4 CPU?E5-4640 (32 core?2.40GHz) >>>>>>> but with this more powerful hardware it happens that the limit is >>>>>>> reached with less sessions. >>>>>>> It seems that performance are related to the speed of single core >>>>>>> instead of speed of the box. >>>>>>> Make sense? >>>>>>> >>>>>>> A example of "status" issued before the crash is: >>>>>>> >>>>>>> 14969852 session(s) since startup >>>>>>> 13765 session(s) - 538 out of max 1000 per sec >>>>>>> 30000 session(s) max >>>>>>> min idle cpu 0.00/36.00 >>>>>>> Current Stack Size/Max 240K/8192K >>>>>>> >>>>>>> >>>>>>> Any advice? >>>>>>> >>>>>>> ________________________________________________________________________>>>>>>> _ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://www.ss7.us/cluecon-gplus FB ClueCon : http://www.ss7.us/cluecon-fb G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus FB FreeSWITCH : http://www.ss7.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/eb4c22ed/attachment.html From nandy1925 at gmail.com Wed Oct 23 18:26:10 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 23 Oct 2013 22:26:10 +0800 Subject: [Freeswitch-users] FreeTDM tones.conf Disconnect Supervision In-Reply-To: <1382501825929-7595923.post@n2.nabble.com> References: <1382436346467-7595879.post@n2.nabble.com> <1382501825929-7595923.post@n2.nabble.com> Message-ID: Hi, I used tone_detect. I thought it was solved in your post. I inserted this before bridge or transfer app: 450 is the busy tone. 3 is the number of times required to execute "hangup". I increased 3 to 5 to prevent false hangup (under observation). See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect /Nandy On Wed, Oct 23, 2013 at 12:17 PM, xmppser wrote: > Hi, > thanks, so, how did you resolve this problem? can you give more detail? > thanks. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeTDM-tones-conf-Disconnect-Supervision-tp7595763p7595923.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/25121306/attachment.html From nandy1925 at gmail.com Wed Oct 23 18:34:33 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 23 Oct 2013 22:34:33 +0800 Subject: [Freeswitch-users] FreeTDM tones.conf Disconnect Supervision In-Reply-To: References: <1382436346467-7595879.post@n2.nabble.com> <1382501825929-7595923.post@n2.nabble.com> Message-ID: Furthermore, you must record the progress tones and analyze the frequencies. I used vanilla feature code to record calls. I think it's *2. On Wed, Oct 23, 2013 at 10:26 PM, Nandy Dagondon wrote: > Hi, > > I used tone_detect. I thought it was solved in your post. I inserted this > before bridge or transfer app: > > > > 450 is the busy tone. 3 is the number of times required to execute > "hangup". I increased 3 to 5 to prevent false hangup (under observation). > > See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect > > /Nandy > > On Wed, Oct 23, 2013 at 12:17 PM, xmppser wrote: > >> Hi, >> thanks, so, how did you resolve this problem? can you give more detail? >> thanks. >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/FreeTDM-tones-conf-Disconnect-Supervision-tp7595763p7595923.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/b1d75855/attachment-0001.html From hcoin at quietfountain.com Wed Oct 23 19:07:49 2013 From: hcoin at quietfountain.com (hcoin) Date: Wed, 23 Oct 2013 10:07:49 -0500 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling References: <526614D6.3020401@quietfountain.com> <5267694A.7090107@quietfountain.com> Message-ID: <5267E645.1090608@quietfountain.com> Donny, It's a balancing act, people choose commercial routers because they want it all to 'just work' and not have to get into the guts of it. That's what please expect and pay for-- 'just working', you pay for not being forced to learn the guts and 'deal with it'. My policy has been that if the commercial router doesn't come with a staff member at the commercial company to make the problem go away, go to an open source solution. If you're going to be made to 'deal with it', then you might as well have access to all the guts, all the tools, the whole thing yourself. Otherwise you wind up working for free making someone else's commercial product better, and who knows if the next release will break your fix or not. No, if you are being forced to deal with a problem in the guts of commercial software yourself, you aren't getting any value and the answer is find out whether an open source version is solid enough and if it is go with that. Might was well learn 'everything' about something you can compile yourself if it comes down to that. You pick up a lot of dubious skills though, for example I can now edit freeswitch transport protocols and the sip stack. A thing I hope never to have to do.... Seriously whoever came up with RTP and SIP using a bezillion ports and the whole NAT nightmare.... arg. Look how much of freeswitch is not dealing with telephone and talk issues, but routing issues. It's half a router itself. This business of weaving together products made by various vendors: routers, soft phones, pstn-voip legacy boxen, freeswitch, routers, 'guis on top of X', it's every bit as tough as programming. In programming you control 'the world' and have a narrow focus. This business of integrating lots of work by lots of folks, not for the timid. On 10/23/2013 01:42 AM, Donny Hardyanto wrote: > > I am now practicing not using standard port because some hacks couple > month ago. It was quite bad, it cost thousand of dollars and we cannot > find the culprit IP address because the router ALG rewrites them and > there is no accessible log on the router. > > Donny > > On Oct 23, 2013 1:17 PM, "hcoin" > wrote: > > Anthony and Donny, thanks for replying. > > Putting a packet capture on the line revealed the problem to be a > combination of quirks in both linphone (windows version ignores fs > nonstandard destination port) and dns-forwarder (override of > foo.bar.com fails if foo.bar.com > is a cname on the public internet, not an A > record). The call was coming in on the external profile because > the dns forwarder was letting the resolution go to the public > internet and so the local systems were sending out to the router, > which sent it back in to... the external interface. However, I > do now know how to watch calls pass through freeswitch and have > read most of the source code in the sofia endpoint, nta, nua, > etc. etc... and had lots of fun with gdb stepping around watching > the packets flow. > > The main lession I think is worth sharing is this: Use 5060 for > sip. If you are thinking of various profiles using the same > address but different ports on the one hand, or on the other hand > using ip aliases so each profile uses the 'standard' ports but a > different ip--- go with the ip alias approach. So in > /etc/network/interfaces , supposing your main nic is eth0: > > iface eth0 inet dhcp <-- or whatnot on your system> > .. > post-up ifup eth0:1 > pre-down ifdown eth0:1 > .. > > > iface eth0:1 inet static > address > netmask > > Problems all melted away as if they never were. > > Thanks again for trying to help! I even bought the freeswitch > book. Ka-Ching for someone on this list... > > > > On 10/22/2013 06:38 PM, Anthony Minessale wrote: >> >> Did you change all the fields in the new profile you duplicated >> that were relevant to the name like name... >> >> I usually cp internal.xml new.xml then edit new.xml and global >> replace internal with new right off the bat. >> >> You might find your mistake faster if you backup and revert to >> default sip profiles from sample and slowly make changes again. >> >> On Oct 22, 2013 1:04 AM, "hcoin" > > wrote: >> >> >> This has been a really frustrating problem, I'm sure the >> answer is >> simple but I just can't see it. >> >> I had several extensions registered to the internal profile, >> sending >> calls out the external profile to a sip-pstn gateway, all >> seemed fine. >> >> Then created another internal profile, using a different sip >> port on the >> same lan address, because of 'no device left behind' and NAT >> issues.. >> >> All seemed well, all the phones register normally. Looking at the >> databases in FS they all show the proper ports, the proper >> domains, etc. >> >> However, every single call gets picked up as a new call via >> sophia/external/... and it hits the public dialplan normally >> -- except >> that's the wrong plan, it should hit the default plan and be >> identified >> as sofia/internal/.... and so forth. >> 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New >> Channel >> sofia/external/hcoin at pbx.foobar.com >> >> [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] >> >> What could possibly be the reason phones registered on the >> internal >> profile have their new calls identified as sophia/external >> and don't hit >> the correct plan? Both the phones and the freeswitch are on >> the same >> subnet. This should be so vanilla. What am I missing? >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www..freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/fc1cb4ef/attachment.html From hardyanto.donny at gmail.com Wed Oct 23 19:26:41 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Wed, 23 Oct 2013 22:26:41 +0700 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling In-Reply-To: <5267E645.1090608@quietfountain.com> References: <526614D6.3020401@quietfountain.com> <5267694A.7090107@quietfountain.com> <5267E645.1090608@quietfountain.com> Message-ID: The problem is there were some SIP port scanner constantly scanning ip address for SIP known port in internet, when they found they automatically and systematically try to break SIP authentification and try to make routing. It very vicious world out there for SIP. Changing the port is the least defense we can do very minimaly. Of course they other solution like put on sbc or something but in depend on circumstances. So what ever that we can control such as our deployed softphone or ip phone, we change its the default SIP port listener. My own solution is always open source. In the hacking case I was put our solution in the partner network. I cannot control my partner what hardware they are using. And in sip/netwotk world we do interconnecting with all kind of hardware and software, whether commercial and oss. Donny On Oct 23, 2013 10:10 PM, "hcoin" wrote: > Donny, It's a balancing act, people choose commercial routers because > they want it all to 'just work' and not have to get into the guts of it. > That's what please expect and pay for-- 'just working', you pay for not > being forced to learn the guts and 'deal with it'. > > My policy has been that if the commercial router doesn't come with a staff > member at the commercial company to make the problem go away, go to an open > source solution. If you're going to be made to 'deal with it', then you > might as well have access to all the guts, all the tools, the whole thing > yourself. Otherwise you wind up working for free making someone else's > commercial product better, and who knows if the next release will break > your fix or not. No, if you are being forced to deal with a problem in > the guts of commercial software yourself, you aren't getting any value and > the answer is find out whether an open source version is solid enough and > if it is go with that. Might was well learn 'everything' about something > you can compile yourself if it comes down to that. You pick up a lot of > dubious skills though, for example I can now edit freeswitch transport > protocols and the sip stack. A thing I hope never to have to do.... > Seriously whoever came up with RTP and SIP using a bezillion ports and the > whole NAT nightmare.... arg. Look how much of freeswitch is not dealing > with telephone and talk issues, but routing issues. It's half a router > itself. > > This business of weaving together products made by various vendors: > routers, soft phones, pstn-voip legacy boxen, freeswitch, routers, 'guis > on top of X', it's every bit as tough as programming. In programming you > control 'the world' and have a narrow focus. This business of integrating > lots of work by lots of folks, not for the timid. > > > On 10/23/2013 01:42 AM, Donny Hardyanto wrote: > > I am now practicing not using standard port because some hacks couple > month ago. It was quite bad, it cost thousand of dollars and we cannot find > the culprit IP address because the router ALG rewrites them and there is no > accessible log on the router. > > Donny > On Oct 23, 2013 1:17 PM, "hcoin" wrote: > >> Anthony and Donny, thanks for replying. >> >> Putting a packet capture on the line revealed the problem to be a >> combination of quirks in both linphone (windows version ignores fs >> nonstandard destination port) and dns-forwarder (override of foo.bar.comfails if >> foo.bar.com is a cname on the public internet, not an A record). The >> call was coming in on the external profile because the dns forwarder was >> letting the resolution go to the public internet and so the local systems >> were sending out to the router, which sent it back in to... the external >> interface. However, I do now know how to watch calls pass through >> freeswitch and have read most of the source code in the sofia endpoint, >> nta, nua, etc. etc... and had lots of fun with gdb stepping around >> watching the packets flow. >> >> The main lession I think is worth sharing is this: Use 5060 for sip. >> If you are thinking of various profiles using the same address but >> different ports on the one hand, or on the other hand using ip aliases so >> each profile uses the 'standard' ports but a different ip--- go with the ip >> alias approach. So in /etc/network/interfaces , supposing your main nic >> is eth0: >> >> iface eth0 inet dhcp <-- or whatnot on your system> >> .. >> post-up ifup eth0:1 >> pre-down ifdown eth0:1 >> .. >> >> >> iface eth0:1 inet static >> address >> netmask >> >> Problems all melted away as if they never were. >> >> Thanks again for trying to help! I even bought the freeswitch book. >> Ka-Ching for someone on this list... >> >> >> >> On 10/22/2013 06:38 PM, Anthony Minessale wrote: >> >> Did you change all the fields in the new profile you duplicated that were >> relevant to the name like name... >> >> I usually cp internal.xml new.xml then edit new.xml and global replace >> internal with new right off the bat. >> >> You might find your mistake faster if you backup and revert to default >> sip profiles from sample and slowly make changes again. >> On Oct 22, 2013 1:04 AM, "hcoin" wrote: >> >>> >>> This has been a really frustrating problem, I'm sure the answer is >>> simple but I just can't see it. >>> >>> I had several extensions registered to the internal profile, sending >>> calls out the external profile to a sip-pstn gateway, all seemed fine. >>> >>> Then created another internal profile, using a different sip port on the >>> same lan address, because of 'no device left behind' and NAT issues.. >>> >>> All seemed well, all the phones register normally. Looking at the >>> databases in FS they all show the proper ports, the proper domains, etc. >>> >>> However, every single call gets picked up as a new call via >>> sophia/external/... and it hits the public dialplan normally -- except >>> that's the wrong plan, it should hit the default plan and be identified >>> as sofia/internal/.... and so forth. >>> 2013-10-22 00:31:11.001600 [NOTICE] switch_channel.c:1034 New Channel >>> sofia/external/hcoin at pbx.foobar.com[28ed125a-3adb-11e3-9cc1-cbb8efb09b83] >>> >>> What could possibly be the reason phones registered on the internal >>> profile have their new calls identified as sophia/external and don't hit >>> the correct plan? Both the phones and the freeswitch are on the same >>> subnet. This should be so vanilla. What am I missing? >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www..freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www..freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/f3245b8e/attachment-0001.html From hcoin at quietfountain.com Wed Oct 23 20:00:37 2013 From: hcoin at quietfountain.com (hcoin) Date: Wed, 23 Oct 2013 11:00:37 -0500 Subject: [Freeswitch-users] Phones registered to internal profile hit external profile when calling References: <526614D6.3020401@quietfountain.com> <5267694A.7090107@quietfountain.com> <5267E645.1090608@quietfountain.com> Message-ID: <5267F2A5.4010203@quietfountain.com> For 'bad guys using SIP' I added the package 'fail2ban'. And then set it up to watch for failed auth attempts and attempts to conenct without the proper domain, then set it to ban those ips for a while. When I get the time, I'm going to set it up to answer the vicious person's call, then put them on hold for a long time, then ban them and then also send an email. That way their systems get tied up. Hopefully a group like 'spamhaus' or other will create a pool of sip spammers so we can deny that traffic before it hits freeswitch. I think some legislation is in order so the police can arrest chronic bad actors and fine them to offset the cost of 'spamhaus' type setups and confiscate their gear. On 10/23/2013 10:26 AM, Donny Hardyanto wrote: > > The problem is there were some SIP port scanner constantly scanning ip > address for SIP known port in internet, when they found they > automatically and systematically try to break SIP authentification and > try to make routing. It very vicious world out there for SIP. Changing > the port is the least defense we can do very minimaly. Of course they > other solution like put on sbc or something but in depend on > circumstances. So what ever that we can control such as our deployed > softphone or ip phone, we change its the default SIP port listener. > > My own solution is always open source. In the hacking case I was put > our solution in the partner network. I cannot control my partner what > hardware they are using. And in sip/netwotk world we do > interconnecting with all kind of hardware and software, whether > commercial and oss. > > Donny > > On Oct 23, 2013 10:10 PM, "hcoin" > wrote: > > Donny, It's a balancing act, people choose commercial routers > because they want it all to 'just work' and not have to get into > the guts of it. That's what please expect and pay for-- 'just > working', you pay for not being forced to learn the guts and 'deal > with it'. > > My policy has been that if the commercial router doesn't come with > a staff member at the commercial company to make the problem go > away, go to an open source solution. If you're going to be made > to 'deal with it', then you might as well have access to all the > guts, all the tools, the whole thing yourself. Otherwise you wind > up working for free making someone else's commercial product > better, and who knows if the next release will break your fix or > not. No, if you are being forced to deal with a problem in the > guts of commercial software yourself, you aren't getting any value > and the answer is find out whether an open source version is solid > enough and if it is go with that. Might was well learn > 'everything' about something you can compile yourself if it comes > down to that. You pick up a lot of dubious skills though, for > example I can now edit freeswitch transport protocols and the sip > stack. A thing I hope never to have to do.... Seriously whoever > came up with RTP and SIP using a bezillion ports and the whole NAT > nightmare.... arg. Look how much of freeswitch is not dealing > with telephone and talk issues, but routing issues. It's half a > router itself. > > This business of weaving together products made by various > vendors: routers, soft phones, pstn-voip legacy boxen, > freeswitch, routers, 'guis on top of X', it's every bit as tough > as programming. In programming you control 'the world' and have a > narrow focus. This business of integrating lots of work by lots > of folks, not for the timid. > > > On 10/23/2013 01:42 AM, Donny Hardyanto wrote: >> >> I am now practicing not using standard port because some hacks >> couple month ago. It was quite bad, it cost thousand of dollars >> and we cannot find the culprit IP address because the router ALG >> rewrites them and there is no accessible log on the router. >> >> Donny >> >> On Oct 23, 2013 1:17 PM, "hcoin" > > wrote: >> >> Anthony and Donny, thanks for replying. >> >> Putting a packet capture on the line revealed the problem to >> be a combination of quirks in both linphone (windows version >> ignores fs nonstandard destination port) and dns-forwarder >> (override of foo.bar.com fails if >> foo..bar.com is a cname on the public >> internet, not an A record). The call was coming in on the >> external profile because the dns forwarder was letting the >> resolution go to the public internet and so the local systems >> were sending out to the router, which sent it back in to... >> the external interface. However, I do now know how to watch >> calls pass through freeswitch and have read most of the >> source code in the sofia endpoint, nta, nua, etc. etc... and >> had lots of fun with gdb stepping around watching the packets >> flow. >> >> The main lession I think is worth sharing is this: Use 5060 >> for sip. If you are thinking of various profiles using the >> same address but different ports on the one hand, or on the >> other hand using ip aliases so each profile uses the >> 'standard' ports but a different ip--- go with the ip alias >> approach. So in /etc/network/interfaces , supposing your >> main nic is eth0: >> >> iface eth0 inet dhcp <-- or whatnot on your system> >> .. >> post-up ifup eth0:1 >> pre-down ifdown eth0:1 >> .. >> >> >> iface eth0:1 inet static >> address >> netmask >> >> Problems all melted away as if they never were. >> >> Thanks again for trying to help! I even bought the >> freeswitch book. Ka-Ching for someone on this list... >> >> >> >> On 10/22/2013 06:38 PM, Anthony Minessale wrote: >>> >>> Did you change all the fields in the new profile you >>> duplicated that were relevant to the name like name... >>> >>> I usually cp internal.xml new.xml then edit new.xml and >>> global replace internal with new right off the bat. >>> >>> You might find your mistake faster if you backup and revert >>> to default sip profiles from sample and slowly make changes >>> again. >>> >>> On Oct 22, 2013 1:04 AM, "hcoin" >> > wrote: >>> >>> >>> This has been a really frustrating problem, I'm sure the >>> answer is >>> simple but I just can't see it. >>> >>> I had several extensions registered to the internal >>> profile, sending >>> calls out the external profile to a sip-pstn gateway, >>> all seemed fine. >>> >>> Then created another internal profile, using a different >>> sip port on the >>> same lan address, because of 'no device left behind' and >>> NAT issues.. >>> >>> All seemed well, all the phones register normally. >>> Looking at the >>> databases in FS they all show the proper ports, the >>> proper domains, etc. >>> >>> However, every single call gets picked up as a new call via >>> sophia/external/... and it hits the public dialplan >>> normally -- except >>> that's the wrong plan, it should hit the default plan >>> and be identified >>> as sofia/internal/.... and so forth. >>> 2013-10-22 00:31:11.001600 [NOTICE] >>> switch_channel.c:1034 New Channel >>> sofia/external/hcoin at pbx.foobar.com >>> >>> [28ed125a-3adb-11e3-9cc1-cbb8efb09b83] >>> >>> What could possibly be the reason phones registered on >>> the internal >>> profile have their new calls identified as >>> sophia/external and don't hit >>> the correct plan? Both the phones and the freeswitch >>> are on the same >>> subnet. This should be so vanilla. What am I missing? >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www..freeswitchsolutions.com >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www...freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www..freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www..freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/cc0609bd/attachment-0001.html From matt at williamsmjw.com Wed Oct 23 21:07:56 2013 From: matt at williamsmjw.com (Matt Williams) Date: Wed, 23 Oct 2013 12:07:56 -0500 Subject: [Freeswitch-users] Limit help In-Reply-To: <026701cecf7b$d1225be0$736713a0$@verizon.net> References: <0MV20027DW5OHY90@vms173005.mailsrvcs.net> <026701cecf7b$d1225be0$736713a0$@verizon.net> Message-ID: You need to call limit twice because you are limiting to separate instances. > **** > > expression="^1?[2-9]\d{2}[2-9]\d{6}$"> > > **** > > **** > > ** ** > > **** > > > On Tue, Oct 22, 2013 at 6:09 PM, Andre wrote: > Ok, I think I got it. Can you verify I got it J**** > > ** ** > > I can use this code to limit my gateways**** > > **** > > expression="^1?[2-9]\d{2}[2-9]\d{6}$">**** > > **** > > **** > > **** > > **** > > ** ** > > And this code for my customers**** > > ** ** > > 5 ports > > > 5 cps > > > **** > > ** ** > > How do I add CPS to the gateways using the above example? Is this right?** > ** > > ** ** > > > ** ** > > > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt > Williams > *Sent:* Tuesday, October 22, 2013 6:06 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Limit help**** > > ** ** > > You can use limit_execute to execute any application you wish. The best > way to wrap your head around how simple and useful the limit application is > to use is to play with it a little while. Everything should click into > place. **** > > > **** > > Thank You,**** > > Matthew Williams > IKN Network Operations**** > > ** ** > > ** ** > > On Tue, Oct 22, 2013 at 11:21 AM, Andre Demattia > wrote:**** > > Can I use limit_execute for a outbound gateway in place of the loop back > option?**** > ------------------------------ > > *From: *Matt Williams > *Sent: *10/22/2013 12:01 PM > *To: *FreeSWITCH Users Help > *Subject: *Re: [Freeswitch-users] Limit help**** > > You need to set your CPS and port limits separately. You can set limits > per gateway the same way you would limit any application. It is not bound > to any specific application. The values you use are completely arbitrary. > That said here is an example from the Wiki > http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits > **** > > > **** > > ** ** > > ** ** > > On Tue, Oct 22, 2013 at 10:40 AM, Andre Demattia > wrote:**** > > Hi, how do I set the CPS and Port limits on a bridge > > Can I set both on one application or do I need to set 2 limits? > > 5 ports > > > 5 cps > > > Can I do this for both 5 ports and 5 cps? > > > > Also how do I do the same on a gateway? I need to make sure I don't send > too many calls to the provider.**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/783638e4/attachment.html From roger.castaldo at gmail.com Wed Oct 23 23:06:26 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Wed, 23 Oct 2013 15:06:26 -0400 Subject: [Freeswitch-users] FreeSwitchConfig GUI Demo Server Message-ID: Hi everyone, I have finally had the time and ability to get up a bit of a demo server for the FreeSwitchConfig gui that I have been working on. https://wiki.freeswitch.org/wiki/Freeswitch_Gui#FreeSWITCHConfig It is on a single CPU, low memory, free VPS, so its not going to respond as fast as it could on real hardware. As well as the server is kind of locked down for connecting to the freeswitch instance because its purpose is to demo the site, and hopefully get some feedback on the interface itself, as well as maybe get some interest in people using it, or helping develop it. site: http://freeswitchconfig.dalnet.ca:8080 user name: admin password: admin As per the wiki entry, every night it rebuilds to the current trunk as well as wipes out all the database data. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/3a66302b/attachment.html From grcamauer at gmail.com Wed Oct 23 23:28:58 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 23 Oct 2013 17:28:58 -0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: <526789CB.80905@softnet.si> References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> <52667E76.8010904@coppice.org> <526789CB.80905@softnet.si> Message-ID: Here is some good data on per call real bandwidth consumption for different codecs: Codec BR NEB G.711 64 Kbps 87.2 Kbps G.729 8 Kbps 31.2 Kbps G.723.1 6.4 Kbps 21.9 Kbps G.723.1 5.3 Kbps 20.8 Kbps G.726 32 Kbps 55.2 Kbps G.726 24 Kbps 47.2 Kbps G.728 16 Kbps 31.5 Kbps iLBC 15 Kbps 27.7 Kbps BR = Bit rate NEB = Nominal Ethernet Bandwidth (one direction) On Wed, Oct 23, 2013 at 5:33 AM, Miha wrote: > Hi Steve, > > do you have any good preposal which switch has good performance for voip > trafic? > > miha > > Dne 10/22/2013 3:32 PM, pi?e Steve Underwood: > > Hi Grant, > > > > Two possibilities spring to mind: > > > > - If your audio is coming from a disk, can that disk keep up? > > - How good is your ethernet switch? > > > > Notice in the second point I said how good, not how expensive. Many > > switches choke on a large number of small media packets, including some > > expensive big name products. > > > > Regards, > > Steve > > > > On 10/22/2013 06:01 PM, Grant Bagdasarian wrote: > >> The network shouldn?t be an issue, since we have at least 1Gbps lines. > >> The tests stay within the network. > >> > >> I forgot to mention the calls are being distributed across two > >> machines by a Kamailio instance. > >> > >> So for a total of 800 concurrent calls generated by Sipp, each machine > >> has 400 active calls. > >> > >> CPU load reaches about 70% per machine. > >> > >> At this point both FS machines are virtualized, since the performance > >> gain wasn?t that much compared to physical. > >> > >> The VM host shows it is using ~3/4 of its CPU resources. > >> > >> Htop shows that the normal priority threads(green) and the kernel > >> threads(red) are about the same length. > >> > >> Also, FS is running on Ubuntu Server 12.04 x64. > >> > >> *From:*freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > >> *Stanislav Sinyagin > >> *Sent:* Tuesday, October 22, 2013 11:11 AM > >> *To:* FreeSWITCH Users Help > >> *Subject:* Re: [Freeswitch-users] Audio quality issues > >> > >> 800 calls at 64kbps is 51Mbps. > >> Could there be a network issue, like a 100Mbps line between the > endpoints? > >> > >> How heavy is your CPU load? "htop" command would be helpful in this. > >> > >> ------------------------------------------------------------------------ > >> > >> *From:*Grant Bagdasarian > > >> *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org > >> )" > >> >> > > >> *Sent:* Tuesday, October 22, 2013 10:14 AM > >> *Subject:* [Freeswitch-users] Audio quality issues > >> > >> Hello, > >> > >> I was wondering what the maximum concurrent calls for FS before audio > >> quality becomes an issue? I assume the specs of the machine would also > >> affect this. > >> > >> We are currently running FS on a Six Core (12 Threads) Intel E5-2430 > >> CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality > >> at these rates is still fair, but we do notice some quality issue?s. > >> > >> Going above these numbers screws up the audio quality: choppy sound, > >> audio drops etc. We aren?t doing any heavy media processing, just > >> simply playing a file (G711-Alaw) which lasts about 2 minutes during > >> the load test. > >> > >> These numbers are for one way audio, where Sipp doesn?t echo the RTP > >> back. These numbers get lower once Sipp echo?s the RTP. > >> > >> I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), > >> but the performance gain on physical vs virtual isn?t that much. > >> > >> I disabled all the modules we don?t need, like CDR?s, conferencing, etc. > >> > >> Are there any parameters(config files)/modules that can affect the > >> quality of the audio stream? > >> > >> Regards, > >> > >> Grant > >> > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/63502d5d/attachment-0001.html From grcamauer at gmail.com Wed Oct 23 23:48:04 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 23 Oct 2013 17:48:04 -0200 Subject: [Freeswitch-users] Audio quality issues In-Reply-To: References: <1382433049.67376.YahooMailNeo@web126201.mail.ne1.yahoo.com> <52667E76.8010904@coppice.org> <526789CB.80905@softnet.si> Message-ID: And a link with a great explanation of how to arrive at some of those numbers: http://voiceonbits.com/2010/08/15/codec-bandwidth-calculation-g711g729/ Guillermo On Wed, Oct 23, 2013 at 4:28 PM, Guillermo Ruiz Camauer wrote: > Here is some good data on per call real bandwidth consumption for > different codecs: > > Codec BR NEB > G.711 64 Kbps 87.2 Kbps > G.729 8 Kbps 31.2 Kbps > G.723.1 6.4 Kbps 21.9 Kbps > G.723.1 5.3 Kbps 20.8 Kbps > G.726 32 Kbps 55.2 Kbps > G.726 24 Kbps 47.2 Kbps > G.728 16 Kbps 31.5 Kbps > iLBC 15 Kbps 27.7 Kbps > > BR = Bit rate > NEB = Nominal Ethernet Bandwidth (one direction) > > > On Wed, Oct 23, 2013 at 5:33 AM, Miha wrote: > >> Hi Steve, >> >> do you have any good preposal which switch has good performance for voip >> trafic? >> >> miha >> >> Dne 10/22/2013 3:32 PM, pi?e Steve Underwood: >> > Hi Grant, >> > >> > Two possibilities spring to mind: >> > >> > - If your audio is coming from a disk, can that disk keep up? >> > - How good is your ethernet switch? >> > >> > Notice in the second point I said how good, not how expensive. Many >> > switches choke on a large number of small media packets, including some >> > expensive big name products. >> > >> > Regards, >> > Steve >> > >> > On 10/22/2013 06:01 PM, Grant Bagdasarian wrote: >> >> The network shouldn?t be an issue, since we have at least 1Gbps lines. >> >> The tests stay within the network. >> >> >> >> I forgot to mention the calls are being distributed across two >> >> machines by a Kamailio instance. >> >> >> >> So for a total of 800 concurrent calls generated by Sipp, each machine >> >> has 400 active calls. >> >> >> >> CPU load reaches about 70% per machine. >> >> >> >> At this point both FS machines are virtualized, since the performance >> >> gain wasn?t that much compared to physical. >> >> >> >> The VM host shows it is using ~3/4 of its CPU resources. >> >> >> >> Htop shows that the normal priority threads(green) and the kernel >> >> threads(red) are about the same length. >> >> >> >> Also, FS is running on Ubuntu Server 12.04 x64. >> >> >> >> *From:*freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >> >> *Stanislav Sinyagin >> >> *Sent:* Tuesday, October 22, 2013 11:11 AM >> >> *To:* FreeSWITCH Users Help >> >> *Subject:* Re: [Freeswitch-users] Audio quality issues >> >> >> >> 800 calls at 64kbps is 51Mbps. >> >> Could there be a network issue, like a 100Mbps line between the >> endpoints? >> >> >> >> How heavy is your CPU load? "htop" command would be helpful in this. >> >> >> >> >> ------------------------------------------------------------------------ >> >> >> >> *From:*Grant Bagdasarian > >> >> *To:* "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org >> >> )" >> >> > >> > >> >> *Sent:* Tuesday, October 22, 2013 10:14 AM >> >> *Subject:* [Freeswitch-users] Audio quality issues >> >> >> >> Hello, >> >> >> >> I was wondering what the maximum concurrent calls for FS before audio >> >> quality becomes an issue? I assume the specs of the machine would also >> >> affect this. >> >> >> >> We are currently running FS on a Six Core (12 Threads) Intel E5-2430 >> >> CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality >> >> at these rates is still fair, but we do notice some quality issue?s. >> >> >> >> Going above these numbers screws up the audio quality: choppy sound, >> >> audio drops etc. We aren?t doing any heavy media processing, just >> >> simply playing a file (G711-Alaw) which lasts about 2 minutes during >> >> the load test. >> >> >> >> These numbers are for one way audio, where Sipp doesn?t echo the RTP >> >> back. These numbers get lower once Sipp echo?s the RTP. >> >> >> >> I?ve tried FS on a physical box and also on a virtual box (ESXi 5.1), >> >> but the performance gain on physical vs virtual isn?t that much. >> >> >> >> I disabled all the modules we don?t need, like CDR?s, conferencing, >> etc. >> >> >> >> Are there any parameters(config files)/modules that can affect the >> >> quality of the audio stream? >> >> >> >> Regards, >> >> >> >> Grant >> >> >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Guillermo Ruiz Camauer > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/250d8692/attachment.html From mgandikota at nts.net Thu Oct 24 00:08:53 2013 From: mgandikota at nts.net (Murthy Gandikota) Date: Wed, 23 Oct 2013 13:08:53 -0700 Subject: [Freeswitch-users] Out of band DTMF in Javascript Message-ID: <96E5DBE070D1624EAD3C8C6E3257AA630184FA65@MAILER01.ad.nts.net> Hi Can anyone please confirm that the following template is RFC2833 compliant out-of-band DTMF for the #3#? "%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)"; I am trying to use TeleTone to generate the #3# response to a caller as follows: var pound3pound ="v=-7;%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)"; var tts = new TeleTone(session); tts.generate(pound3pound); I can hear the digits play, but don't know if they are RFC2833 compliant out-of-band DTMF. Thanks Murthy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/04072140/attachment.html From abaci64 at gmail.com Thu Oct 24 00:33:49 2013 From: abaci64 at gmail.com (Abaci) Date: Wed, 23 Oct 2013 16:33:49 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> Message-ID: <526832AD.3090202@gmail.com> Can someone please confirm if 100rel is fixed and stable in FreeSWITCH? I know that the ticket was closed but if you read the last comment by Mike Jerris "this issue is fixed in svn now (at least with 100rel disabled). 100rel support is a large known issue and needs to be fixed upstream in the sofia-sip library" it only says that when it's disabled it's not crashing. in the default config in the comments it says "There are known issues (asserts and segfaults) when 100rel is enabled. It is not recommended to enable 100rel at this time." I would appreciate if anyone that actually knows can confirm if it's fixed, if not is there anything that can be done to get it fixed. On 8/24/2012 5:33 PM, Peter Olsson wrote: > Since the ticket is closed, I guess it works just fine. > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [mike.burlingame at me.com] > Skickat: den 24 augusti 2012 21:48 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] enable-100rel question > > I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? > > > >From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files > enable-100rel > This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. > > > > !DSPAM:5037d87c32761841043968! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Oct 24 00:50:35 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 16:50:35 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: <526832AD.3090202@gmail.com> References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> Message-ID: <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> Those comments below are accurate to my knowledge. On Oct 23, 2013, at 4:33 PM, Abaci wrote: > Can someone please confirm if 100rel is fixed and stable in FreeSWITCH? > I know that the ticket was closed but if you read the last comment by > Mike Jerris "this issue is fixed in svn now (at least with 100rel > disabled). 100rel support is a large known issue and needs to be fixed > upstream in the sofia-sip library" it only says that when it's disabled > it's not crashing. in the default config in the comments it says "There > are known issues (asserts and segfaults) when 100rel is enabled. It is > not recommended to enable 100rel at this time." > I would appreciate if anyone that actually knows can confirm if it's > fixed, if not is there anything that can be done to get it fixed. > > On 8/24/2012 5:33 PM, Peter Olsson wrote: >> Since the ticket is closed, I guess it works just fine. >> ________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [mike.burlingame at me.com] >> Skickat: den 24 augusti 2012 21:48 >> Till: FreeSWITCH Users Help >> ?mne: [Freeswitch-users] enable-100rel question >> >> I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? >> >> >>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >> enable-100rel >> This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. >> >> >> >> !DSPAM:5037d87c32761841043968! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From abaci64 at gmail.com Thu Oct 24 00:57:38 2013 From: abaci64 at gmail.com (Abaci) Date: Wed, 23 Oct 2013 16:57:38 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> Message-ID: <52683842.3030606@gmail.com> Is there anything that be done to get it fixed? I have had issues lately with some phones not getting the 183 and therefor not playing ringback (changing to tcp seems to help but would prefer not have to do that). On 10/23/2013 4:50 PM, Michael Jerris wrote: > Those comments below are accurate to my knowledge. > > On Oct 23, 2013, at 4:33 PM, Abaci wrote: > >> Can someone please confirm if 100rel is fixed and stable in FreeSWITCH? >> I know that the ticket was closed but if you read the last comment by >> Mike Jerris "this issue is fixed in svn now (at least with 100rel >> disabled). 100rel support is a large known issue and needs to be fixed >> upstream in the sofia-sip library" it only says that when it's disabled >> it's not crashing. in the default config in the comments it says "There >> are known issues (asserts and segfaults) when 100rel is enabled. It is >> not recommended to enable 100rel at this time." >> I would appreciate if anyone that actually knows can confirm if it's >> fixed, if not is there anything that can be done to get it fixed. >> >> On 8/24/2012 5:33 PM, Peter Olsson wrote: >>> Since the ticket is closed, I guess it works just fine. >>> ________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [mike.burlingame at me.com] >>> Skickat: den 24 augusti 2012 21:48 >>> Till: FreeSWITCH Users Help >>> ?mne: [Freeswitch-users] enable-100rel question >>> >>> I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? >>> >>> >>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >>> enable-100rel >>> This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. >>> >>> >>> >>> !DSPAM:5037d87c32761841043968! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vermeulen.deon at gmail.com Thu Oct 24 01:08:12 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Wed, 23 Oct 2013 23:08:12 +0200 Subject: [Freeswitch-users] SDReporter Integration with FS In-Reply-To: <526666D6.6010901@gmail.com> References: <526666D6.6010901@gmail.com> Message-ID: <52683ABC.2060202@gmail.com> I manage to get CDRs imported into SDReporter. This is what my cdr_csv.conf.xml file looks like. The "default" asterisk profile does not work. SDReporter is very specific in the format of the CDR and especially with the Channel, so I had to "normalize" to Asterisk equivalent. Now off to getting the QoS variable outputs into SDReporter. Kind Regards -- > Deon Vermeulen > October 22, 2013 1:51 PM > Hi > > > I've successfully setup SDReporter, FreeSWITCH to export CDRs using > cdr_csv with the Asterisk template. > I've setup a cron to "automatically" push the csv file(s) to > SDReporter $HOME_DIR/SDReporter/CDRConverter-4.0.8/data/INPUT/Asterisk/1/ > > I've tried different options in the format of the CDRs ,but no matter > what I do I keep getting an " CDRFieldChannelAsterisk(String): wrong > Asterisk channel. " ERROR. > I've removed the "${channel_name}","${bridge_channel}" variables, but > still I get the above error. > > I currently run FreeSWITCH Version 1.5.6b+git~20131018T052734Z~e0054af96f. > > Has anyone successfully got SDReporter working with FS? > > > Thanks for any assistance. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/26a42bda/attachment.html From mike at jerris.com Thu Oct 24 01:10:03 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Oct 2013 17:10:03 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: <52683842.3030606@gmail.com> References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> Message-ID: <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> It's all code, of course there are things that can be done. On Oct 23, 2013, at 4:57 PM, Abaci wrote: > Is there anything that be done to get it fixed? > I have had issues lately with some phones not getting the 183 and > therefor not playing ringback (changing to tcp seems to help but would > prefer not have to do that). > > On 10/23/2013 4:50 PM, Michael Jerris wrote: >> Those comments below are accurate to my knowledge. >> >> On Oct 23, 2013, at 4:33 PM, Abaci wrote: >> >>> Can someone please confirm if 100rel is fixed and stable in FreeSWITCH? >>> I know that the ticket was closed but if you read the last comment by >>> Mike Jerris "this issue is fixed in svn now (at least with 100rel >>> disabled). 100rel support is a large known issue and needs to be fixed >>> upstream in the sofia-sip library" it only says that when it's disabled >>> it's not crashing. in the default config in the comments it says "There >>> are known issues (asserts and segfaults) when 100rel is enabled. It is >>> not recommended to enable 100rel at this time." >>> I would appreciate if anyone that actually knows can confirm if it's >>> fixed, if not is there anything that can be done to get it fixed. >>> >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: >>>> Since the ticket is closed, I guess it works just fine. >>>> ________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [mike.burlingame at me.com] >>>> Skickat: den 24 augusti 2012 21:48 >>>> Till: FreeSWITCH Users Help >>>> ?mne: [Freeswitch-users] enable-100rel question >>>> >>>> I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? >>>> >>>> >>>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >>>> enable-100rel >>>> This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. >>>> >>>> >>>> >>>> !DSPAM:5037d87c32761841043968! From cal.leeming at simplicitymedialtd.co.uk Thu Oct 24 01:38:12 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 23 Oct 2013 22:38:12 +0100 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: Message-ID: For those that missed it, you can watch the whole thing here; http://www.youtube.com/watch?v=raXkHi_uGF8 Slides here; https://www.dropbox.com/s/hp5fj7e7o1mdnyt/Auto%20provisioning%20sucks.pptx Cal On Tue, Oct 22, 2013 at 7:36 PM, Ken Rice wrote: > ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM > Eastern Live! > > Cal Lemming, Will be making a 0day disclosure! > > Don?t miss this exclusive presentation for the FreeSwitch community! > > This is one you will not want to miss! > > Join us on the FreeSWITCH Bridge for this Special Presentation! > > For conference call access information visit *http://ss7.us/call888 > * > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > G+ ClueCon : http://www.ss7.us/cluecon-gplus > FB ClueCon : http://www.ss7.us/cluecon-fb > G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus > FB FreeSWITCH : http://www.ss7.us/freeswitch-fb > Twitter : @FreeSWITCH_WIRE > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/f0d1ea09/attachment.html From sertys at gmail.com Thu Oct 24 02:24:23 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 24 Oct 2013 00:24:23 +0200 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: Message-ID: It is much appreciated that you share this with the community. I have seen known issues in proprietary firmware in embedded devices for AGES. And the vendors just think their security through obscurity modus operandi will go forever and noone would care to inspect their .bin files. In times like this i want to dedicate myself to toll fraud just to stick it to imbeciles pushing firmware. On Wed, Oct 23, 2013 at 11:38 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > For those that missed it, you can watch the whole thing here; > http://www.youtube.com/watch?v=raXkHi_uGF8 > > Slides here; > https://www.dropbox.com/s/hp5fj7e7o1mdnyt/Auto%20provisioning%20sucks.pptx > > Cal > > > On Tue, Oct 22, 2013 at 7:36 PM, Ken Rice wrote: > >> ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM >> Eastern Live! >> >> Cal Lemming, Will be making a 0day disclosure! >> >> Don?t miss this exclusive presentation for the FreeSwitch community! >> >> This is one you will not want to miss! >> >> Join us on the FreeSWITCH Bridge for this Special Presentation! >> >> For conference call access information visit *http://ss7.us/call888 >> * >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> G+ ClueCon : http://www.ss7.us/cluecon-gplus >> FB ClueCon : http://www.ss7.us/cluecon-fb >> G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus >> FB FreeSWITCH : http://www.ss7.us/freeswitch-fb >> Twitter : @FreeSWITCH_WIRE >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/aed61d26/attachment-0001.html From krice at freeswitch.org Thu Oct 24 02:30:39 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 23 Oct 2013 17:30:39 -0500 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: Message-ID: Again Thanks Cal! Today was great! Check out the links below for those that missed it! Hopefully this will get some people thinking about what not to do in the future and get some vendors to address things. Tl;dr version: Cal found a multi-vendor problem with auto-provisioning services... See the youtube link for the presentation. On 10/23/13 4:38 PM, "Cal Leeming [Simplicity Media Ltd]" wrote: > For those that missed it, you can watch the whole thing here; > http://www.youtube.com/watch?v=raXkHi_uGF8 > > Slides here;? > https://www.dropbox.com/s/hp5fj7e7o1mdnyt/Auto%20provisioning%20sucks.pptx > > Cal > > > On Tue, Oct 22, 2013 at 7:36 PM, Ken Rice wrote: >> ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM >> Eastern Live! >> >> Cal Lemming, Will be making a 0day disclosure! >> >> Don?t miss this exclusive presentation for the FreeSwitch community! >> >> This is one you will not want to miss! >> >> Join us on the FreeSWITCH Bridge for this Special Presentation! >> >> For conference call access information visit http://ss7.us/call888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/49dce3f0/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Oct 24 03:31:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 24 Oct 2013 00:31:57 +0100 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: Message-ID: I completely agree. There are some great talks about there about reverse engineering firmware. For anyone with some spare time on their hands, this makes for a really fun side project, just remember to disclose in a responsible manner! Cal On Wed, Oct 23, 2013 at 11:24 PM, Daniel Ivanov wrote: > It is much appreciated that you share this with the community. I have seen > known issues in proprietary firmware in embedded devices for AGES. And the > vendors just think their security through obscurity modus operandi will go > forever and noone would care to inspect their .bin files. > In times like this i want to dedicate myself to toll fraud just to stick > it to imbeciles pushing firmware. > > > On Wed, Oct 23, 2013 at 11:38 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> For those that missed it, you can watch the whole thing here; >> http://www.youtube.com/watch?v=raXkHi_uGF8 >> >> Slides here; >> https://www.dropbox.com/s/hp5fj7e7o1mdnyt/Auto%20provisioning%20sucks.pptx >> >> Cal >> >> >> On Tue, Oct 22, 2013 at 7:36 PM, Ken Rice wrote: >> >>> ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM >>> Eastern Live! >>> >>> Cal Lemming, Will be making a 0day disclosure! >>> >>> Don?t miss this exclusive presentation for the FreeSwitch community! >>> >>> This is one you will not want to miss! >>> >>> Join us on the FreeSWITCH Bridge for this Special Presentation! >>> >>> For conference call access information visit *http://ss7.us/call888 >>> * >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> G+ ClueCon : http://www.ss7.us/cluecon-gplus >>> FB ClueCon : http://www.ss7.us/cluecon-fb >>> G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus >>> FB FreeSWITCH : http://www.ss7.us/freeswitch-fb >>> Twitter : @FreeSWITCH_WIRE >>> irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/95e3dc70/attachment.html From ben at langfeld.co.uk Thu Oct 24 03:58:38 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 23 Oct 2013 21:58:38 -0200 Subject: [Freeswitch-users] Out of band DTMF in Javascript In-Reply-To: <96E5DBE070D1624EAD3C8C6E3257AA630184FA65@MAILER01.ad.nts.net> References: <96E5DBE070D1624EAD3C8C6E3257AA630184FA65@MAILER01.ad.nts.net> Message-ID: Why not use the send_dtmf app? https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf On 23 October 2013 18:08, Murthy Gandikota wrote: > Hi**** > > ** ** > > Can anyone please confirm that the following template is RFC2833 compliant > out-of-band DTMF for the #3#? **** > > ** ** > > "%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)";**** > > ** ** > > I am trying to use TeleTone to generate the #3# response to a caller as > follows:**** > > ** ** > > var pound3pound ="v=-7;%(200,0,941.0,1477.0);%(200,0,697.0, > 1477.0);%(200,0,941.0,1477.0)";**** > > var tts = new TeleTone(session);**** > > tts.generate(pound3pound);**** > > ** ** > > I can hear the digits play, but don't know if they are RFC2833 compliant > out-of-band DTMF.**** > > ** ** > > Thanks**** > > Murthy**** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/99fd6131/attachment-0001.html From steveayre at gmail.com Thu Oct 24 04:37:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Oct 2013 01:37:10 +0100 Subject: [Freeswitch-users] Out of band DTMF in Javascript In-Reply-To: <96E5DBE070D1624EAD3C8C6E3257AA630184FA65@MAILER01.ad.nts.net> References: <96E5DBE070D1624EAD3C8C6E3257AA630184FA65@MAILER01.ad.nts.net> Message-ID: TeleTone generates audible tones. That's inband dtmf. That won't give you anything out of band / rfc2833 Rfc2833 packets simply contain the character (3 # etc) being pressed and its duration. You need to use session.execute("send_dtmf", "#3#") to generate those. On Wednesday, October 23, 2013, Murthy Gandikota wrote: > Hi**** > > ** ** > > Can anyone please confirm that the following template is RFC2833 compliant > out-of-band DTMF for the #3#? **** > > ** ** > > "%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)";**** > > ** ** > > I am trying to use TeleTone to generate the #3# response to a caller as > follows:**** > > ** ** > > var pound3pound ="v=-7;%(200,0,941.0,1477.0);%(200,0,697.0, > 1477.0);%(200,0,941.0,1477.0)";**** > > var tts = new TeleTone(session);**** > > tts.generate(pound3pound);**** > > ** ** > > I can hear the digits play, but don't know if they are RFC2833 compliant > out-of-band DTMF.**** > > ** ** > > Thanks**** > > Murthy**** > > ** ** > > ** ** > > ** ** > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/b8aca7e8/attachment.html From steveayre at gmail.com Thu Oct 24 04:40:50 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Oct 2013 01:40:50 +0100 Subject: [Freeswitch-users] codec transcodation not working In-Reply-To: References: Message-ID: As well as loading the modules you need to list the supported incoming and outgoing codecs on the sip profile. My guess is speex isn't in that list. "sofia status profile " will show the settings in use On Tuesday, October 22, 2013, neeraj.p wrote: > Hey, > > I tried to establish audio call between two clients having different codec > supports . I expected that transcoding would take place. > But I am getting a codec negotiation error . > > Here is the information about client I am using > > leg A : > suppoerted codec : speex > > leg B: > supported codec : G722,PCMU,PCMA,GSM,OPUS > > > Here is some lines from my freeswitch logs > > 2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending > early media > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec > Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200] > 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in > SDP. Disable 2833 dtmf and switch to INFO > 2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR. > SDP: > v=0 > o=Zoiper 0 0 IN IP4 120.63.38.94 > s=Zoiper > c=IN IP4 120.63.38.94 > t=0 0 > m=audio 11525 RTP/AVP 110 > a=rtpmap:110 speex/8000 > > Here is some info about my freeswitch > > version - 1.4 beta > > codecs supported - codec,ADPCM (IMA),mod_spandsp > codec,AMR,mod_amr > codec,B64 (STANDARD),mod_b64 > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_spandsp > codec,G.723.1 6.3k,mod_g723_1 > codec,G.726 16k,mod_spandsp > codec,G.726 16k (AAL2),mod_spandsp > codec,G.726 24k,mod_spandsp > codec,G.726 24k (AAL2),mod_spandsp > codec,G.726 32k,mod_spandsp > codec,G.726 32k (AAL2),mod_spandsp > codec,G.726 40k,mod_spandsp > codec,G.726 40k (AAL2),mod_spandsp > codec,G.729,mod_g729 > codec,GSM,mod_spandsp > codec,H.261 Video (passthru),mod_h26x > codec,H.263 Video (passthru),mod_h26x > codec,H.263+ Video (passthru),mod_h26x > codec,H.263++ Video (passthru),mod_h26x > codec,H.264 Video (passthru),mod_h26x > codec,LPC-10,mod_spandsp > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE > codec,Speex,mod_speex > codec,VP8 Video (passthru),mod_vp8 > > My sip_profile configurations > > > > > > > I also tried with inbound-late-negotiaion=false . But still getting codec > negotiation error. > > Please help . > > > > Regards, > Neeraj > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/eaba8a31/attachment.html From steveayre at gmail.com Thu Oct 24 04:44:27 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Oct 2013 01:44:27 +0100 Subject: [Freeswitch-users] Forwarding SIP sessions In-Reply-To: References: <5267CE33.3030700@digitalmail.com> Message-ID: Also note not all clients will support it. FreeSWITCH doesn't automatically, you need to configure it to follow it or handle it in dialplan where you can perform checks that you're happy to follow the new Location. The reason is you can get redirected anywhere. Unlike HTTP which is free telecoms is not, and there is a risk of being silently redirected to a premium rate number. On Wednesday, October 23, 2013, Andrew Cassidy wrote: > You mean like this? > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect > > > On 23 October 2013 14:51, Regis M > > wrote: > >> +1 >> I'm very interested by this feature. >> >> >> >> 2013/10/23 Alex Lake > 'alex at digitalmail.com');>> >> >>> Is it possible to get Freeswitch to send a re-invite message in order to >>> do the equivalent of "I'm not accepting this call, try xx.xx.xx.xx"? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >> 'consulting at freeswitch.org');> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >> 'FreeSWITCH-users at lists.freeswitch.org');> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org > 'consulting at freeswitch.org');> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org > 'FreeSWITCH-users at lists.freeswitch.org');> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 > 100 960 *F > *03300 100 961 > *E * > andrew at cassidywebservices.co.uk 'andrew at cassidywebservices.co.uk');> > *W * > www.cassidywebservices.co.uk > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/4a2f4e73/attachment.html From lists at telefaks.de Thu Oct 24 05:31:32 2013 From: lists at telefaks.de (Peter Steinbach) Date: Thu, 24 Oct 2013 03:31:32 +0200 Subject: [Freeswitch-users] loopback_bowout In-Reply-To: <93392EFB-3CC9-4B00-B09A-A7A1449E00B8@freeswitch.org> References: <5266DF4F.2000002@telefaks.de> <93392EFB-3CC9-4B00-B09A-A7A1449E00B8@freeswitch.org> Message-ID: <52687874.6050700@telefaks.de> I updated today, but I still have 4 call legs with no audio when routing from external to external. When bridging from internal to external, it reduces to 2 legs on answer with the same dialplan and audio is fine. Peter On 10/22/13 22:36, Brian West wrote: > What git rev are you on because we just fixed this last week! > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > On Oct 22, 2013, at 3:25 PM, Peter Steinbach wrote: > >> Hello, >> >> I have a strange behaviour with forwarding a call from an external gateway to another number on the same gateway. >> ? when I forward a call from an internal number to an external number via loopback, I firstly get 4 channels. After answering the call, I have 2 channels remaining and audio is fine >> ? when I forward a call from an _external_ number to an external number via loopback, I firstly get 4 channels. After answering the call, I still have 4 channels remaining and audio is _not_ there. Freeswitch is dialling via "external" profile via port 5080. I also tried to set "loopback_bowout=true", but this did not change anything. >> The dialstring is: >> The dialplan for both scenarios is the same. >> Anybody had the same issue and knows how to overcome this? >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> >> mailto:lists >> (att) telefaks.de >> Internet: >> www.telefaks.de >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From karl at xtronics.com Thu Oct 24 05:56:45 2013 From: karl at xtronics.com (Karl Schmidt) Date: Wed, 23 Oct 2013 20:56:45 -0500 Subject: [Freeswitch-users] xmllint and default config Message-ID: <52687E5D.4050900@xtronics.com> Complains about a bit found in /etc/autoload_configs/abstraction.conf $ xmllint /var/log/freeswitch/freeswitch.xml.fsxml /var/log/freeswitch/freeswitch.xml.fsxml:269: parser error : Unescaped '<' not allowed in attributes values References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> Message-ID: 100rel is not going to fix not getting a packet. On Wed, Oct 23, 2013 at 4:10 PM, Michael Jerris wrote: > It's all code, of course there are things that can be done. > > On Oct 23, 2013, at 4:57 PM, Abaci wrote: > > > Is there anything that be done to get it fixed? > > I have had issues lately with some phones not getting the 183 and > > therefor not playing ringback (changing to tcp seems to help but would > > prefer not have to do that). > > > > On 10/23/2013 4:50 PM, Michael Jerris wrote: > >> Those comments below are accurate to my knowledge. > >> > >> On Oct 23, 2013, at 4:33 PM, Abaci wrote: > >> > >>> Can someone please confirm if 100rel is fixed and stable in FreeSWITCH? > >>> I know that the ticket was closed but if you read the last comment by > >>> Mike Jerris "this issue is fixed in svn now (at least with 100rel > >>> disabled). 100rel support is a large known issue and needs to be fixed > >>> upstream in the sofia-sip library" it only says that when it's disabled > >>> it's not crashing. in the default config in the comments it says "There > >>> are known issues (asserts and segfaults) when 100rel is enabled. It is > >>> not recommended to enable 100rel at this time." > >>> I would appreciate if anyone that actually knows can confirm if it's > >>> fixed, if not is there anything that can be done to get it fixed. > >>> > >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: > >>>> Since the ticket is closed, I guess it works just fine. > >>>> ________________________________ > >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [ > mike.burlingame at me.com] > >>>> Skickat: den 24 augusti 2012 21:48 > >>>> Till: FreeSWITCH Users Help > >>>> ?mne: [Freeswitch-users] enable-100rel question > >>>> > >>>> I was looking to turn on 100rel however based on the below info from > the wiki I am not sure I want to run the risk of it crashing FS is this > information still accurate? > >>>> > >>>> > >>>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files > >>>> enable-100rel > >>>> This enable support for 100rel (100% reliability - PRACK message as > defined in RFC3262) This fixes a > problem with SIP where provisional messages like "180 Ringing" are not > ACK'd and therefore could be dropped over a poor connection without > retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, > see FSCORE-392. > >>>> > >>>> > >>>> > >>>> !DSPAM:5037d87c32761841043968! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/1c28cfa6/attachment.html From yehavi.bourvine at gmail.com Thu Oct 24 08:12:42 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 24 Oct 2013 07:12:42 +0300 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: Sorry, but I am still puzzled... It has been fixed before 1.2.14, but the problem is only at the 1.2 branch... So does 1.2.14 has the fix or not? BTW, I had the same problem with the HEAD when I tried it a few months ago... Thanks! __Yehavi: 2013/10/23 Ken Rice > The problem was only in the stable branch, (that would be the 1.2 line) > and was fixed prior to the 1.2.14 release... > > > On 10/23/13 9:03 AM, "Yehavi Bourvine" wrote: > > HI, > > A question: Is it solved on 1.2.14 or only on HEAD? I tried today 1.2.14 > and failed with the same sympthoms... > > Thanks, __Yehavi: > > > 2013/10/14 Anthony Minessale > > It is for many. It depends on the general goals etc. > The problem you describe was only on stable and all the symptoms, not > shutting down, channels count being wrong, zombies etc are all the same 1 > problem now fixed. > > > > On Mon, Oct 14, 2013 at 11:33 AM, Stephen Wilde > wrote: > > I have tried only branch v1.2.stable not master. Is the master good for a > production environment? > > > On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > That's all the same thing...... > > Do you have this problem on master branch as well? > > > > On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde > wrote: > > Thank you for your advice. > I have tried the 1.2.stable but I have due to abandon this branch due to a > some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting > x sessions ..." messages. > I have seen that this issue (FS-5848) has already been solved so I can > try with 1.2.stable but I don't see any fix that can affect my issue. > > > On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > Try latest HEAD for a probable fix. > > > > On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde > wrote: > > Hi all, > we have an issue with our FreeSwitch box that to us seems to be > performance related. > The effect is that in FreeSwitch the number of session reaches the limit > we set in config as: > > > > The limit is reached independently of its value because when FreeSwitch is > in this state, the number of sessions grows indefinitely. > > We have tried to upgrade the hardware of the box moving from a Xeon 2 CPU > E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but with > this more powerful hardware it happens that the limit is reached with less > sessions. > It seems that performance are related to the speed of single core instead > of speed of the box. > Make sense? > > A example of "status" issued before the crash is: > > 14969852 session(s) since startup > 13765 session(s) - 538 out of max 1000 per sec > 30000 session(s) max > min idle cpu 0.00/36.00 > Current Stack Size/Max 240K/8192K > > > Any advice? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > G+ ClueCon : http://www.ss7.us/cluecon-gplus > FB ClueCon : http://www.ss7.us/cluecon-fb > G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus > FB FreeSWITCH : http://www.ss7.us/freeswitch-fb > Twitter : @FreeSWITCH_WIRE > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/f2f38a24/attachment-0001.html From avi at avimarcus.net Thu Oct 24 08:22:14 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 24 Oct 2013 04:22:14 +0000 Subject: [Freeswitch-users] SDReporter Integration with FS In-Reply-To: <52683ABC.2060202@gmail.com> References: <526666D6.6010901@gmail.com> <52683ABC.2060202@gmail.com> Message-ID: <00000141e8b2ce6c-a14a8634-e37e-4bcc-8aa7-e9d994d66af2-000000@email.amazonses.com> Thanks for reporting back! -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/f9af5707/attachment.html From sirimmfs at gmail.com Thu Oct 24 08:55:36 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 24 Oct 2013 15:55:36 +1100 Subject: [Freeswitch-users] Individual party recording Message-ID: Hi, Is it possible for me to record each party of a call individually? If I were to transfer the call a few times to different extensions, I would like store each party's conversation separately - is this feasible? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/2a87c9ba/attachment.html From allen.schultz at gmail.com Thu Oct 24 03:20:40 2013 From: allen.schultz at gmail.com (Allen Schultz) Date: Wed, 23 Oct 2013 17:20:40 -0600 Subject: [Freeswitch-users] disconnect after 30-60 with freeswitch and polycom phones on same network Message-ID: <526859C8.4050308@gmail.com> I'm having an issue with the 30-60 second disconnect on both calls (from outside sip provider) and local voicemail (seperate incident, but related). Below I have some pastebin logs of some information with some preliminary steps with the #freeswitch irc group. Event: with sofia siptrace on sofia status sofia status profile internal sofia status profile external Polycom soundpoint ip 430 rev a configured polycoms to go directly to 192.168.123.6 Any help will be appreciated. Allen -------------- next part -------------- A non-text attachment was scrubbed... Name: allen_schultz.vcf Type: text/x-vcard Size: 181 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131023/8b80be00/attachment.vcf From gerald.weber at besharp.at Thu Oct 24 11:11:14 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Thu, 24 Oct 2013 07:11:14 +0000 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: Message-ID: Thanks, but youtube says this video is private. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Cal Leeming [Simplicity Media Ltd] Gesendet: Mittwoch, 23. Oktober 2013 23:38 An: FreeSWITCH Users Help Cc: freeswitch-dev at lists.freeswitch.org; freeswitch-cluecon at lists.freeswitch.org Betreff: Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern For those that missed it, you can watch the whole thing here; http://www.youtube.com/watch?v=raXkHi_uGF8 Slides here; https://www.dropbox.com/s/hp5fj7e7o1mdnyt/Auto%20provisioning%20sucks.pptx Cal On Tue, Oct 22, 2013 at 7:36 PM, Ken Rice > wrote: ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM Eastern Live! Cal Lemming, Will be making a 0day disclosure! Don't miss this exclusive presentation for the FreeSwitch community! This is one you will not want to miss! Join us on the FreeSWITCH Bridge for this Special Presentation! For conference call access information visit http://ss7.us/call888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://www.ss7.us/cluecon-gplus FB ClueCon : http://www.ss7.us/cluecon-fb G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus FB FreeSWITCH : http://www.ss7.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/0a6ae211/attachment.html From neeraj.p at directi.com Thu Oct 24 11:34:19 2013 From: neeraj.p at directi.com (neeraj.p) Date: Thu, 24 Oct 2013 13:04:19 +0530 Subject: [Freeswitch-users] codec transcodation not working Message-ID: <2DFB081E-8E37-4A9D-9BC5-B776426643DD@directi.com> Hey Steven Here is the result of command - sofia status profile internal I can see speex in the list. freeswitch at 127.0.0.1:5222 at internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 172.16.138.74,172.16.138.74 Dialplan XML Context public Challenge Realm auto_from RTP-IP 172.16.138.74 SIP-IP 172.16.138.74 URL sip:mod_sofia at 172.16.138.74:5060 BIND-URL sip:mod_sofia at 172.16.138.74:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU,GSM,PCMA,speex at 8000h@20i CODECS OUT PCMU,GSM,PCMA,speex at 8000h@20i TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 --Neeraj From miha at softnet.si Thu Oct 24 11:45:04 2013 From: miha at softnet.si (Miha) Date: Thu, 24 Oct 2013 09:45:04 +0200 Subject: [Freeswitch-users] Parsing from sip header In-Reply-To: <1809852E-41E5-4E15-BC30-D36703331C88@jerris.com> References: <526783C8.7070408@softnet.si> <1809852E-41E5-4E15-BC30-D36703331C88@jerris.com> Message-ID: <5268D000.6020507@softnet.si> Hi Michael, you are right. When I was adding header with opensips I did not add prefex "X-". Tnx!!!! Miha Dne 10/23/2013 2:04 PM, pi?e Michael Jerris: > I don't think we currently parse the "Moved" Header (I've never actually seen that header before, is it standard?). We do parse x-headers > > On Oct 23, 2013, at 4:07 AM, Miha wrote: > >> Hi, >> >> I am sending to FS some attribute in header which I have added on proxy side. How can I parse this header, to get this data and send it further? >> >> I can not see it in varibles which are printed with info application. >> >> I added: >> Moved: 38618108751. >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From miha at softnet.si Thu Oct 24 11:47:17 2013 From: miha at softnet.si (Miha) Date: Thu, 24 Oct 2013 09:47:17 +0200 Subject: [Freeswitch-users] 481 Call Does Not Exist Message-ID: <5268D085.6060509@softnet.si> Hi, I need a little help with figuring out why FS sends "481". This is happing when call has been forward (302). All calls are relayed throught opensips as I am using it for registrations and load_balancing. http://pastebin.freeswitch.org/21555 tnx! miha From rafal.gwizdala at gmail.com Thu Oct 24 12:24:21 2013 From: rafal.gwizdala at gmail.com (Rafal Gwizdala) Date: Thu, 24 Oct 2013 10:24:21 +0200 Subject: [Freeswitch-users] mod_shout in Windows distribution? Message-ID: Hi, is there a reason why mod_shout is not included with Windows binary distribution? Best regards RG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/2f5b3ea9/attachment-0001.html From alex at digitalmail.com Thu Oct 24 12:54:24 2013 From: alex at digitalmail.com (Alex Lake) Date: Thu, 24 Oct 2013 09:54:24 +0100 Subject: [Freeswitch-users] Forwarding SIP sessions In-Reply-To: References: <5267CE33.3030700@digitalmail.com> Message-ID: <5268E040.30307@digitalmail.com> That might be what I need. Can you just clarify - does this mean that a redirected call will then require no further involvement from the redirecting FS server? I can try this out, though. Thanks for the tip. > You mean like this? > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect > > > On 23 October 2013 14:51, Regis M > wrote: > > +1 > I'm very interested by this feature. > > > > 2013/10/23 Alex Lake > > > Is it possible to get Freeswitch to send a re-invite message > in order to > do the equivalent of "I'm not accepting this call, try > xx.xx.xx.xx"? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/305640c8/attachment.html From steveayre at gmail.com Thu Oct 24 13:15:24 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Oct 2013 10:15:24 +0100 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> Message-ID: There's various ways you can help... If you can test the feature and still get it to crash that helps (it shows us there is indeed still a problem). If you can identify an easy way to reproduce it, that helps (it provides a testcase). If you can identify what part if the code is crashing (logs & backtrace) that helps (we can see where the bug lies). If you can spot the problem in the code and perhaps even contribute a patch that'd be enormously helpful This appears to be the latest Jira bug ticket. From the last comment there it looks like the issue lies within the Sofia-SIP library. We could patch it and push a fix upstream, or perhaps this has already been corrected since the ticket is over 4 years old. I guess the main reason no-one knows is that with that warning no-one has dared to test it on a production server. :) -Steve On Wednesday, October 23, 2013, Michael Jerris wrote: > It's all code, of course there are things that can be done. > > On Oct 23, 2013, at 4:57 PM, Abaci wrote: > > > Is there anything that be done to get it fixed? > > I have had issues lately with some phones not getting the 183 and > > therefor not playing ringback (changing to tcp seems to help but would > > prefer not have to do that). > > > > On 10/23/2013 4:50 PM, Michael Jerris wrote: > >> Those comments below are accurate to my knowledge. > >> > >> On Oct 23, 2013, at 4:33 PM, Abaci wrote: > >> > >>> Can someone please confirm if 100rel is fixed and stable in FreeSWITCH? > >>> I know that the ticket was closed but if you read the last comment by > >>> Mike Jerris "this issue is fixed in svn now (at least with 100rel > >>> disabled). 100rel support is a large known issue and needs to be fixed > >>> upstream in the sofia-sip library" it only says that when it's disabled > >>> it's not crashing. in the default config in the comments it says "There > >>> are known issues (asserts and segfaults) when 100rel is enabled. It is > >>> not recommended to enable 100rel at this time." > >>> I would appreciate if anyone that actually knows can confirm if it's > >>> fixed, if not is there anything that can be done to get it fixed. > >>> > >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: > >>>> Since the ticket is closed, I guess it works just fine. > >>>> ________________________________ > >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [ > mike.burlingame at me.com] > >>>> Skickat: den 24 augusti 2012 21:48 > >>>> Till: FreeSWITCH Users Help > >>>> ?mne: [Freeswitch-users] enable-100rel question > >>>> > >>>> I was looking to turn on 100rel however based on the below info from > the wiki I am not sure I want to run the risk of it crashing FS is this > information still accurate? > >>>> > >>>> > >>>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files > >>>> enable-100rel > >>>> This enable support for 100rel (100% reliability - PRACK message as > defined in RFC3262) This fixes a > problem with SIP where provisional messages like "180 Ringing" are not > ACK'd and therefore could be dropped over a poor connection without > retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, > see FSCORE-392. > >>>> > >>>> > >>>> > >>>> !DSPAM:5037d87c32761841043968! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/815751cb/attachment.html From steveayre at gmail.com Thu Oct 24 13:17:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Oct 2013 10:17:52 +0100 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> Message-ID: What might is using SIP over TCP, if that's an option to you. If packets are being lost you're still going to get audio problems so it's worth investigating why there's packet loss in the first place. On 24 October 2013 03:15, Anthony Minessale wrote: > 100rel is not going to fix not getting a packet. > > > > On Wed, Oct 23, 2013 at 4:10 PM, Michael Jerris wrote: > >> It's all code, of course there are things that can be done. >> >> On Oct 23, 2013, at 4:57 PM, Abaci wrote: >> >> > Is there anything that be done to get it fixed? >> > I have had issues lately with some phones not getting the 183 and >> > therefor not playing ringback (changing to tcp seems to help but would >> > prefer not have to do that). >> > >> > On 10/23/2013 4:50 PM, Michael Jerris wrote: >> >> Those comments below are accurate to my knowledge. >> >> >> >> On Oct 23, 2013, at 4:33 PM, Abaci wrote: >> >> >> >>> Can someone please confirm if 100rel is fixed and stable in >> FreeSWITCH? >> >>> I know that the ticket was closed but if you read the last comment by >> >>> Mike Jerris "this issue is fixed in svn now (at least with 100rel >> >>> disabled). 100rel support is a large known issue and needs to be fixed >> >>> upstream in the sofia-sip library" it only says that when it's >> disabled >> >>> it's not crashing. in the default config in the comments it says >> "There >> >>> are known issues (asserts and segfaults) when 100rel is enabled. It is >> >>> not recommended to enable 100rel at this time." >> >>> I would appreciate if anyone that actually knows can confirm if it's >> >>> fixed, if not is there anything that can be done to get it fixed. >> >>> >> >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: >> >>>> Since the ticket is closed, I guess it works just fine. >> >>>> ________________________________ >> >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [ >> mike.burlingame at me.com] >> >>>> Skickat: den 24 augusti 2012 21:48 >> >>>> Till: FreeSWITCH Users Help >> >>>> ?mne: [Freeswitch-users] enable-100rel question >> >>>> >> >>>> I was looking to turn on 100rel however based on the below info from >> the wiki I am not sure I want to run the risk of it crashing FS is this >> information still accurate? >> >>>> >> >>>> >> >>>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >> >>>> enable-100rel >> >>>> This enable support for 100rel (100% reliability - PRACK message as >> defined in RFC3262) This fixes a >> problem with SIP where provisional messages like "180 Ringing" are not >> ACK'd and therefore could be dropped over a poor connection without >> retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, >> see FSCORE-392. >> >>>> >> >>>> >> >>>> >> >>>> !DSPAM:5037d87c32761841043968! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/9dfa2731/attachment-0001.html From hardyanto.donny at gmail.com Thu Oct 24 13:22:51 2013 From: hardyanto.donny at gmail.com (Donny Hardyanto) Date: Thu, 24 Oct 2013 16:22:51 +0700 Subject: [Freeswitch-users] codec transcodation not working In-Reply-To: References: Message-ID: In the client speex 8 is 110, in FS speex 8 is 98. So it was mismatch. What I know that speex 98 not 110. So why do you get 110? Donny On Tue, Oct 22, 2013 at 9:23 PM, neeraj.p wrote: > Here is the full log when leg A calls leg B > http://pastebin.com/0JAZQpe7 > > Here is the log when leg B calls leg A > http://pastebin.com/8Q5abKEC > > I can see different errors in these two cases . > > Regards, > Neeraj > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/b5cd90f8/attachment.html From idokan at gmail.com Thu Oct 24 14:12:11 2013 From: idokan at gmail.com (ik) Date: Thu, 24 Oct 2013 12:12:11 +0200 Subject: [Freeswitch-users] bind_digit_action constantly reporting syntax error Message-ID: Hello, I'ved created the following dialplan (fs v1.2.14) : bind_digit_action constantly reporting Syntax Error, USAGE ,,< string>[,][,][,] The original aim of the program is the capture the numbers after 200*, and bridge leg-a to the new number, but first disconnecting current leg-b. When using execute_extension with "exec_after_bridge_app=execute_extension "exec_after_bridge_arg=digits ${bridging_capture}" It makes leg-b constantly ring after the call hangup. So does the transfer action that is commented out. So what am I doing wrong here ? Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/55f51f34/attachment.html From shahzad.bhatti at g-r-v.com Thu Oct 24 15:31:17 2013 From: shahzad.bhatti at g-r-v.com (Shahzad_Sab) Date: Thu, 24 Oct 2013 04:31:17 -0700 (PDT) Subject: [Freeswitch-users] Calling stored procedure in freeswitch database or using luasql In-Reply-To: References: <1369129093912-7590895.post@n2.nabble.com> <1382524377956-7595937.post@n2.nabble.com> Message-ID: <1382614277535-7595965.post@n2.nabble.com> Thanks for the replay but when i try assert( con:execute("call csp_test();", function(qrow) for key, val in pairs(qrow) do myrow[key] = val found = found + 1 end end ) ) *i got error:* LuaSQL: Error executing query. MySQL: PROCEDURE test.csp_test can't return a result set in the given context and when i try assert( con:query("call csp_test();", function(qrow) for key, val in pairs(qrow) do myrow[key] = val found = found + 1 end end ) ) *i got Error:* attempt to call method 'query' (a nil value) any further help. ----- Regards Shahzad Bhatti +92-300-433-4033 +92-321-401-8917 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calling-stored-procedure-in-freeswitch-database-or-using-luasql-tp7590895p7595965.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jpyle at fidelityvoice.com Thu Oct 24 16:28:16 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Thu, 24 Oct 2013 08:28:16 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> Message-ID: I have a few instances of 1.2.11-n20130816T111409Z-1~wheezy+1+git~20130816T135609Z~64ade54a73 with 100rel enabled and I haven't encountered any problems so far. - Jeff On Thu, Oct 24, 2013 at 5:17 AM, Steven Ayre wrote: > What might is using SIP over TCP, if that's an option to you. > > If packets are being lost you're still going to get audio problems so it's > worth investigating why there's packet loss in the first place. > > > On 24 October 2013 03:15, Anthony Minessale wrote: > >> 100rel is not going to fix not getting a packet. >> >> >> >> On Wed, Oct 23, 2013 at 4:10 PM, Michael Jerris wrote: >> >>> It's all code, of course there are things that can be done. >>> >>> On Oct 23, 2013, at 4:57 PM, Abaci wrote: >>> >>> > Is there anything that be done to get it fixed? >>> > I have had issues lately with some phones not getting the 183 and >>> > therefor not playing ringback (changing to tcp seems to help but would >>> > prefer not have to do that). >>> > >>> > On 10/23/2013 4:50 PM, Michael Jerris wrote: >>> >> Those comments below are accurate to my knowledge. >>> >> >>> >> On Oct 23, 2013, at 4:33 PM, Abaci wrote: >>> >> >>> >>> Can someone please confirm if 100rel is fixed and stable in >>> FreeSWITCH? >>> >>> I know that the ticket was closed but if you read the last comment by >>> >>> Mike Jerris "this issue is fixed in svn now (at least with 100rel >>> >>> disabled). 100rel support is a large known issue and needs to be >>> fixed >>> >>> upstream in the sofia-sip library" it only says that when it's >>> disabled >>> >>> it's not crashing. in the default config in the comments it says >>> "There >>> >>> are known issues (asserts and segfaults) when 100rel is enabled. It >>> is >>> >>> not recommended to enable 100rel at this time." >>> >>> I would appreciate if anyone that actually knows can confirm if it's >>> >>> fixed, if not is there anything that can be done to get it fixed. >>> >>> >>> >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: >>> >>>> Since the ticket is closed, I guess it works just fine. >>> >>>> ________________________________ >>> >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>> freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [ >>> mike.burlingame at me.com] >>> >>>> Skickat: den 24 augusti 2012 21:48 >>> >>>> Till: FreeSWITCH Users Help >>> >>>> ?mne: [Freeswitch-users] enable-100rel question >>> >>>> >>> >>>> I was looking to turn on 100rel however based on the below info >>> from the wiki I am not sure I want to run the risk of it crashing FS is >>> this information still accurate? >>> >>>> >>> >>>> >>> >>>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >>> >>>> enable-100rel >>> >>>> This enable support for 100rel (100% reliability - PRACK message as >>> defined in RFC3262) This fixes a >>> problem with SIP where provisional messages like "180 Ringing" are not >>> ACK'd and therefore could be dropped over a poor connection without >>> retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, >>> see FSCORE-392. >>> >>>> >>> >>>> >>> >>>> >>> >>>> !DSPAM:5037d87c32761841043968! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/a181bdaf/attachment-0001.html From nbhatti at gmail.com Thu Oct 24 17:03:18 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 24 Oct 2013 16:03:18 +0300 Subject: [Freeswitch-users] Limit app using bgapi Message-ID: <52691A96.4080009@gmail.com> Hi, Is there a way to set limit app using bgapi? I am originating calls from ESL, and want to rate limit the CPS for outgoing gateways. Or if there is any other way around? -- Thanks, Muhammad Naseer Bhatti From abaci64 at gmail.com Thu Oct 24 17:10:54 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 24 Oct 2013 09:10:54 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> Message-ID: <52691C5E.9070600@gmail.com> it seems to fix the problem in my testing, as it will resend the 183 to the phone. On 10/23/2013 10:15 PM, Anthony Minessale wrote: > 100rel is not going to fix not getting a packet. > > > > On Wed, Oct 23, 2013 at 4:10 PM, Michael Jerris > wrote: > > It's all code, of course there are things that can be done. > > On Oct 23, 2013, at 4:57 PM, Abaci > wrote: > > > Is there anything that be done to get it fixed? > > I have had issues lately with some phones not getting the 183 and > > therefor not playing ringback (changing to tcp seems to help but > would > > prefer not have to do that). > > > > On 10/23/2013 4:50 PM, Michael Jerris wrote: > >> Those comments below are accurate to my knowledge. > >> > >> On Oct 23, 2013, at 4:33 PM, Abaci > wrote: > >> > >>> Can someone please confirm if 100rel is fixed and stable in > FreeSWITCH? > >>> I know that the ticket was closed but if you read the last > comment by > >>> Mike Jerris "this issue is fixed in svn now (at least with 100rel > >>> disabled). 100rel support is a large known issue and needs to > be fixed > >>> upstream in the sofia-sip library" it only says that when it's > disabled > >>> it's not crashing. in the default config in the comments it > says "There > >>> are known issues (asserts and segfaults) when 100rel is > enabled. It is > >>> not recommended to enable 100rel at this time." > >>> I would appreciate if anyone that actually knows can confirm > if it's > >>> fixed, if not is there anything that can be done to get it fixed. > >>> > >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: > >>>> Since the ticket is closed, I guess it works just fine. > >>>> ________________________________ > >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > > [freeswitch-users-bounces at lists.freeswitch.org > ] f?r Mike > Burlingame [mike.burlingame at me.com ] > >>>> Skickat: den 24 augusti 2012 21:48 > >>>> Till: FreeSWITCH Users Help > >>>> ?mne: [Freeswitch-users] enable-100rel question > >>>> > >>>> I was looking to turn on 100rel however based on the below > info from the wiki I am not sure I want to run the risk of it > crashing FS is this information still accurate? > >>>> > >>>> > >>>>> From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files > >>>> enable-100rel > >>>> This enable support for 100rel (100% reliability - PRACK > message as defined in RFC3262) > This fixes a problem with SIP where provisional messages like "180 > Ringing" are not ACK'd and therefore could be dropped over a poor > connection without retransmission. *2009-07-08:* Enabling this may > cause FreeSWITCH to crash, see > FSCORE-392. > >>>> > >>>> > >>>> > >>>> !DSPAM:5037d87c32761841043968! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/f8383663/attachment.html From abaci64 at gmail.com Thu Oct 24 17:14:44 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 24 Oct 2013 09:14:44 -0400 Subject: [Freeswitch-users] enable-100rel question In-Reply-To: References: <03F639A3-5709-462F-884E-348F18074DBB@me.com> <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se> <526832AD.3090202@gmail.com> <2C8ED3D0-E05D-4518-8B93-F2C733ECA6C2@jerris.com> <52683842.3030606@gmail.com> <579514A8-2BC8-45D9-BA59-087A0548289A@jerris.com> Message-ID: <52691D44.6000402@gmail.com> As stated in my original email, tcp seems to help but I would prefer not having to move everyone over to tcp. I think I'll enable it on one server and put those that have the problem on that server. If there is someone that know there is indeed a problem in the sofia lib and can fix it I would offer a bounty for that. On 10/24/2013 5:17 AM, Steven Ayre wrote: > What might is using SIP over TCP, if that's an option to you. > > If packets are being lost you're still going to get audio problems so > it's worth investigating why there's packet loss in the first place. > > > On 24 October 2013 03:15, Anthony Minessale > > wrote: > > 100rel is not going to fix not getting a packet. > > > > On Wed, Oct 23, 2013 at 4:10 PM, Michael Jerris > wrote: > > It's all code, of course there are things that can be done. > > On Oct 23, 2013, at 4:57 PM, Abaci > wrote: > > > Is there anything that be done to get it fixed? > > I have had issues lately with some phones not getting the > 183 and > > therefor not playing ringback (changing to tcp seems to help > but would > > prefer not have to do that). > > > > On 10/23/2013 4:50 PM, Michael Jerris wrote: > >> Those comments below are accurate to my knowledge. > >> > >> On Oct 23, 2013, at 4:33 PM, Abaci > wrote: > >> > >>> Can someone please confirm if 100rel is fixed and stable > in FreeSWITCH? > >>> I know that the ticket was closed but if you read the last > comment by > >>> Mike Jerris "this issue is fixed in svn now (at least with > 100rel > >>> disabled). 100rel support is a large known issue and needs > to be fixed > >>> upstream in the sofia-sip library" it only says that when > it's disabled > >>> it's not crashing. in the default config in the comments > it says "There > >>> are known issues (asserts and segfaults) when 100rel is > enabled. It is > >>> not recommended to enable 100rel at this time." > >>> I would appreciate if anyone that actually knows can > confirm if it's > >>> fixed, if not is there anything that can be done to get it > fixed. > >>> > >>> On 8/24/2012 5:33 PM, Peter Olsson wrote: > >>>> Since the ticket is closed, I guess it works just fine. > >>>> ________________________________ > >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org > > [freeswitch-users-bounces at lists.freeswitch.org > ] f?r > Mike Burlingame [mike.burlingame at me.com > ] > >>>> Skickat: den 24 augusti 2012 21:48 > >>>> Till: FreeSWITCH Users Help > >>>> ?mne: [Freeswitch-users] enable-100rel question > >>>> > >>>> I was looking to turn on 100rel however based on the > below info from the wiki I am not sure I want to run the risk > of it crashing FS is this information still accurate? > >>>> > >>>> > >>>>> From > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files > >>>> enable-100rel > >>>> This enable support for 100rel (100% reliability - PRACK > message as defined in > RFC3262) This fixes a > problem with SIP where provisional messages like "180 Ringing" > are not ACK'd and therefore could be dropped over a poor > connection without retransmission. *2009-07-08:* Enabling this > may cause FreeSWITCH to crash, see > FSCORE-392. > >>>> > >>>> > >>>> > >>>> !DSPAM:5037d87c32761841043968! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/6f4b1bc6/attachment-0001.html From erhuabushuo at gmail.com Thu Oct 24 12:34:55 2013 From: erhuabushuo at gmail.com (erhuabushuo) Date: Thu, 24 Oct 2013 16:34:55 +0800 Subject: [Freeswitch-users] issue of cdr_mongodb Message-ID: <5268DBAF.2090102@gmail.com> established connection, after finished call, when it try to insert cdr to mongodb, I got this error '2013-10-23 17:21:29.745772 [ERR] mod_cdr_mongodb.c:393 mongo_insert: (error code 15)' From krice at freeswitch.org Thu Oct 24 18:05:45 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 24 Oct 2013 09:05:45 -0500 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: Message-ID: The video has been marked private at vendor request... On 10/24/13 2:11 AM, "Gerald Weber" wrote: > Thanks, but youtube says this video is private. > > Von: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Cal > Leeming [Simplicity Media Ltd] > Gesendet: Mittwoch, 23. Oktober 2013 23:38 > An: FreeSWITCH Users Help > Cc: freeswitch-dev at lists.freeswitch.org; > freeswitch-cluecon at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special > Security Edition! Wed Oct 23rd @ 1PM Eastern > > > For those that missed it, you can watch the whole thing here; > > http://www.youtube.com/watch?v=raXkHi_uGF8 > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/0cb7e341/attachment.html From abaci64 at gmail.com Thu Oct 24 18:13:22 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 24 Oct 2013 10:13:22 -0400 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: Message-ID: <52692B02.5070303@gmail.com> Are they doing something about it and just need some time, or are they just trying to silence it. On 10/24/2013 10:05 AM, Ken Rice wrote: > Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special > Security Edition! Wed Oct 23rd @ 1PM Eastern The video has been marked > private at vendor request... > > > On 10/24/13 2:11 AM, "Gerald Weber" wrote: > > Thanks, but youtube says this video is private. > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag > von *Cal Leeming [Simplicity Media Ltd] > *Gesendet:* Mittwoch, 23. Oktober 2013 23:38 > *An:* FreeSWITCH Users Help > *Cc:* freeswitch-dev at lists.freeswitch.org; > freeswitch-cluecon at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] [Special Announcement] ClueCon > Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern > > > For those that missed it, you can watch the whole thing here; > > http://www.youtube.com/watch?v=raXkHi_uGF8 > > _ > _-- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > G+ ClueCon : http://fs0.us/cluecon-gplus > FB ClueCon : http://fs0.us/cluecon-fb > G+ FreeSwitch : http://fs0.us/freeswitch-gplus > FB FreeSWITCH : http://fs0.us/freeswitch-fb > Twitter : @FreeSWITCH_WIRE > irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/db05ef1c/attachment.html From mike at jerris.com Thu Oct 24 18:40:48 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 24 Oct 2013 10:40:48 -0400 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: <52692B02.5070303@gmail.com> References: <52692B02.5070303@gmail.com> Message-ID: They are indeed working hard to very quickly address this issue, but it is going to take them a week or 2 to get a new firmware out and otherwise address the issue. We are in active communications with them about progress on the solution, including actively discussing and testing with them until late last night. On Oct 24, 2013, at 10:13 AM, Abaci wrote: > Are they doing something about it and just need some time, or are they just trying to silence it. > > On 10/24/2013 10:05 AM, Ken Rice wrote: >> The video has been marked private at vendor request... >> >> >> On 10/24/13 2:11 AM, "Gerald Weber" wrote: >> >> Thanks, but youtube says this video is private. >> >> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Cal Leeming [Simplicity Media Ltd] >> Gesendet: Mittwoch, 23. Oktober 2013 23:38 >> An: FreeSWITCH Users Help >> Cc: freeswitch-dev at lists.freeswitch.org; freeswitch-cluecon at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern >> >> >> For those that missed it, you can watch the whole thing here; >> >> http://www.youtube.com/watch?v=raXkHi_uGF8 >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> G+ ClueCon : http://fs0.us/cluecon-gplus >> FB ClueCon : http://fs0.us/cluecon-fb >> G+ FreeSwitch : http://fs0.us/freeswitch-gplus >> FB FreeSWITCH : http://fs0.us/freeswitch-fb >> Twitter : @FreeSWITCH_WIRE >> irc.freenode.net #freeswitch >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/a1c0a1ad/attachment-0001.html From moshe3t at gmail.com Thu Oct 24 18:47:17 2013 From: moshe3t at gmail.com (Moshe3t) Date: Thu, 24 Oct 2013 10:47:17 -0400 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: Message-ID: <526932F5.3010702@gmail.com> Hi I would suggest (I have Bcc'd Yealink as well) that as they have when a phone is reset to factory default it pops up on the screen of the phone local network option (DHCP/Static) and modifiable via phone keypad it should also ask encrypted provisioning or not and if encrypted is chosen it should let the end user put in the decryption key via phone keypad which will be available on the website of the provisioning server (assuming the its secure as in most cases might hold sip credes as well) so the end user will be able to setup and auto provision their phone without login to the phone web gui (ok! it won't zero touch it will be 1 touch provisioning, as if the doesn't have to touch the phone at all ;-) ) before submitting this approach to any vendors i would like to hear input and make sure the issue is addressed properly and hear if anyone have a better approach to fix this issue globally (at least with Yealink line ofproduct as they seem to be very cooperative and understanding in general especially when it comes to security, in hopes other companies will follow suit) Sincerely Moshe BT On 10/24/2013 10:05 AM, Ken Rice wrote: > Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special > Security Edition! Wed Oct 23rd @ 1PM Eastern The video has been marked > private at vendor request... > > > On 10/24/13 2:11 AM, "Gerald Weber" wrote: > > Thanks, but youtube says this video is private. > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag > von *Cal Leeming [Simplicity Media Ltd] > *Gesendet:* Mittwoch, 23. Oktober 2013 23:38 > *An:* FreeSWITCH Users Help > *Cc:* freeswitch-dev at lists.freeswitch.org; > freeswitch-cluecon at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] [Special Announcement] ClueCon > Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern > > > For those that missed it, you can watch the whole thing here; > > http://www.youtube.com/watch?v=raXkHi_uGF8 > > _ > _-- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > G+ ClueCon : http://fs0.us/cluecon-gplus > FB ClueCon : http://fs0.us/cluecon-fb > G+ FreeSwitch : http://fs0.us/freeswitch-gplus > FB FreeSWITCH : http://fs0.us/freeswitch-fb > Twitter : @FreeSWITCH_WIRE > irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/cfc728f3/attachment.html From max at nysolutions.com Thu Oct 24 19:17:17 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 24 Oct 2013 15:17:17 +0000 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: <526932F5.3010702@gmail.com> References: <526932F5.3010702@gmail.com> Message-ID: I wonder if Yealink phones have a certificate that can verify the MAC address being provisioned, the way the newer Snom do. http://wiki.snomone.com/index.php?title=Plug_and_Play_for_snom_phones#Pairing_the_phone_with_snom_ONE Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moshe3t Sent: Thursday, October 24, 2013 10:47 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern Hi I would suggest (I have Bcc'd Yealink as well) that as they have when a phone is reset to factory default it pops up on the screen of the phone local network option (DHCP/Static) and modifiable via phone keypad it should also ask encrypted provisioning or not and if encrypted is chosen it should let the end user put in the decryption key via phone keypad which will be available on the website of the provisioning server (assuming the its secure as in most cases might hold sip credes as well) so the end user will be able to setup and auto provision their phone without login to the phone web gui (ok! it won't zero touch it will be 1 touch provisioning, as if the doesn't have to touch the phone at all ;-) ) before submitting this approach to any vendors i would like to hear input and make sure the issue is addressed properly and hear if anyone have a better approach to fix this issue globally (at least with Yealink line ofproduct as they seem to be very cooperative and understanding in general especially when it comes to security, in hopes other companies will follow suit) Sincerely Moshe BT On 10/24/2013 10:05 AM, Ken Rice wrote: The video has been marked private at vendor request... On 10/24/13 2:11 AM, "Gerald Weber" wrote: Thanks, but youtube says this video is private. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Cal Leeming [Simplicity Media Ltd] Gesendet: Mittwoch, 23. Oktober 2013 23:38 An: FreeSWITCH Users Help Cc: freeswitch-dev at lists.freeswitch.org; freeswitch-cluecon at lists.freeswitch.org Betreff: Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern For those that missed it, you can watch the whole thing here; http://www.youtube.com/watch?v=raXkHi_uGF8 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/a628f4f9/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/a628f4f9/attachment-0001.jpg From anthony.minessale at gmail.com Thu Oct 24 19:36:15 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Oct 2013 10:36:15 -0500 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: What is the "same problem" exactly. If you are talking about the stuck channels issues stemming from calls being put on hold. It was fixed in stable HEAD. git commit: 4edd7b74223a7ddedd7dd65ed033b4d1da7fb6ba http://jira.freeswitch.org/browse/FS-5835 This is why we are not supposed to be discussing bugs in the mailing list. Bugs have a home called jira where they are tracked and tied to commits with git. On Wed, Oct 23, 2013 at 11:12 PM, Yehavi Bourvine wrote: > Sorry, but I am still puzzled... > It has been fixed before 1.2.14, but the problem is only at the 1.2 > branch... So does 1.2.14 has the fix or not? > > BTW, I had the same problem with the HEAD when I tried it a few months > ago... > > Thanks! __Yehavi: > > > 2013/10/23 Ken Rice > >> The problem was only in the stable branch, (that would be the 1.2 line) >> and was fixed prior to the 1.2.14 release... >> >> >> On 10/23/13 9:03 AM, "Yehavi Bourvine" wrote: >> >> HI, >> >> A question: Is it solved on 1.2.14 or only on HEAD? I tried today 1.2.14 >> and failed with the same sympthoms... >> >> Thanks, __Yehavi: >> >> >> 2013/10/14 Anthony Minessale >> >> It is for many. It depends on the general goals etc. >> The problem you describe was only on stable and all the symptoms, not >> shutting down, channels count being wrong, zombies etc are all the same 1 >> problem now fixed. >> >> >> >> On Mon, Oct 14, 2013 at 11:33 AM, Stephen Wilde >> wrote: >> >> I have tried only branch v1.2.stable not master. Is the master good for a >> production environment? >> >> >> On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >> That's all the same thing...... >> >> Do you have this problem on master branch as well? >> >> >> >> On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde >> wrote: >> >> Thank you for your advice. >> I have tried the 1.2.stable but I have due to abandon this branch due to >> a some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting >> x sessions ..." messages. >> I have seen that this issue (FS-5848) has already been solved so I can >> try with 1.2.stable but I don't see any fix that can affect my issue. >> >> >> On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >> Try latest HEAD for a probable fix. >> >> >> >> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde >> wrote: >> >> Hi all, >> we have an issue with our FreeSwitch box that to us seems to be >> performance related. >> The effect is that in FreeSwitch the number of session reaches the limit >> we set in config as: >> >> >> >> The limit is reached independently of its value because when FreeSwitch >> is in this state, the number of sessions grows indefinitely. >> >> We have tried to upgrade the hardware of the box moving from a Xeon 2 CPU >> E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but with >> this more powerful hardware it happens that the limit is reached with less >> sessions. >> It seems that performance are related to the speed of single core instead >> of speed of the box. >> Make sense? >> >> A example of "status" issued before the crash is: >> >> 14969852 session(s) since startup >> 13765 session(s) - 538 out of max 1000 per sec >> 30000 session(s) max >> min idle cpu 0.00/36.00 >> Current Stack Size/Max 240K/8192K >> >> >> Any advice? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> G+ ClueCon : http://www.ss7.us/cluecon-gplus >> FB ClueCon : http://www.ss7.us/cluecon-fb >> G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus >> FB FreeSWITCH : http://www.ss7.us/freeswitch-fb >> Twitter : @FreeSWITCH_WIRE >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/9fcda84e/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Oct 24 19:44:59 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 24 Oct 2013 16:44:59 +0100 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: <526932F5.3010702@gmail.com> Message-ID: This is one of the concepts I'd raised yesterday, generating a nonce using a unique value that only the vendor and the phone knows (i.e. a serial number, or in future phones, TPM/RSA module). However if the keys are ever compromised, then this would be rendered useless. Encryption is also good, but relies on a strong password which is difficult to type in on a phone and removes the concept of zero touch. To be honest, I don't think zero touch is going to be feasible if we want to keep security, because you are trusting that the keys stored at haven't been compromised. One touch would be a much better solution (there have been some great suggestions on that so far). Cal On Thu, Oct 24, 2013 at 4:17 PM, Moishe Grunstein wrote: > I wonder if Yealink phones have a certificate that can verify the MAC > address being provisioned, the way the newer Snom do.**** > > > http://wiki.snomone.com/index.php?title=Plug_and_Play_for_snom_phones#Pairing_the_phone_with_snom_ONE > **** > > ** ** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg at 01C72F94.9EE45D60] * > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Moshe3t > *Sent:* Thursday, October 24, 2013 10:47 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly > Special Security Edition! Wed Oct 23rd @ 1PM Eastern**** > > ** ** > > Hi > > I would suggest (I have Bcc'd Yealink as well) that as they have when a > phone is reset to factory default it pops up on the screen of the phone > local network option (DHCP/Static) and modifiable via phone keypad it > should also ask encrypted provisioning or not and if encrypted is chosen it > should let the end user put in the decryption key via phone keypad which > will be available on the website of the provisioning server (assuming the > its secure as in most cases might hold sip credes as well) so the end user > will be able to setup and auto provision their phone without login to the > phone web gui (ok! it won't zero touch it will be 1 touch provisioning, as > if the doesn't have to touch the phone at all ;-) ) > > before submitting this approach to any vendors i would like to hear input > and make sure the issue is addressed properly and hear if anyone have a > better approach to fix this issue globally (at least with Yealink line > ofproduct as they seem to be very cooperative and understanding in general > especially when it comes to security, in hopes other companies will follow > suit) > > Sincerely > > > Moshe BT > > > > > On 10/24/2013 10:05 AM, Ken Rice wrote:**** > > The video has been marked private at vendor request... > > > On 10/24/13 2:11 AM, "Gerald Weber" wrote:**** > > Thanks, but youtube says this video is private. > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *Im Auftrag von *Cal Leeming [Simplicity Media Ltd] > *Gesendet:* Mittwoch, 23. Oktober 2013 23:38 > *An:* FreeSWITCH Users Help > *Cc:* freeswitch-dev at lists.freeswitch.org; > freeswitch-cluecon at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly > Special Security Edition! Wed Oct 23rd @ 1PM Eastern > > > For those that missed it, you can watch the whole thing here; > > http://www.youtube.com/watch?v=raXkHi_uGF8**** > > * > *-- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > G+ ClueCon : http://fs0.us/cluecon-gplus > FB ClueCon : http://fs0.us/cluecon-fb > G+ FreeSwitch : http://fs0.us/freeswitch-gplus > FB FreeSWITCH : http://fs0.us/freeswitch-fb > Twitter : @FreeSWITCH_WIRE > irc.freenode.net #freeswitch > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > ** ** > > **** > > **** > > ** ** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > ** ** > > FreeSWITCH-users mailing list**** > > FreeSWITCH-users at lists.freeswitch.org**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/048f598e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/048f598e/attachment-0001.jpe From nneul at mst.edu Thu Oct 24 20:22:32 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 24 Oct 2013 11:22:32 -0500 Subject: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern In-Reply-To: References: <526932F5.3010702@gmail.com> Message-ID: <52694948.30805@mst.edu> Polycom has something like that functionality in current firmware - but I believe if you _upgrade_ to a current firmware, it doesn't work - it only works if it had that newer key installed in the factory image. -- Nathan On 10/24/2013 10:44 AM, Cal Leeming [Simplicity Media Ltd] wrote: > This is one of the concepts I'd raised yesterday, generating a nonce using a unique value that only the vendor and the > phone knows (i.e. a serial number, or in future phones, TPM/RSA module). However if the keys are ever compromised, then > this would be rendered useless. Encryption is also good, but relies on a strong password which is difficult to type in > on a phone and removes the concept of zero touch. > > To be honest, I don't think zero touch is going to be feasible if we want to keep security, because you are trusting > that the keys stored at haven't been compromised. One touch would be a much better > solution (there have been some great suggestions on that so far). > > Cal > > > On Thu, Oct 24, 2013 at 4:17 PM, Moishe Grunstein > wrote: > > I wonder if Yealink phones have a certificate that can verify the MAC address being provisioned, the way the newer > Snom do.____ > > http://wiki.snomone.com/index.php?title=Plug_and_Play_for_snom_phones#Pairing_the_phone_with_snom_ONE____ > > __ __ > > __ __ > > __ __ > > Thanks,____ > > __ __ > > Moishe Grunstein____ > > Tornado Computer Systems, Inc.____ > > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com *____ > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified____ > > cid:image001.jpg at 01C72F94.9EE45D60 ____ > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP > * Disaster Recovery * Network Security * Site Surveys * CMS____ > > __ __ > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] *On > Behalf Of *Moshe3t > *Sent:* Thursday, October 24, 2013 10:47 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM > Eastern____ > > __ __ > > Hi > > I would suggest (I have Bcc'd Yealink as well) that as they have when a phone is reset to factory default it pops up > on the screen of the phone local network option (DHCP/Static) and modifiable via phone keypad it should also ask > encrypted provisioning or not and if encrypted is chosen it should let the end user put in the decryption key via > phone keypad which will be available on the website of the provisioning server (assuming the its secure as in most > cases might hold sip credes as well) so the end user will be able to setup and auto provision their phone without > login to the phone web gui (ok! it won't zero touch it will be 1 touch provisioning, as if the doesn't have to touch > the phone at all ;-) ) > > before submitting this approach to any vendors i would like to hear input and make sure the issue is addressed > properly and hear if anyone have a better approach to fix this issue globally (at least with Yealink line ofproduct > as they seem to be very cooperative and understanding in general especially when it comes to security, in hopes > other companies will follow suit) > > Sincerely > > > Moshe BT > > > > > On 10/24/2013 10:05 AM, Ken Rice wrote:____ > > The video has been marked private at vendor request... > > > On 10/24/13 2:11 AM, "Gerald Weber" > wrote:____ > > Thanks, but youtube says this video is private. > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Cal Leeming [Simplicity Media Ltd] > *Gesendet:* Mittwoch, 23. Oktober 2013 23:38 > *An:* FreeSWITCH Users Help > *Cc:* freeswitch-dev at lists.freeswitch.org ; > freeswitch-cluecon at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ > 1PM Eastern > > > For those that missed it, you can watch the whole thing here; > > http://www.youtube.com/watch?v=raXkHi_uGF8____ > > _ > _-- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > G+ ClueCon : http://fs0.us/cluecon-gplus > FB ClueCon : http://fs0.us/cluecon-fb > G+ FreeSwitch : http://fs0.us/freeswitch-gplus > FB FreeSWITCH : http://fs0.us/freeswitch-fb > Twitter : @FreeSWITCH_WIRE > irc.freenode.net #freeswitch > > > > ____ > > _____________________________________________________________________________ > > Professional FreeSWITCH Consulting Services:____ > > consulting at freeswitch.org ____ > > http://www.freeswitchsolutions.com____ > > __ __ > > ____ > > ____ > > __ __ > > Official FreeSWITCH Sites____ > > http://www.freeswitch.org____ > > http://wiki.freeswitch.org____ > > http://www.cluecon.com____ > > __ __ > > FreeSWITCH-users mailing list____ > > FreeSWITCH-users at lists.freeswitch.org ____ > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users____ > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users____ > > http://www.freeswitch.org____ > > __ __ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From karl at xtronics.com Thu Oct 24 21:13:34 2013 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 24 Oct 2013 12:13:34 -0500 Subject: [Freeswitch-users] gateways testing DNS SRV records Message-ID: <5269553E.2030806@xtronics.com> I'm trying to figure out a way to see if a freeswitch gateway respects SRV records. I've been trying different sofia commands without finding the IP that it is using. sofia_dig returns just the A records. To be clear - let's say you are using example.com as your DID and have a gateway set up. The provider should have SRV records so that there can be graceful failures. To see the IPs to use - do dig SRV _sip._upd.example.com So my precise question is how to see the IP freeswitch is using for a gateway. (I'm assuming that freeswitch knows how to use SRV records?) -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 It?s no wonder that truth is stranger than fiction. Fiction has to make sense. Mark Twain -------------------------------------------------------------------------------- From yehavi.bourvine at gmail.com Thu Oct 24 21:23:25 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 24 Oct 2013 19:23:25 +0200 Subject: [Freeswitch-users] Performance issue In-Reply-To: References: Message-ID: Yes, this is the problem I refered to. I'll try testing it on next Thursday (have some constraints) and report back. Thanks! __Yehavi: 2013/10/24 Anthony Minessale > What is the "same problem" exactly. > > If you are talking about the stuck channels issues stemming from calls > being put on hold. It was fixed in stable HEAD. > > git commit: 4edd7b74223a7ddedd7dd65ed033b4d1da7fb6ba > http://jira.freeswitch.org/browse/FS-5835 > > > This is why we are not supposed to be discussing bugs in the mailing list. > Bugs have a home called jira where they are tracked and tied to commits > with git. > > > > > > > On Wed, Oct 23, 2013 at 11:12 PM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Sorry, but I am still puzzled... >> It has been fixed before 1.2.14, but the problem is only at the 1.2 >> branch... So does 1.2.14 has the fix or not? >> >> BTW, I had the same problem with the HEAD when I tried it a few months >> ago... >> >> Thanks! __Yehavi: >> >> >> 2013/10/23 Ken Rice >> >>> The problem was only in the stable branch, (that would be the 1.2 >>> line) and was fixed prior to the 1.2.14 release... >>> >>> >>> On 10/23/13 9:03 AM, "Yehavi Bourvine" >>> wrote: >>> >>> HI, >>> >>> A question: Is it solved on 1.2.14 or only on HEAD? I tried today 1.2.14 >>> and failed with the same sympthoms... >>> >>> Thanks, __Yehavi: >>> >>> >>> 2013/10/14 Anthony Minessale >>> >>> It is for many. It depends on the general goals etc. >>> The problem you describe was only on stable and all the symptoms, not >>> shutting down, channels count being wrong, zombies etc are all the same 1 >>> problem now fixed. >>> >>> >>> >>> On Mon, Oct 14, 2013 at 11:33 AM, Stephen Wilde >>> wrote: >>> >>> I have tried only branch v1.2.stable not master. Is the master good for >>> a production environment? >>> >>> >>> On Mon, Oct 14, 2013 at 6:27 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>> That's all the same thing...... >>> >>> Do you have this problem on master branch as well? >>> >>> >>> >>> On Mon, Oct 14, 2013 at 11:21 AM, Stephen Wilde >>> wrote: >>> >>> Thank you for your advice. >>> I have tried the 1.2.stable but I have due to abandon this branch due to >>> a some "zombie sessions" in FreeSwitch visible also during shutdown with a "Waiting >>> x sessions ..." messages. >>> I have seen that this issue (FS-5848) has already been solved so I can >>> try with 1.2.stable but I don't see any fix that can affect my issue. >>> >>> >>> On Mon, Oct 14, 2013 at 6:03 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>> Try latest HEAD for a probable fix. >>> >>> >>> >>> On Mon, Oct 14, 2013 at 10:53 AM, Stephen Wilde >>> wrote: >>> >>> Hi all, >>> we have an issue with our FreeSwitch box that to us seems to be >>> performance related. >>> The effect is that in FreeSwitch the number of session reaches the limit >>> we set in config as: >>> >>> >>> >>> The limit is reached independently of its value because when FreeSwitch >>> is in this state, the number of sessions grows indefinitely. >>> >>> We have tried to upgrade the hardware of the box moving from a Xeon 2 >>> CPU E5649 (12 core 2.53GHz) to a Xeon 4 CPU E5-4640 (32 core 2.40GHz) but >>> with this more powerful hardware it happens that the limit is reached with >>> less sessions. >>> It seems that performance are related to the speed of single core >>> instead of speed of the box. >>> Make sense? >>> >>> A example of "status" issued before the crash is: >>> >>> 14969852 session(s) since startup >>> 13765 session(s) - 538 out of max 1000 per sec >>> 30000 session(s) max >>> min idle cpu 0.00/36.00 >>> Current Stack Size/Max 240K/8192K >>> >>> >>> Any advice? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> G+ ClueCon : http://www.ss7.us/cluecon-gplus >>> FB ClueCon : http://www.ss7.us/cluecon-fb >>> G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus >>> FB FreeSWITCH : http://www.ss7.us/freeswitch-fb >>> Twitter : @FreeSWITCH_WIRE >>> irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/04f9201a/attachment-0001.html From vbvbrj at gmail.com Thu Oct 24 21:43:02 2013 From: vbvbrj at gmail.com (Mimiko) Date: Thu, 24 Oct 2013 20:43:02 +0300 Subject: [Freeswitch-users] api:execute in lua Message-ID: <52695C26.7030005@gmail.com> Hello. I recently encountered a strange lua script execution problem. For example, a line in script is like this: 1) api:execute("some command here") and with return: 2) result = api:execute("some command here") Now I have a question to experts. Is it true that lua module of FS will not wait execution end of the command in first case and continua to next line in lua script, but will wait execution end of command for second example and then continua to next line? If it is so, then I will update wiki for api:execute. If it is not true, then its a bug and will fire a jira. Thanks. -- Mimiko desu. From intralanman at freeswitch.org Thu Oct 24 22:49:35 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 24 Oct 2013 14:49:35 -0400 Subject: [Freeswitch-users] gateways testing DNS SRV records In-Reply-To: <5269553E.2030806@xtronics.com> References: <5269553E.2030806@xtronics.com> Message-ID: <52696BBF.5090800@freeswitch.org> Yes, FreeSWITCH supports SRV records, and handles them correctly. When I use sofia_dig, I see the IP and the port that are returned from DNS lookups via dig. If you're not, then you or your provider must have something misconfigured. freeswitch at internal> sofia_dig sip.flowroute.com Preference Weight Transport Port Address ================================================================================ 1 1.000 udp 5060 216.115.69.144 2 1.000 udp 5060 70.167.153.130 freeswitch at internal> sofia_dig callcentric.com Preference Weight Transport Port Address ================================================================================ 1 0.000 udp 5080 204.11.192.161 1 0.000 udp 5080 204.11.192.163 1 0.000 udp 5080 204.11.192.164 1 0.000 udp 5080 204.11.192.169 1 0.000 udp 5080 204.11.192.170 1 0.000 udp 5080 204.11.192.171 1 0.000 udp 5080 204.11.192.159 1 0.000 udp 5080 204.11.192.160 -Ray On 10/24/2013 01:13 PM, Karl Schmidt wrote: > I'm trying to figure out a way to see if a freeswitch gateway respects SRV records. > > I've been trying different sofia commands without finding the IP that it is using. > > > sofia_dig returns just the A records. > > > To be clear - let's say you are using example.com as your DID and have a gateway set up. > > The provider should have SRV records so that there can be graceful failures. > > To see the IPs to use - do > > dig SRV _sip._upd.example.com > > > So my precise question is how to see the IP freeswitch is using for a gateway. (I'm assuming that > freeswitch knows how to use SRV records?) > > > > > > > -------------------------------------------------------------------------------- > Karl Schmidt EMail Karl at xtronics.com > Transtronics, Inc. WEB http://secure.transtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 FAX (785) 841-0434 > > It?s no wonder that truth is stranger than fiction. Fiction has to make sense. > Mark Twain > -------------------------------------------------------------------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From karl at xtronics.com Thu Oct 24 23:46:05 2013 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 24 Oct 2013 14:46:05 -0500 Subject: [Freeswitch-users] gateways testing DNS SRV records In-Reply-To: <52696BBF.5090800@freeswitch.org> References: <5269553E.2030806@xtronics.com> <52696BBF.5090800@freeswitch.org> Message-ID: <526978FD.6060705@xtronics.com> On 10/24/2013 01:49 PM, Raymond Chandler wrote: > Yes, FreeSWITCH supports SRV records, and handles them correctly. When > I use sofia_dig, I see the IP and the port that are returned from DNS > lookups via dig. If you're not, then you or your provider must have > something misconfigured. This is correct - I was able to see the details by setting sofia loglevel all 9 ( I must say that the SRV record syntax is poorly designed - and shouldn't have required multiple lookups - needlessly complex. ) > > On 10/24/2013 01:13 PM, Karl Schmidt wrote: >> I'm trying to figure out a way to see if a freeswitch gateway respects SRV records. >> >> I've been trying different sofia commands without finding the IP that it is using. >> >> >> sofia_dig returns just the A records. >> >> >> To be clear - let's say you are using example.com as your DID and have a gateway set up. >> >> The provider should have SRV records so that there can be graceful failures. >> >> To see the IPs to use - do >> >> dig SRV _sip._upd.example.com >> >> >> So my precise question is how to see the IP freeswitch is using for a gateway. (I'm assuming that >> freeswitch knows how to use SRV records?) >> -------------------------------------------------------------------------------- Karl Schmidt EMail Karl at xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Taking subsidy money from the government is immoral. Realize that you are taking the money of people that work at McDonald?s by threat of force. kps -------------------------------------------------------------------------------- From alipey at gmail.com Fri Oct 25 00:11:09 2013 From: alipey at gmail.com (Ali Pey) Date: Thu, 24 Oct 2013 16:11:09 -0400 Subject: [Freeswitch-users] silence detection not working for record Message-ID: Hello, I have the following line in my perl ESL code to record a message. No matter how long I wait in mute after recording, it does not stop recording. con->execute('record', '${file} 300 200 4' ); Please help. Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/0fde5faf/attachment.html From dujinfang at gmail.com Fri Oct 25 04:19:03 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 25 Oct 2013 08:19:03 +0800 Subject: [Freeswitch-users] xmllint and default config In-Reply-To: <52687E5D.4050900@xtronics.com> References: <52687E5D.4050900@xtronics.com> Message-ID: +1 to fix that ? 2013?10?24? ??10:00?"Karl Schmidt" ??? > Complains about a bit found in /etc/autoload_configs/abstraction.conf > > > $ xmllint /var/log/freeswitch/freeswitch.xml.fsxml > /var/log/freeswitch/freeswitch.xml.fsxml:269: parser error : Unescaped '<' > not allowed in attributes > values > syntax="" > ..... > > > > > > -------------------------------------------------------------------------------- > Karl Schmidt EMail Karl at xtronics.com > Transtronics, Inc. WEB > http://secure.transtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 FAX (785) 841-0434 > > When the government tries to do something; > half the time it accomplishes the opposite, > the other 49% of the time it produces a negative consequence > that is worse than the problem they set out to solve, and > Finally, 10% of the time someone absconds with the money. -kps > > > > -------------------------------------------------------------------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/9773097e/attachment.html From max at nysolutions.com Fri Oct 25 04:24:27 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 25 Oct 2013 00:24:27 +0000 Subject: [Freeswitch-users] xmllint and default config In-Reply-To: References: <52687E5D.4050900@xtronics.com> Message-ID: I doubt they will fix without a Jira. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du Sent: Thursday, October 24, 2013 8:19 PM To: freeswitch-users Subject: Re: [Freeswitch-users] xmllint and default config +1 to fix that ? 2013?10?24? ??10:00?"Karl Schmidt" >??? Complains about a bit found in /etc/autoload_configs/abstraction.conf $ xmllint /var/log/freeswitch/freeswitch.xml.fsxml /var/log/freeswitch/freeswitch.xml.fsxml:269: parser error : Unescaped '<' not allowed in attributes values Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 When the government tries to do something; half the time it accomplishes the opposite, the other 49% of the time it produces a negative consequence that is worse than the problem they set out to solve, and Finally, 10% of the time someone absconds with the money. -kps -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/81dcf541/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/81dcf541/attachment-0001.jpg From krice at freeswitch.org Fri Oct 25 04:49:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 24 Oct 2013 19:49:41 -0500 Subject: [Freeswitch-users] xmllint and default config In-Reply-To: References: <52687E5D.4050900@xtronics.com> Message-ID: indoubt it will be fixed at all... freeswitch xml actually can do things that will make xmlint seriously complain Ken Sent from my iPad On Oct 24, 2013, at 19:24, Moishe Grunstein wrote: > I doubt they will fix without a Jira. > > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du > Sent: Thursday, October 24, 2013 8:19 PM > To: freeswitch-users > Subject: Re: [Freeswitch-users] xmllint and default config > > +1 to fix that > > ? 2013?10?24? ??10:00?"Karl Schmidt" ??? > Complains about a bit found in /etc/autoload_configs/abstraction.conf > > > $ xmllint /var/log/freeswitch/freeswitch.xml.fsxml > /var/log/freeswitch/freeswitch.xml.fsxml:269: parser error : Unescaped '<' not allowed in attributes > values > ..... > > > > > -------------------------------------------------------------------------------- > Karl Schmidt EMail Karl at xtronics.com > Transtronics, Inc. WEB http://secure.transtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 FAX (785) 841-0434 > > When the government tries to do something; > half the time it accomplishes the opposite, > the other 49% of the time it produces a negative consequence > that is worse than the problem they set out to solve, and > Finally, 10% of the time someone absconds with the money. -kps > > > -------------------------------------------------------------------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131024/fa4f49a9/attachment.html From gmangudai at gmail.com Fri Oct 25 05:55:18 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Fri, 25 Oct 2013 09:55:18 +0800 Subject: [Freeswitch-users] has anybody ever succeeded in building fs 1.2.14 with VS2010 or the similiar? Message-ID: i tried but cannot build mod_sofia while other mods as well as the fs core are ok, libs\sofia-sip\win32\config.h got emptified every time the project libsofia_sip_ua_static is built as this doesn't happen with FS 1.2.10, without config.h the pre-definitions are lost which leads to a lot of errors. so could somebody shed a light? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/27786c83/attachment.html From max at nysolutions.com Fri Oct 25 06:48:41 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Fri, 25 Oct 2013 02:48:41 +0000 Subject: [Freeswitch-users] has anybody ever succeeded in building fs 1.2.14 with VS2010 or the similiar? In-Reply-To: References: Message-ID: AlexForster, posted the following on IRC the past weekend. 1) Extract freeswitch-1.2.14.tar.bz2 to a 10G ramdisk 2) Build the make_cielab_luts project 3) Build the rest of the solution 4) Build iksemel project 5) Build the rest of the solution again ** DO NOT EVER "REBUILD" DURING THIS PROCESS ** Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vincent Xia Sent: Thursday, October 24, 2013 9:55 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] has anybody ever succeeded in building fs 1.2.14 with VS2010 or the similiar? i tried but cannot build mod_sofia while other mods as well as the fs core are ok, libs\sofia-sip\win32\config.h got emptified every time the project libsofia_sip_ua_static is built as this doesn't happen with FS 1.2.10, without config.h the pre-definitions are lost which leads to a lot of errors. so could somebody shed a light? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/6c611089/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/6c611089/attachment-0001.jpg From karl at xtronics.com Fri Oct 25 07:26:19 2013 From: karl at xtronics.com (Karl Schmidt) Date: Thu, 24 Oct 2013 22:26:19 -0500 Subject: [Freeswitch-users] xmllint and default config In-Reply-To: References: <52687E5D.4050900@xtronics.com> Message-ID: <5269E4DB.7030908@xtronics.com> On 10/24/2013 07:49 PM, Ken Rice wrote: > indoubt it will be fixed at all... freeswitch xml actually can do things that will make xml[l]int > seriously complain > > Ken > Sent from my iPad Actually, only two places have an issue I'm not sure what abstraction.xml is about - and I'm not sure if replacing the '<' with > will not break anything? original: Hello everybody, I need to bridge an inbound call from a gateway number to another gateway number (same provider). Call is made, but there is no sound exchange, nor any ringback while calling "GATEWAY_1_NUMBER". Here is the public dialplan (in /usr/local/freeswitch/conf/dialplan/public/.): I have searched in this mailing list's archives but have not found any answer. Is there anyone who knows what appends and what can I do to make it work. Thanks in advance, TB -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sound-issue-on-bridging-one-inbound-gateway-call-to-another-gateway-tp7595967.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vbvbrj at gmail.com Fri Oct 25 13:23:53 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 25 Oct 2013 12:23:53 +0300 Subject: [Freeswitch-users] Sound issue on bridging one inbound gateway call to another gateway In-Reply-To: <1382692066387-7595967.post@n2.nabble.com> References: <1382692066387-7595967.post@n2.nabble.com> Message-ID: <526A38A9.7090609@gmail.com> On 25.10.2013 12:07, obbyone wrote: > I need to bridge an inbound call from a gateway number to another gateway > number (same provider). Call is made, but there is no sound exchange, nor > any ringback while calling "GATEWAY_1_NUMBER". Hi. Read this page https://wiki.freeswitch.org/wiki/Bypass_Media and related pages to understand how RTP works and why you have this problem. -- Mimiko desu. From juanito1982 at gmail.com Fri Oct 25 16:42:33 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 25 Oct 2013 14:42:33 +0200 Subject: [Freeswitch-users] Bad request answer In-Reply-To: References: Message-ID: You are the man Steven! How painful SIP ALG is... Do you know if possible to be more permissive with SIP ALG changes like that? Regards 2013/10/20 Steven Ayre > Looks like the old router changed the IP in the SDP without updating the > Content-Length*, so the SDP length was different from what the packet > claimed hence the error. > > *The public IP is 3 characters shorter than the internal one. The SDP is > otherwise identical but both have 258 as content-length. The router was > removing 6 bytes from the SDP so should have changed this to 252. > > > On 20 October 2013 16:30, Juan Antonio Iba?ez Santorum < > juanito1982 at gmail.com> wrote: > >> New routers solved the problem. The old one seems to modify SDP. New >> INVITE requests seems as: >> >> http://pastebin.com/hQFjDC2c >> >> We can see local IPv4 ips instead the public ones. Whan I don't know is >> why FS didn't like previous requests. Is there any way to know why? >> >> Regards >> >> >> 2013/10/18 Anthony Minessale >> >>> Maybe try factory resetting the device? >>> >>> >>> >>> >>> On Fri, Oct 18, 2013 at 6:24 AM, Juan Antonio Iba?ez Santorum < >>> juanito1982 at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> I am getting "Bad request" answer from FS but I don't know why. I >>>> have phone as that (AT810) in other locations without problems. You can see >>>> one INVITE and answer here >>>> >>>> http://pastebin.com/Widd5j2i >>>> >>>> Do you know why FS doesn't like it? >>>> Regards >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/81df51b7/attachment.html From steveayre at gmail.com Fri Oct 25 17:13:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 25 Oct 2013 14:13:23 +0100 Subject: [Freeswitch-users] Bad request answer In-Reply-To: References: Message-ID: Unsure. It's saying there's more data than there is so FS is expecting more. It's an invalid packet. You just need to complain to the manufacturer and hope they release a firmware upgrade. Out of curiosity, what router and firmware is it? Might be worth documenting it on http://wiki.freeswitch.org/wiki/ALG On 25 October 2013 13:42, Juan Antonio Iba?ez Santorum < juanito1982 at gmail.com> wrote: > You are the man Steven! > > How painful SIP ALG is... > > Do you know if possible to be more permissive with SIP ALG changes like > that? > > Regards > > > 2013/10/20 Steven Ayre > >> Looks like the old router changed the IP in the SDP without updating the >> Content-Length*, so the SDP length was different from what the packet >> claimed hence the error. >> >> *The public IP is 3 characters shorter than the internal one. The SDP is >> otherwise identical but both have 258 as content-length. The router was >> removing 6 bytes from the SDP so should have changed this to 252. >> >> >> On 20 October 2013 16:30, Juan Antonio Iba?ez Santorum < >> juanito1982 at gmail.com> wrote: >> >>> New routers solved the problem. The old one seems to modify SDP. New >>> INVITE requests seems as: >>> >>> http://pastebin.com/hQFjDC2c >>> >>> We can see local IPv4 ips instead the public ones. Whan I don't know is >>> why FS didn't like previous requests. Is there any way to know why? >>> >>> Regards >>> >>> >>> 2013/10/18 Anthony Minessale >>> >>>> Maybe try factory resetting the device? >>>> >>>> >>>> >>>> >>>> On Fri, Oct 18, 2013 at 6:24 AM, Juan Antonio Iba?ez Santorum < >>>> juanito1982 at gmail.com> wrote: >>>> >>>>> Hello, >>>>> >>>>> I am getting "Bad request" answer from FS but I don't know why. I >>>>> have phone as that (AT810) in other locations without problems. You can see >>>>> one INVITE and answer here >>>>> >>>>> http://pastebin.com/Widd5j2i >>>>> >>>>> Do you know why FS doesn't like it? >>>>> Regards >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/7ce26484/attachment-0001.html From dujinfang at gmail.com Fri Oct 25 17:24:53 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 25 Oct 2013 21:24:53 +0800 Subject: [Freeswitch-users] freeswitch.Session origination disposition in Lua Message-ID: <64319776C7BC484FA6A22F78EFB29F3A@gmail.com> Hi, I'm trying to get the disposition of a failed call in lua, but log shows session is not initialised on the "else" case, is there a way to get the reason why it fails? Thanks. Lua: local bleg = freeswitch.Session("user/" .. digits) if bleg:ready() then bleg:transfer("conference:3000", "inline", "") else disposition = freeswitch.getVariable("origination_disposition") session:speak("Call failed " .. disposition) end -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/d2d69073/attachment.html From mike at jerris.com Fri Oct 25 17:43:50 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Oct 2013 09:43:50 -0400 Subject: [Freeswitch-users] has anybody ever succeeded in building fs 1.2.14 with VS2010 or the similiar? In-Reply-To: References: Message-ID: <03BEBC69-7035-44D9-87A2-637D0533F57D@jerris.com> That is really strange, sounds like some missing project dependencies. Can you confirm those couple that you have to build manually are depends of the things that need them? On Oct 24, 2013, at 10:48 PM, Moishe Grunstein wrote: > AlexForster, posted the following on IRC the past weekend. > 1) Extract freeswitch-1.2.14.tar.bz2 to a 10G ramdisk > 2) Build the make_cielab_luts project > 3) Build the rest of the solution > 4) Build iksemel project > 5) Build the rest of the solution again > ** DO NOT EVER "REBUILD" DURING THIS PROCESS ** > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vincent Xia > Sent: Thursday, October 24, 2013 9:55 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] has anybody ever succeeded in building fs 1.2.14 with VS2010 or the similiar? > > i tried but cannot build mod_sofia while other mods as well as the fs core are ok, > libs\sofia-sip\win32\config.h got emptified every time the project libsofia_sip_ua_static is built as this doesn't happen with FS 1.2.10, > without config.h the pre-definitions are lost which leads to a lot of errors. so could somebody shed a light? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/24071351/attachment.html From abdullah at smonte.com Fri Oct 25 17:13:13 2013 From: abdullah at smonte.com (Abdullah) Date: Fri, 25 Oct 2013 06:13:13 -0700 (PDT) Subject: [Freeswitch-users] freeswitch XML CDR LEG A DISABLE Message-ID: <1382706793603-7595969.post@n2.nabble.com> HI , PLEASE HELP ME HOW TO DISABLE LEG-A CDR IN FREESWITCH , I JUST WANT B-LEG CDR IN MY FREESWITCH. THANKS IN ADVANCE . WAITING FOR YOUR RESPONSE.... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-XML-CDR-LEG-A-DISABLE-tp7595969.html Sent from the freeswitch-users mailing list archive at Nabble.com. From abdullah at smonte.com Fri Oct 25 17:43:35 2013 From: abdullah at smonte.com (Abdullah) Date: Fri, 25 Oct 2013 06:43:35 -0700 (PDT) Subject: [Freeswitch-users] sched_hangup not working in perl Message-ID: <1382708615270-7595970.post@n2.nabble.com> hi every one, my call session has been answer already , because i need to play ivr , so sched_hangup variable not working in perl script. please help me .. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sched-hangup-not-working-in-perl-tp7595970.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nasida at live.ru Fri Oct 25 19:40:19 2013 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 25 Oct 2013 19:40:19 +0400 Subject: [Freeswitch-users] FS + ZRTP + SAS Message-ID: Hi guys! I have problem with transmission of SAS (Short Authentication String).FS version 1.2.10 I enabledbut SAS strings is still different. I also tried but this didn't help. I heard that it is fixed with 1.3.X branch. Can somebody please confirm or deny this? Please advice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/0ff971f0/attachment.html From smontour at verizon.net Fri Oct 25 21:20:04 2013 From: smontour at verizon.net (Sam Montour) Date: Fri, 25 Oct 2013 12:20:04 -0500 Subject: [Freeswitch-users] sched_hangup not working in perl In-Reply-To: <1382708615270-7595970.post@n2.nabble.com> References: <1382708615270-7595970.post@n2.nabble.com> Message-ID: <004201ced1a6$721e0770$565a1650$@verizon.net> Can you post the Perl statement line you are executing for sched_hangup? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abdullah Sent: Friday, October 25, 2013 8:44 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] sched_hangup not working in perl hi every one, my call session has been answer already , because i need to play ivr , so sched_hangup variable not working in perl script. please help me .. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sched-hangup-not-working-in-pe rl-tp7595970.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Fri Oct 25 22:25:51 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 25 Oct 2013 13:25:51 -0500 Subject: [Freeswitch-users] Friday Free For All Activate! Get in here Message-ID: sip:888 at conference.freeswitch.org or http://fs0.us/call888 for other access info. Todays topic?? What ever we want it to be! Come on and find out what its all about! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/15331067/attachment-0001.html From happy at hour.com Fri Oct 25 23:11:42 2013 From: happy at hour.com (Peter) Date: Fri, 25 Oct 2013 21:11:42 +0200 Subject: [Freeswitch-users] freeswitch-mod-html5 broken package debian wheezy repo Message-ID: <526AC26E.1050606@hour.com> # apt-cache showpkg freeswitch-mod-html5 Package: freeswitch-mod-html5 Versions: 1.2.13~1-1~wheezy+1 (/var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages MD5: 216e5c26c5a4b68dca0f1da1b5770ff3 Reverse Depends: freeswitch-mod-html5-dbg,freeswitch-mod-html5 1.2.13~1-1~wheezy+1 freeswitch-meta-sorbet,freeswitch-mod-html5 1.2.14~1-1~wheezy+1 freeswitch-meta-all,freeswitch-mod-html5 1.2.14~1-1~wheezy+1 Dependencies: 1.2.13~1-1~wheezy+1 - libc6 (2 2.3.6-6~) libfreeswitch1 (5 1.2.13~1-1~wheezy+1) libjpeg62 (2 6b1) libncurses5 (2 5.5-5~) libodbc1 (18 2.2.11) unixodbc (2 2.2.11) libpq5 (0 (null)) libssl1.0.0 (2 1.0.0) libtinfo5 (0 (null)) freeswitch-mod-html5-dbg (0 (null)) Provides: 1.2.13~1-1~wheezy+1 - Reverse Provides: # apt-cache showpkg libfreeswitch1 Package: libfreeswitch1 Versions: 1.2.14~1-1~wheezy+1 (/var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages MD5: d04d816084986903a32136379fc1afa7 ... Dependencies: 1.2.14~1-1~wheezy+1 - libc6 (2 2.8) libgcc1 (2 1:4.1.1) libjpeg62 (2 6b1) libncurses5 (2 5.5-5~) libodbc1 (18 2.2.11) unixodbc (2 2.2.11) libpq5 (0 (null)) libssl1.0.0 (2 1.0.0) libstdc++6 (2 4.1.1) libtinfo5 (0 (null)) libuuid1 (2 2.16) zlib1g (2 1:1.1.4) libfreeswitch1-dbg (0 (null)) Provides: 1.2.14~1-1~wheezy+1 - Reverse Provides: # apt-get install freeswitch-meta-vanilla freeswitch-mod-html5 .... The following packages have unmet dependencies: freeswitch-mod-html5 : Depends: libfreeswitch1 (= 1.2.13~1-1~wheezy+1) but 1.2.14~1-1~wheezy+1 is to be installed E: Unable to correct problems, you have held broken packages. I noticed this problem since the 16th of October 2013 From jkr888 at gmail.com Fri Oct 25 23:41:22 2013 From: jkr888 at gmail.com (Johny Kadarisman Kwan) Date: Fri, 25 Oct 2013 15:41:22 -0400 Subject: [Freeswitch-users] How to failover with fail_on_single_reject construct In-Reply-To: References: Message-ID: Like to report back, in case anybody bumping to the same problem. After digging more and looking at debug log. Apparently the culprit is on the way I set channel variable. On my case, dialstring should be : Once that in-place, it work as expected. Thats all, my bad, Freeswitch Rock!! On Fri, Oct 18, 2013 at 3:58 PM, Johny Kadarisman Kwan wrote: > I?m attempting to setup FS to failover from gw1, gw2, and gw3 > subsequently. AND if any gateway response with NO_ANSWER or > ALLOTED_TIMEOUT, then the next attempt should be stop. Using > ?fail_on_single_reject? var, I tried following script : > > > > > > > > > > > > data="{ignore_early_media=true,fail_on_single_reject=NO_ANSWER, > ALLOTTED_TIMEOUT}[leg_timeout=20}]sofia/gateway/gw1/7321231234|[leg_timeout=20]sofia/gateway/gw2/7321231234|[leg_timeout=20]sofia/gateway/gw3/7321231234}"/> > > > > > > > > > > Bridge app still continue attempting call to each gw1, gw2, and gw3 when > ALLOTED_TIMEOUT response. From searching lists, above dialplan should work > out fine. Did I misses any steps? > > > > Btw, tried above on FreeSWITCH Version > 1.5.6b+git~20131010T172322Z~1cd6d44b06 (git 1cd6d44 2013-10-10 17:23:22Z) > windows built. > > > > Thanks, > > Jkwan > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131025/67fc3ddb/attachment.html From gmangudai at gmail.com Sat Oct 26 05:19:22 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Sat, 26 Oct 2013 09:19:22 +0800 Subject: [Freeswitch-users] has anybody ever succeeded in building fs 1.2.14 with VS2010 or the similiar? In-Reply-To: <03BEBC69-7035-44D9-87A2-637D0533F57D@jerris.com> References: <03BEBC69-7035-44D9-87A2-637D0533F57D@jerris.com> Message-ID: well it's almost ok(178 projects succeeded out of 183) now and the problem is that the gawk.exe get corrupt, i didn't follow the building order mentioned above. 2013/10/25 Michael Jerris > That is really strange, sounds like some missing project dependencies. > Can you confirm those couple that you have to build manually are depends > of the things that need them? > > On Oct 24, 2013, at 10:48 PM, Moishe Grunstein > wrote: > > AlexForster, posted the following on IRC the past weekend.**** > 1) Extract freeswitch-1.2.14.tar.bz2 to a 10G ramdisk**** > 2) Build the make_cielab_luts project**** > 3) Build the rest of the solution**** > 4) Build iksemel project**** > 5) Build the rest of the solution again**** > ** DO NOT EVER "REBUILD" DURING THIS PROCESS ****** > > Thanks,**** > > Moishe Grunstein**** > Tornado Computer Systems, Inc.**** > 212.400.7650 888.IPPBX.US > *Service Request Email: support at nysolutions.com***** > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > **** > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Vincent Xia > *Sent:* Thursday, October 24, 2013 9:55 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] has anybody ever succeeded in building fs > 1.2.14 with VS2010 or the similiar?**** > ** ** > i tried but cannot build mod_sofia while other mods as well as the fs core > are ok, **** > libs\sofia-sip\win32\config.h got emptified every time the project > libsofia_sip_ua_static is built as this doesn't happen with FS 1.2.10,**** > without config.h the pre-definitions are lost which leads to a lot of > errors. so could somebody shed a light?**** > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131026/09fbf27f/attachment.html From steveayre at gmail.com Sat Oct 26 15:39:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 26 Oct 2013 12:39:03 +0100 Subject: [Freeswitch-users] freeswitch-mod-html5 broken package debian wheezy repo In-Reply-To: <526AC26E.1050606@hour.com> References: <526AC26E.1050606@hour.com> Message-ID: Looks like that module wasn't built for 1.2.14 in the files.freeswitch.orgrepo... Ken? (or whoever maintains that repo) On 25 October 2013 20:11, Peter wrote: > # apt-cache showpkg freeswitch-mod-html5 > Package: freeswitch-mod-html5 > Versions: > 1.2.13~1-1~wheezy+1 > > (/var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages) > Description Language: > File: > > /var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages > MD5: 216e5c26c5a4b68dca0f1da1b5770ff3 > > Reverse Depends: > freeswitch-mod-html5-dbg,freeswitch-mod-html5 1.2.13~1-1~wheezy+1 > freeswitch-meta-sorbet,freeswitch-mod-html5 1.2.14~1-1~wheezy+1 > freeswitch-meta-all,freeswitch-mod-html5 1.2.14~1-1~wheezy+1 > Dependencies: > 1.2.13~1-1~wheezy+1 - libc6 (2 2.3.6-6~) libfreeswitch1 (5 > 1.2.13~1-1~wheezy+1) libjpeg62 (2 6b1) libncurses5 (2 5.5-5~) libodbc1 > (18 2.2.11) unixodbc (2 2.2.11) libpq5 (0 (null)) libssl1.0.0 (2 1.0.0) > libtinfo5 (0 (null)) freeswitch-mod-html5-dbg (0 (null)) > Provides: > 1.2.13~1-1~wheezy+1 - > Reverse Provides: > > # apt-cache showpkg libfreeswitch1 > Package: libfreeswitch1 > Versions: > 1.2.14~1-1~wheezy+1 > > (/var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages) > Description Language: > File: > > /var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_binary-i386_Packages > MD5: d04d816084986903a32136379fc1afa7 > ... > Dependencies: > 1.2.14~1-1~wheezy+1 - libc6 (2 2.8) libgcc1 (2 1:4.1.1) libjpeg62 (2 > 6b1) libncurses5 (2 5.5-5~) libodbc1 (18 2.2.11) unixodbc (2 2.2.11) > libpq5 (0 (null)) libssl1.0.0 (2 1.0.0) libstdc++6 (2 4.1.1) libtinfo5 > (0 (null)) libuuid1 (2 2.16) zlib1g (2 1:1.1.4) libfreeswitch1-dbg (0 > (null)) > Provides: > 1.2.14~1-1~wheezy+1 - > Reverse Provides: > > # apt-get install freeswitch-meta-vanilla freeswitch-mod-html5 > .... > The following packages have unmet dependencies: > freeswitch-mod-html5 : Depends: libfreeswitch1 (= 1.2.13~1-1~wheezy+1) > but 1.2.14~1-1~wheezy+1 is to be installed > E: Unable to correct problems, you have held broken packages. > > I noticed this problem since the 16th of October 2013 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131026/0829a78b/attachment-0001.html From krice at freeswitch.org Sat Oct 26 17:48:30 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 26 Oct 2013 08:48:30 -0500 Subject: [Freeswitch-users] freeswitch-mod-html5 broken package debian wheezy repo In-Reply-To: Message-ID: There is no mod_html5... Has anyone opened a jira on this? On 10/26/13 6:39 AM, "Steven Ayre" wrote: > Looks like that module wasn't built for 1.2.14 in the files.freeswitch.org > repo... Ken? (or whoever maintains that repo) > > > On 25 October 2013 20:11, Peter wrote: >> # apt-cache showpkg ?freeswitch-mod-html5 >> Package: freeswitch-mod-html5 >> Versions: >> 1.2.13~1-1~wheezy+1 >> (/var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_bi >> nary-i386_Packages) >> ?Description Language: >> ? ? ? ? ? ? ? ? ?File: >> /var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_bin >> ary-i386_Packages >> ? ? ? ? ? ? ? ? ? MD5: 216e5c26c5a4b68dca0f1da1b5770ff3 >> >> Reverse Depends: >> ? freeswitch-mod-html5-dbg,freeswitch-mod-html5 1.2.13~1-1~wheezy+1 >> ? freeswitch-meta-sorbet,freeswitch-mod-html5 1.2.14~1-1~wheezy+1 >> ? freeswitch-meta-all,freeswitch-mod-html5 1.2.14~1-1~wheezy+1 >> Dependencies: >> 1.2.13~1-1~wheezy+1 - libc6 (2 2.3.6-6~) libfreeswitch1 (5 >> 1.2.13~1-1~wheezy+1) libjpeg62 (2 6b1) libncurses5 (2 5.5-5~) libodbc1 >> (18 2.2.11) unixodbc (2 2.2.11) libpq5 (0 (null)) libssl1.0.0 (2 1.0.0) >> libtinfo5 (0 (null)) freeswitch-mod-html5-dbg (0 (null)) >> Provides: >> 1.2.13~1-1~wheezy+1 - >> Reverse Provides: >> >> # apt-cache showpkg ?libfreeswitch1 >> Package: libfreeswitch1 >> Versions: >> 1.2.14~1-1~wheezy+1 >> (/var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_bi >> nary-i386_Packages) >> ?Description Language: >> ? ? ? ? ? ? ? ? ?File: >> /var/lib/apt/lists/files.freeswitch.org_repo_deb_debian_dists_wheezy_main_bin >> ary-i386_Packages >> ? ? ? ? ? ? ? ? ? MD5: d04d816084986903a32136379fc1afa7 >> ... >> Dependencies: >> 1.2.14~1-1~wheezy+1 - libc6 (2 2.8) libgcc1 (2 1:4.1.1) libjpeg62 (2 >> 6b1) libncurses5 (2 5.5-5~) libodbc1 (18 2.2.11) unixodbc (2 2.2.11) >> libpq5 (0 (null)) libssl1.0.0 (2 1.0.0) libstdc++6 (2 4.1.1) libtinfo5 >> (0 (null)) libuuid1 (2 2.16) zlib1g (2 1:1.1.4) libfreeswitch1-dbg (0 >> (null)) >> Provides: >> 1.2.14~1-1~wheezy+1 - >> Reverse Provides: >> >> # apt-get install freeswitch-meta-vanilla freeswitch-mod-html5 >> .... >> The following packages have unmet dependencies: >> ?freeswitch-mod-html5 : Depends: libfreeswitch1 (= 1.2.13~1-1~wheezy+1) >> but 1.2.14~1-1~wheezy+1 is to be installed >> E: Unable to correct problems, you have held broken packages. >> >> I noticed this problem since the 16th of October 2013 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131026/c5a67165/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sat Oct 26 17:54:30 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 26 Oct 2013 14:54:30 +0100 Subject: [Freeswitch-users] freeswitch XML CDR LEG A DISABLE In-Reply-To: <1382706793603-7595969.post@n2.nabble.com> References: <1382706793603-7595969.post@n2.nabble.com> Message-ID: Hello, Please do not ask questions in full caps lock, you're less likely to get a reply and it will annoy a lot of people :) You also did not explain what you had done to find the answer yourself, or what you had tried so far, which again will result in people being less likely to respond. In order to get a good answer from the community, you need to show you've done your homework. For some great tips on how to get your post noticed in the future, please refer to; http://wiki.freeswitch.org/wiki/UsingTheMailingList In ref to your original question, read up on these; http://wiki.freeswitch.org/wiki/Channel_Variables#CDR_related http://wiki.freeswitch.org/wiki/CDR Although I haven't used the channel variables approach, the docs suggest this will stop the A leg from being processed at all. The other option would be to ignore it in post processing. If you are still stuck after reading through those, please reply back and tell us what you have tried, and where you are getting stuck. Hope this helps Cal On Friday, October 25, 2013, Abdullah wrote: > HI , > > PLEASE HELP ME HOW TO DISABLE LEG-A CDR IN FREESWITCH , I JUST WANT B-LEG > CDR IN MY FREESWITCH. > > > THANKS IN ADVANCE . > > > WAITING FOR YOUR RESPONSE.... > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/freeswitch-XML-CDR-LEG-A-DISABLE-tp7595969.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131026/32b3bbba/attachment.html From andretodd at verizon.net Sat Oct 26 18:37:02 2013 From: andretodd at verizon.net (Andre) Date: Sat, 26 Oct 2013 10:37:02 -0400 Subject: [Freeswitch-users] Freeswitch Limit execute_on_originate Question Message-ID: <016601ced258$d63b7960$82b26c20$@verizon.net> Hello, I'm calling execute_on_originate to an app LimitOutbound on my bridge action for LIMIT. var ses = context.Session; ses.Execute("limit", string.Format("hash outbound gw_{0} {1} !NORMAL_TEMPORARY_FAILURE", gwnamne, PortLimit)); ses.Execute("limit", string.Format("hash outbound gw_{0} {1}/1 !NORMAL_TEMPORARY_FAILURE", gwnamne, cps)); When I call it the limit goes back to 0 after the call routes. If I use a new thread :: the numbers don't go back to 0 until after the call however the new thread allows the call to go through and then it hangs up after the limit is checked. Per a post I saw it said to use execute_on_post_originate however that too allows the call to process then hangs up the call if the limit is reached either on single thread or multi-threaded. Since my limit has 3 bridge actions I expect if the first gateway is at limit it would route to the next one until one is available or no gateways are available. The code above (assuming 3 bridges) will call and hang up 3 times. I hope someone can help me. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131026/fefbabd9/attachment-0001.html From 4orbit at gmail.com Sun Oct 27 09:15:00 2013 From: 4orbit at gmail.com (Sergey Zhuravlov) Date: Sun, 27 Oct 2013 10:15:00 +0400 Subject: [Freeswitch-users] raspberry pi B FS latest not start Message-ID: Hi! Do not start on the latest version of FS raspberry pi B. There is a normal start, but still hangs on loading modules and applications with no errors in the log file. This processor is loaded by almost 100% 08:04:14 up 20:47, 3 users, load average: 10.78, 10.20, 9.76 USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 27184 94.8 9.8 58932 44364 pts/3 R Message-ID: Some actual details here would be nice since many of us run FS on Rpi (B Mod Rev2 and Rev1 here) Are you sure you don?t have something like build skew going on? On 10/27/13 1:15 AM, "Sergey Zhuravlov" <4orbit at gmail.com> wrote: > Hi! > > Do not start on the latest version of FS raspberry pi B. > > > There is a normal start, but still hangs on loading modules and applications > with no errors in the log file. > > This processor is loaded by almost 100% > > 08:04:14 up 20:47,? 3 users,? load average: 10.78, 10.20, 9.76 > > USER?????? PID %CPU %MEM??? VSZ?? RSS TTY????? STAT START?? TIME COMMAND > root???? 27184 94.8? 9.8? 58932 44364 pts/3??? R ../bin/freeswitch > > What's changed with the stable version? > > Can not load any modules, and resource-intensive applications? > > At the same time, what a shame on the same raspberry works well with asterisk > 11, FreePBX, web server, MySQL server and others. > > Where is the scalability of the FS in this case? > > I understand that. I'm doing something wrong;-) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131027/08e48166/attachment.html From manavid at gmail.com Sun Oct 27 22:28:34 2013 From: manavid at gmail.com (Moe Navid) Date: Sun, 27 Oct 2013 22:58:34 +0330 Subject: [Freeswitch-users] freeswitch XML CDR LEG A DISABLE In-Reply-To: <1382706793603-7595969.post@n2.nabble.com> References: <1382706793603-7595969.post@n2.nabble.com> Message-ID: have a look here http://wiki.freeswitch.org/wiki/Variable_process_cdr On Fri, Oct 25, 2013 at 4:43 PM, Abdullah wrote: > HI , > > PLEASE HELP ME HOW TO DISABLE LEG-A CDR IN FREESWITCH , I JUST WANT B-LEG > CDR IN MY FREESWITCH. > > > THANKS IN ADVANCE . > > > WAITING FOR YOUR RESPONSE.... > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/freeswitch-XML-CDR-LEG-A-DISABLE-tp7595969.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131027/e0a3a869/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Oct 27 23:10:06 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 27 Oct 2013 20:10:06 +0000 Subject: [Freeswitch-users] freeswitch XML CDR LEG A DISABLE In-Reply-To: References: <1382706793603-7595969.post@n2.nabble.com> Message-ID: Yeah that's the same as mentioned here; http://wiki.freeswitch.org/wiki/Channel_Variables#CDR_related Cal On Sun, Oct 27, 2013 at 7:28 PM, Moe Navid wrote: > have a look here http://wiki.freeswitch.org/wiki/Variable_process_cdr > > > On Fri, Oct 25, 2013 at 4:43 PM, Abdullah wrote: >> >> HI , >> >> PLEASE HELP ME HOW TO DISABLE LEG-A CDR IN FREESWITCH , I JUST WANT B-LEG >> CDR IN MY FREESWITCH. >> >> >> THANKS IN ADVANCE . >> >> >> WAITING FOR YOUR RESPONSE.... >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/freeswitch-XML-CDR-LEG-A-DISABLE-tp7595969.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alipey at gmail.com Mon Oct 28 03:25:24 2013 From: alipey at gmail.com (Ali Pey) Date: Sun, 27 Oct 2013 20:25:24 -0400 Subject: [Freeswitch-users] silence detection not working for record In-Reply-To: References: Message-ID: Hello, I just wanted to provide an update in case anyone else ran into this problem. The reason the silence detection was not working was that I hadn't specified .wav at the end of the file name. So it was giving me this error: *Can't detect silence on a native** recording.* Which means you cannot detect silence on an encoded format because you would have to decode it to be able to detect the silence. If you want to detect silence use a standard wav/pcm format. I added .wav at the end of my file name and it is working now. Regards, Ali Pey On Thu, Oct 24, 2013 at 4:11 PM, Ali Pey wrote: > Hello, > > I have the following line in my perl ESL code to record a message. No > matter how long I wait in mute after recording, it does not stop recording. > > con->execute('record', '${file} 300 200 4' ); > > Please help. > > Thanks, > Ali Pey > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131027/108ee0d1/attachment.html From rgelfand2 at gmail.com Mon Oct 28 03:54:45 2013 From: rgelfand2 at gmail.com (Roman Gelfand) Date: Sun, 27 Oct 2013 20:54:45 -0400 Subject: [Freeswitch-users] ivr application Message-ID: I am new to freeswitch and would like to develop an ivr application. I understand there are many ivr freeswitch-based platforms. Could you point me to the most widely used? Coding examples,etc...? Thanks in advance From ssinyagin at yahoo.com Mon Oct 28 05:28:10 2013 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Sun, 27 Oct 2013 19:28:10 -0700 (PDT) Subject: [Freeswitch-users] ivr application In-Reply-To: References: Message-ID: <1382927290.91318.YahooMailNeo@web126201.mail.ne1.yahoo.com> everything is in the wiki and in books :) Static scenarios with some basic database lookup can be done in XML: https://wiki.freeswitch.org/wiki/IVR_Menu More complex IVR can be programmed in Lua, Javascript, Perl,... http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js https://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example And, before you go any deeper, you will need to read the book: http://www.packtpub.com/freeswitch-1-2/book ________________________________ From: Roman Gelfand To: FreeSWITCH Users Help Sent: Monday, October 28, 2013 1:54 AM Subject: [Freeswitch-users] ivr application I am new to freeswitch and would like to develop an ivr application. I understand there are many ivr freeswitch-based platforms.? Could you point me to the most widely used? Coding examples,etc...? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131027/bd9b4670/attachment-0001.html From zev at machshevet.com Sun Oct 27 18:18:00 2013 From: zev at machshevet.com (Zev Spitz) Date: Sun, 27 Oct 2013 17:18:00 +0200 Subject: [Freeswitch-users] Sound files for mod_say_he Message-ID: According to the link at http://wiki.freeswitch.org/wiki/Mod_say_he, the sound files for mod_say_he will be available soon. The page was last updated 4 April 2011. Has anything happened since? Thanks in advance, Zev Spitz From yehavi.bourvine at gmail.com Mon Oct 28 07:11:44 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 28 Oct 2013 06:11:44 +0200 Subject: [Freeswitch-users] Sound files for mod_say_he In-Reply-To: References: Message-ID: I will check this tomorrow. I'll also send you tmorrow the files to be used in the meantime. __Yehavi: 2013/10/27 Zev Spitz > According to the link at http://wiki.freeswitch.org/wiki/Mod_say_he, > the sound files for mod_say_he will be available soon. The page was > last updated 4 April 2011. Has anything happened since? > > Thanks in advance, > Zev Spitz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/eabd3696/attachment.html From avi at avimarcus.net Mon Oct 28 07:48:17 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 Oct 2013 04:48:17 +0000 Subject: [Freeswitch-users] Sound files for mod_say_he In-Reply-To: References: Message-ID: <00000141fd64194d-deefa96d-9cbe-44fa-95e2-c359d5490e63-000000@email.amazonses.com> It seems the link on the bottom of that page is down. The last hebrew.tar.gz that I have is now available here: http://ge.tt/39aI4Sw/v/0?c It doesn't seem to include the sound xml files, and I'm not sure if they were modified much from the english ones. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/3188574a/attachment.html From vbvbrj at gmail.com Mon Oct 28 10:21:53 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 28 Oct 2013 09:21:53 +0200 Subject: [Freeswitch-users] How to failover with fail_on_single_reject construct In-Reply-To: References: Message-ID: <526E1091.6040509@gmail.com> On 25.10.2013 22:41, Johny Kadarisman Kwan wrote: > Like to report back, in case anybody bumping to the same problem. > > After digging more and looking at debug log. Apparently the culprit is > on the way I set channel variable. On my case, dialstring should be : > > data="{ignore_early_media=true,fail_on_single_reject=^^:NO_ANSWER:ALLOTTED_TIMEOUT}[leg_timeout=20}]sofia/gateway/gw1/7321231234|[leg_timeout=20]sofia/gateway/gw2/7321231234|[leg_timeout=20]sofia/gateway/gw3/7321231234}"/> > > Once that in-place, it work as expected. > > Thats all, my bad, Freeswitch Rock!! Thank you. I am too in implementing this and was wandering how to do it. As I understand, fail_on_single_reject=^^:NO_ANSWER:ALLOTTED_TIMEOUT will make the bridge to try next gateway if there is a network failure with the gateway or its server does not respond, but will not try next gateway if caller does not answer or it is invalid number? Also could you update wiki to reflect this example of yours? -- Mimiko desu. From ivan at c3i.bg Mon Oct 28 10:49:46 2013 From: ivan at c3i.bg (Ivan) Date: Mon, 28 Oct 2013 09:49:46 +0200 Subject: [Freeswitch-users] centos/rhel rpms not signed for 1.2.14 ? Message-ID: <526E171A.3050600@c3i.bg> Hi there, Did something change with how rpm packages are signed ? Yum complains that the 1.2.14 packages are not signed, which is indeed the case. 1.2.12 was OK: $rpm -K freeswitch-1.2.12-1.el6.x86_64.rpm freeswitch-1.2.12-1.el6.x86_64.rpm: (sha1) dsa sha1 md5 gpg OK 1.2.14 is not (missing gpg/dsa flags): $rpm -K freeswitch-1.2.14-1.el6.x86_64.rpm freeswitch-1.2.14-1.el6.x86_64.rpm: sha1 md5 OK I was waiting to see if other people had this problem - it's already 10 days since the rpms are available, so either I'm doing something wrong, or people don't care about package signing (bad!). The yum repo release rpm is set up with gpgcheck=1 but the wiki mentions using "--nogpgcheck" [1]. What's the "official" policy ? cheers ivan [1] http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide From tonybecq at yahoo.fr Mon Oct 28 12:28:43 2013 From: tonybecq at yahoo.fr (obbyone) Date: Mon, 28 Oct 2013 02:28:43 -0700 (PDT) Subject: [Freeswitch-users] Sound issue on bridging one inbound gateway call to another gateway In-Reply-To: <1382692066387-7595967.post@n2.nabble.com> References: <1382692066387-7595967.post@n2.nabble.com> Message-ID: <1382952523864-7595973.post@n2.nabble.com> Hy, I see no answer so maybe you need some more informations : I put here the logs... <<< 2013-10-25 11:22:42.752485 [NOTICE] switch_channel.c:1034 New Channel sofia/external/@ [495c9dd1-1713-4dd1-b41d-cf1a53537e52] 2013-10-25 11:22:42.752485 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.752485 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.752485 [DEBUG] switch_core_state_machine.c:418 (sofia/external/@) Running State Change CS_NEW 2013-10-25 11:22:42.752485 [DEBUG] switch_core_state_machine.c:436 (sofia/external/@) State NEW 2013-10-25 11:22:42.772480 [DEBUG] sofia.c:5834 Channel sofia/external/@ entering state [received][100] 2013-10-25 11:22:42.772480 [DEBUG] sofia.c:5844 Remote SDP: v=0 o=cp10 138269296293 138269296293 IN IP4 10.7.1.129 s=SIP Call c=IN IP4 91.121.128.144 t=0 0 m=audio 38132 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G729:18:8000:30:8000]/[G7221:115:32000:20:48000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G729:18:8000:30:8000]/[G7221:107:16000:20:32000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G729:18:8000:30:8000]/[G722:9:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G729:18:8000:30:8000]/[PCMU:0:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G729:18:8000:30:8000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G729:18:8000:30:8000]/[GSM:3:8000:20:13200] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G723:4:8000:30:6300]/[G7221:115:32000:20:48000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G723:4:8000:30:6300]/[G7221:107:16000:20:32000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G723:4:8000:30:6300]/[G722:9:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [G723:4:8000:30:6300]/[GSM:3:8000:20:13200] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[G7221:115:32000:20:48000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[G7221:107:16000:20:32000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[G7221:115:32000:20:48000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[G7221:107:16000:20:32000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[G722:9:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMU:0:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:1913 Set Codec sofia/external/@ PCMA/8000 20 ms 160 samples 64000 bits 2013-10-25 11:22:42.782468 [DEBUG] switch_core_codec.c:111 sofia/external/@ Original read codec set to PCMA:8 2013-10-25 11:22:42.782468 [DEBUG] switch_core_media.c:3051 Set 2833 dtmf send/recv payload to 101 2013-10-25 11:22:42.782468 [DEBUG] sofia.c:6076 (sofia/external/@) State Change CS_NEW -> CS_INIT 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:418 (sofia/external/@) Running State Change CS_INIT 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:457 (sofia/external/@) State INIT 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:87 sofia/external/@ SOFIA INIT 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:114 (sofia/external/@) State Change CS_INIT -> CS_ROUTING 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:457 (sofia/external/@) State INIT going to sleep 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:418 (sofia/external/@) Running State Change CS_ROUTING 2013-10-25 11:22:42.782468 [DEBUG] switch_channel.c:2120 (sofia/external/@) Callstate Change DOWN -> RINGING 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:473 (sofia/external/@) State ROUTING 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:137 sofia/external/@ SOFIA ROUTING 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:117 sofia/external/@ Standard ROUTING 2013-10-25 11:22:42.782468 [INFO] mod_dialplan_xml.c:558 Processing <>-> in context public Dialplan: sofia/external/@ parsing [public->unloop] continue=false Dialplan: sofia/external/@ Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/@ Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/@ parsing [public->outside_call] continue=true Dialplan: sofia/external/@ Absolute Condition [outside_call] Dialplan: sofia/external/@ Action set(outside_call=true) Dialplan: sofia/external/@ parsing [public->call_debug] continue=true Dialplan: sofia/external/@ Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/@ parsing [public->public_extensions] continue=false Dialplan: sofia/external/@ Regex (FAIL) [public_extensions] destination_number() =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/@ parsing [public->public_did] continue=false Dialplan: sofia/external/@ Regex (FAIL) [public_did] destination_number() =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/@ parsing [public->public_openIP] continue=false Dialplan: sofia/external/@ Regex (FAIL) [public_openIP] destination_number() =~ /^(970710200)$/ break=on-false Dialplan: sofia/external/@ parsing [public->public_ovh] continue=false Dialplan: sofia/external/@ Regex (FAIL) [public_ovh] destination_number() =~ /^()$/ break=on-false Dialplan: sofia/external/@ parsing [public->transfert_08] continue=false Dialplan: sofia/external/@ Regex (PASS) [transfert_08] destination_number() =~ /^()$/ break=on-false Dialplan: sofia/external/@ Action ring_ready() Dialplan: sofia/external/@ Action bridge(sofia/gateway/outbound02_ovh/) 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:167 (sofia/external/@) State Change CS_ROUTING -> CS_EXECUTE 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:473 (sofia/external/@) State ROUTING going to sleep 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:418 (sofia/external/@) Running State Change CS_EXECUTE 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:480 (sofia/external/@) State EXECUTE 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:230 sofia/external/@ SOFIA EXECUTE 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:209 sofia/external/@ Standard EXECUTE EXECUTE sofia/external/@ set(outside_call=true) 2013-10-25 11:22:42.782468 [DEBUG] mod_dptools.c:1395 sofia/external/@ SET [outside_call]=[true] EXECUTE sofia/external/@ ring_ready() 2013-10-25 11:22:42.782468 [NOTICE] mod_sofia.c:2078 Ring-Ready sofia/external/@! 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] sofia.c:5834 Channel sofia/external/@ entering state [early][180] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:865 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:42.782468 [NOTICE] mod_dptools.c:941 Ring Ready sofia/external/@! EXECUTE sofia/external/@ bridge(sofia/gateway/outbound02_ovh/) 2013-10-25 11:22:42.782468 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2013-10-25 11:22:42.782468 [NOTICE] switch_channel.c:1034 New Channel sofia/external/ [e564400f-6932-4d3b-b3cb-ba0c7fdac9df] 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:4480 (sofia/external/) State Change CS_NEW -> CS_INIT 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:418 (sofia/external/) Running State Change CS_INIT 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:457 (sofia/external/) State INIT 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:87 sofia/external/ SOFIA INIT 2013-10-25 11:22:42.782468 [DEBUG] sofia_glue.c:1225 Local SDP: v=0 o=FreeSWITCH 1382661558 1382661559 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 m=audio 31404 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:114 (sofia/external/) State Change CS_INIT -> CS_ROUTING 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:457 (sofia/external/) State INIT going to sleep 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:418 (sofia/external/) Running State Change CS_ROUTING 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:473 (sofia/external/) State ROUTING 2013-10-25 11:22:42.782468 [DEBUG] mod_sofia.c:137 sofia/external/ SOFIA ROUTING 2013-10-25 11:22:42.782468 [DEBUG] switch_ivr_originate.c:67 (sofia/external/) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-10-25 11:22:42.782468 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:473 (sofia/external/) State ROUTING going to sleep 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:418 (sofia/external/) Running State Change CS_CONSUME_MEDIA 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:492 (sofia/external/) State CONSUME_MEDIA 2013-10-25 11:22:42.782468 [DEBUG] switch_core_state_machine.c:492 (sofia/external/) State CONSUME_MEDIA going to sleep 2013-10-25 11:22:42.782468 [DEBUG] sofia.c:5834 Channel sofia/external/ entering state [calling][0] 2013-10-25 11:22:42.792471 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:42.792471 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:42.792471 [DEBUG] sofia.c:5834 Channel sofia/external/ entering state [calling][0] 2013-10-25 11:22:47.812527 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:47.812527 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:47.812527 [DEBUG] sofia.c:5834 Channel sofia/external/ entering state [proceeding][180] 2013-10-25 11:22:47.812527 [DEBUG] sofia.c:5844 Remote SDP: v=0 o=cp10 138269296200 138269296201 IN IP4 10.7.1.133 s=SIP Call c=IN IP4 91.121.128.146 t=0 0 m=audio 33910 RTP/AVP 0 8 101 b=AS:77 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:1913 Set Codec sofia/external/ PCMA/8000 20 ms 160 samples 64000 bits 2013-10-25 11:22:47.812527 [DEBUG] switch_core_codec.c:111 sofia/external/ Original read codec set to PCMA:8 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:3042 Set 2833 dtmf send payload to 101 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:4098 AUDIO RTP [sofia/external/] 91.204.116.116 port 31404 -> 91.121.128.146 port 33910 codec: 8 ms: 20 2013-10-25 11:22:47.812527 [DEBUG] switch_rtp.c:2881 Starting timer [soft] 160 bytes per 20ms 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:4440 Set 2833 dtmf send payload to 101 2013-10-25 11:22:47.812527 [DEBUG] switch_core_media.c:4446 Set 2833 dtmf receive payload to 101 2013-10-25 11:22:47.812527 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/! 2013-10-25 11:22:47.812527 [DEBUG] switch_channel.c:3328 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:47.812527 [DEBUG] switch_channel.c:3332 (sofia/external/) Callstate Change DOWN -> EARLY 2013-10-25 11:22:47.822472 [INFO] switch_ivr_originate.c:3443 Sending early media 2013-10-25 11:22:47.822472 [DEBUG] switch_core_media.c:4098 AUDIO RTP [sofia/external/@] 91.204.116.116 port 28082 -> 91.121.128.144 port 38132 codec: 8 ms: 20 2013-10-25 11:22:47.822472 [DEBUG] switch_rtp.c:2881 Starting timer [soft] 160 bytes per 20ms 2013-10-25 11:22:47.822472 [DEBUG] switch_core_media.c:4440 Set 2833 dtmf send payload to 101 2013-10-25 11:22:47.822472 [DEBUG] switch_core_media.c:4446 Set 2833 dtmf receive payload to 101 2013-10-25 11:22:47.822472 [DEBUG] mod_sofia.c:2143 Ring SDP: v=0 o=FreeSWITCH 1382664885 1382664886 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 m=audio 28082 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-10-25 11:22:47.822472 [NOTICE] mod_sofia.c:2146 Pre-Answer sofia/external/@! 2013-10-25 11:22:47.822472 [DEBUG] switch_channel.c:3332 (sofia/external/@) Callstate Change RINGING -> EARLY 2013-10-25 11:22:47.822472 [DEBUG] switch_core_session.c:865 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:47.822472 [DEBUG] switch_ivr_originate.c:3494 Originate Resulted in Success: [sofia/external/] 2013-10-25 11:22:47.822472 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:47.822472 [DEBUG] sofia.c:5834 Channel sofia/external/@ entering state [early][183] 2013-10-25 11:22:47.822472 [DEBUG] switch_core_session.c:865 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:47.822472 [DEBUG] switch_core_session.c:865 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:47.822472 [DEBUG] switch_ivr_bridge.c:1440 (sofia/external/) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2013-10-25 11:22:47.822472 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:47.822472 [DEBUG] switch_core_state_machine.c:418 (sofia/external/) Running State Change CS_EXCHANGE_MEDIA 2013-10-25 11:22:47.822472 [DEBUG] switch_core_state_machine.c:483 (sofia/external/) State EXCHANGE_MEDIA 2013-10-25 11:22:47.822472 [DEBUG] mod_sofia.c:644 SOFIA EXCHANGE_MEDIA 2013-10-25 11:22:55.972482 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:55.972482 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:55.992491 [DEBUG] sofia.c:5834 Channel sofia/external/ entering state [completing][200] 2013-10-25 11:22:55.992491 [DEBUG] sofia.c:5841 Duplicate SDP v=0 o=cp10 138269296200 138269296201 IN IP4 10.7.1.133 s=SIP Call c=IN IP4 91.121.128.146 t=0 0 m=audio 33910 RTP/AVP 0 8 101 b=AS:77 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-10-25 11:22:55.992491 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:55.992491 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:55.992491 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:55.992491 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:55.992491 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:56.012473 [DEBUG] sofia.c:5834 Channel sofia/external/ entering state [ready][200] 2013-10-25 11:22:56.012473 [DEBUG] switch_channel.c:3567 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.012473 [NOTICE] sofia.c:6526 Channel [sofia/external/] has been answered 2013-10-25 11:22:56.012473 [DEBUG] switch_channel.c:3613 (sofia/external/) Callstate Change EARLY -> ACTIVE 2013-10-25 11:22:56.012473 [DEBUG] sofia.c:5834 Channel sofia/external/ entering state [ready][200] 2013-10-25 11:22:56.012473 [DEBUG] sofia.c:5844 Remote SDP: v=0 o=cp10 138269296200 138269296202 IN IP4 10.7.1.133 s=SIP Call c=IN IP4 91.121.128.146 t=0 0 m=audio 33910 RTP/AVP 0 8 101 b=AS:77 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-10-25 11:22:56.012473 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:56.012473 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:56.012473 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:56.012473 [DEBUG] switch_core_media.c:2961 Substituting codec PCMA at 30i@8000h 2013-10-25 11:22:56.012473 [DEBUG] switch_core_media.c:1822 Changing Codec from PCMA at 20ms@8000hz to PCMA at 30ms@8000hz 2013-10-25 11:22:56.032474 [DEBUG] mod_sofia.c:824 Local SDP sofia/external/@: v=0 o=FreeSWITCH 1382664885 1382664887 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 m=audio 28082 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-10-25 11:22:56.032474 [DEBUG] switch_core_session.c:865 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.032474 [NOTICE] switch_ivr_bridge.c:484 Channel [sofia/external/@] has been answered 2013-10-25 11:22:56.032474 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.032474 [DEBUG] switch_channel.c:3613 (sofia/external/@) Callstate Change EARLY -> ACTIVE 2013-10-25 11:22:56.032474 [DEBUG] sofia.c:5834 Channel sofia/external/@ entering state [completed][200] 2013-10-25 11:22:56.032474 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.032474 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.032474 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.052478 [DEBUG] sofia.c:5834 Channel sofia/external/@ entering state [ready][200] 2013-10-25 11:22:56.052478 [DEBUG] switch_core_session.c:927 Send signal sofia/external/ [BREAK] 2013-10-25 11:22:56.052478 [DEBUG] switch_core_session.c:927 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.052478 [DEBUG] switch_rtp.c:2764 RE-Starting timer [soft] 240 bytes per 30ms 2013-10-25 11:22:56.052478 [DEBUG] switch_core_media.c:1913 Set Codec sofia/external/ PCMA/8000 30 ms 240 samples 64000 bits 2013-10-25 11:22:56.052478 [DEBUG] switch_core_codec.c:123 sofia/external/ Original read codec replaced with PCMA:8 2013-10-25 11:22:56.052478 [DEBUG] switch_core_media.c:3042 Set 2833 dtmf send payload to 101 2013-10-25 11:22:56.052478 [DEBUG] sofia.c:6429 Processing updated SDP 2013-10-25 11:22:56.052478 [DEBUG] switch_core_media.c:4068 Audio params are unchanged for sofia/external/. 2013-10-25 11:22:56.052478 [DEBUG] switch_core_media.c:4078 sofia/external/ Setting audio receive payload in Re-INVITE to 8 2013-10-25 11:22:56.052478 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.052478 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.052478 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:22:56.072495 [DEBUG] sofia.c:5834 Channel sofia/external/@ entering state [ready][200] 2013-10-25 11:22:56.072495 [DEBUG] sofia.c:5844 Remote SDP: v=0 o=cp10 138269296293 138269296295 IN IP4 10.7.1.129 s=SIP Call c=IN IP4 91.121.128.144 t=0 0 m=audio 38132 RTP/AVP 8 101 b=AS:77 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2013-10-25 11:22:56.072495 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:56.072495 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000] 2013-10-25 11:22:56.072495 [DEBUG] switch_core_media.c:2961 Substituting codec PCMA at 30i@8000h 2013-10-25 11:22:56.072495 [DEBUG] switch_core_media.c:1822 Changing Codec from PCMA at 20ms@8000hz to PCMA at 30ms@8000hz 2013-10-25 11:22:56.112493 [DEBUG] switch_rtp.c:2764 RE-Starting timer [soft] 240 bytes per 30ms 2013-10-25 11:22:56.112493 [DEBUG] switch_core_media.c:1913 Set Codec sofia/external/@ PCMA/8000 30 ms 240 samples 64000 bits 2013-10-25 11:22:56.112493 [DEBUG] switch_core_codec.c:123 sofia/external/@ Original read codec replaced with PCMA:8 2013-10-25 11:22:56.112493 [DEBUG] switch_core_media.c:3051 Set 2833 dtmf send/recv payload to 101 2013-10-25 11:22:56.112493 [DEBUG] sofia.c:6429 Processing updated SDP 2013-10-25 11:22:56.112493 [DEBUG] switch_core_media.c:4068 Audio params are unchanged for sofia/external/@. 2013-10-25 11:22:56.112493 [DEBUG] switch_core_media.c:4078 sofia/external/@ Setting audio receive payload in Re-INVITE to 8 2013-10-25 11:23:11.432467 [DEBUG] switch_core_session.c:1010 Send signal sofia/external/@ [BREAK] 2013-10-25 11:23:11.462472 [NOTICE] sofia.c:715 Hangup sofia/external/@ [CS_EXECUTE] [NORMAL_CLEARING] 2013-10-25 11:23:11.462472 [DEBUG] switch_channel.c:3139 Send signal sofia/external/@ [KILL] 2013-10-25 11:23:11.462472 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/@ [BREAK] 2013-10-25 11:23:11.462472 [DEBUG] switch_ivr_bridge.c:647 BRIDGE THREAD DONE [sofia/external/@] 2013-10-25 11:23:11.462472 [DEBUG] switch_ivr_bridge.c:672 Send signal sofia/external/ [BREAK] 2013-10-25 11:23:11.492469 [DEBUG] switch_ivr_bridge.c:647 BRIDGE THREAD DONE [sofia/external/] 2013-10-25 11:23:11.492469 [DEBUG] switch_ivr_bridge.c:672 Send signal sofia/external/@ [BREAK] 2013-10-25 11:23:11.492469 [NOTICE] switch_ivr_bridge.c:735 Hangup sofia/external/ [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-10-25 11:23:11.492469 [DEBUG] switch_channel.c:3139 Send signal sofia/external/ [KILL] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:483 (sofia/external/) State EXCHANGE_MEDIA going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:418 (sofia/external/) Running State Change CS_HANGUP 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:681 (sofia/external/) State HANGUP 2013-10-25 11:23:11.492469 [DEBUG] mod_sofia.c:459 sofia/external/ Overriding SIP cause 480 with 200 from the other leg 2013-10-25 11:23:11.492469 [DEBUG] mod_sofia.c:465 Channel sofia/external/ hanging up, cause: NORMAL_CLEARING 2013-10-25 11:23:11.492469 [DEBUG] mod_sofia.c:517 Sending BYE to sofia/external/ 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:48 sofia/external/ Standard HANGUP, cause: NORMAL_CLEARING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:681 (sofia/external/) State HANGUP going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:694 (sofia/external/) Callstate Change ACTIVE -> HANGUP 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:449 (sofia/external/) State Change CS_HANGUP -> CS_REPORTING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:418 (sofia/external/) Running State Change CS_REPORTING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:766 (sofia/external/) State REPORTING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:92 sofia/external/ Standard REPORTING, cause: NORMAL_CLEARING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:766 (sofia/external/) State REPORTING going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:443 (sofia/external/) State Change CS_REPORTING -> CS_DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/ [BREAK] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1553 Session 605 (sofia/external/) Locked, Waiting on external entities 2013-10-25 11:23:11.492469 [DEBUG] switch_ivr_bridge.c:1541 sofia/external/@ skip receive message [UNBRIDGE] (channel is hungup already) 2013-10-25 11:23:11.492469 [NOTICE] switch_core_session.c:1571 Session 605 (sofia/external/) Ended 2013-10-25 11:23:11.492469 [NOTICE] switch_core_session.c:1575 Close Channel sofia/external/ [CS_DESTROY] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:2814 sofia/external/@ skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:480 (sofia/external/@) State EXECUTE going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:418 (sofia/external/@) Running State Change CS_HANGUP 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:681 (sofia/external/@) State HANGUP 2013-10-25 11:23:11.492469 [DEBUG] mod_sofia.c:465 Channel sofia/external/@ hanging up, cause: NORMAL_CLEARING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:48 sofia/external/@ Standard HANGUP, cause: NORMAL_CLEARING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:681 (sofia/external/@) State HANGUP going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:568 (sofia/external/) Callstate Change HANGUP -> DOWN 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:694 (sofia/external/@) Callstate Change ACTIVE -> HANGUP 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:449 (sofia/external/@) State Change CS_HANGUP -> CS_REPORTING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/@ [BREAK] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:418 (sofia/external/@) Running State Change CS_REPORTING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:571 (sofia/external/) Running State Change CS_DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:766 (sofia/external/@) State REPORTING 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:581 (sofia/external/) State DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:92 sofia/external/@ Standard REPORTING, cause: NORMAL_CLEARING 2013-10-25 11:23:11.492469 [DEBUG] mod_sofia.c:375 sofia/external/ SOFIA DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:766 (sofia/external/@) State REPORTING going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:99 sofia/external/ Standard DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:581 (sofia/external/) State DESTROY going to sleep 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:443 (sofia/external/@) State Change CS_REPORTING -> CS_DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1345 Send signal sofia/external/@ [BREAK] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_session.c:1553 Session 604 (sofia/external/@) Locked, Waiting on external entities 2013-10-25 11:23:11.492469 [NOTICE] switch_core_session.c:1571 Session 604 (sofia/external/@) Ended 2013-10-25 11:23:11.492469 [NOTICE] switch_core_session.c:1575 Close Channel sofia/external/@ [CS_DESTROY] 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:568 (sofia/external/@) Callstate Change HANGUP -> DOWN 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:571 (sofia/external/@) Running State Change CS_DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:581 (sofia/external/@) State DESTROY 2013-10-25 11:23:11.492469 [DEBUG] mod_sofia.c:375 sofia/external/@ SOFIA DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:99 sofia/external/@ Standard DESTROY 2013-10-25 11:23:11.492469 [DEBUG] switch_core_state_machine.c:581 (sofia/external/@) State DESTROY going to sleep >>> If someone understand what happens... Thanks in advance... Obbyone -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sound-issue-on-bridging-one-inbound-gateway-call-to-another-gateway-tp7595967p7595973.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gmangudai at gmail.com Mon Oct 28 13:18:30 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Mon, 28 Oct 2013 18:18:30 +0800 Subject: [Freeswitch-users] SRTP protection failed with code 9 Message-ID: im using TLS + SRTP, when making calls, im getting "SRTP protection failed with code 9" error at the FS console all the time these days, the call is normal, but no SRTP, why is this happening? the log is pastebined at http://pastebin.freeswitch.org/21578 any response is appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/123053e4/attachment.html From Isatya1992 at gmail.com Mon Oct 28 12:32:27 2013 From: Isatya1992 at gmail.com (satya) Date: Mon, 28 Oct 2013 02:32:27 -0700 (PDT) Subject: [Freeswitch-users] connect two freeswitch boxes Message-ID: <1382952747240-7595974.post@n2.nabble.com> Am trying to connect 2 freeswitch boxes following the procedure given in this link,http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes . I am getting the following console log when I try to make a call.http://pastebin.freeswitch.org/21577 . I am a newbie, please help. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/connect-two-freeswitch-boxes-tp7595974.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Mon Oct 28 16:06:12 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Oct 2013 09:06:12 -0400 Subject: [Freeswitch-users] Sound files for mod_say_he In-Reply-To: <00000141fd64194d-deefa96d-9cbe-44fa-95e2-c359d5490e63-000000@email.amazonses.com> References: <00000141fd64194d-deefa96d-9cbe-44fa-95e2-c359d5490e63-000000@email.amazonses.com> Message-ID: <30ED17C4-1163-43CB-89EC-49CD57A91710@jerris.com> If someone wants to fill out this sound set, we can distribute it along side the others we already distribute w/ FreeSWITCH. Who originally created this set? Mike On Oct 28, 2013, at 12:48 AM, Avi Marcus wrote: > It seems the link on the bottom of that page is down. > The last hebrew.tar.gz that I have is now available here: http://ge.tt/39aI4Sw/v/0?c > It doesn't seem to include the sound xml files, and I'm not sure if they were modified much from the english ones. > > -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/9c9d688f/attachment.html From brian at freeswitch.org Mon Oct 28 16:18:05 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Oct 2013 08:18:05 -0500 Subject: [Freeswitch-users] SRTP protection failed with code 9 In-Reply-To: References: Message-ID: <29AEA2F5-D025-4E71-BE8D-4263215030A1@freeswitch.org> What endpoint are you using? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 28, 2013, at 5:18 AM, Vincent Xia wrote: > im using TLS + SRTP, when making calls, im getting "SRTP protection failed with code 9" error at the FS console > > all the time these days, the call is normal, but no SRTP, why is this happening? > > the log is pastebined at http://pastebin.freeswitch.org/21578 > > any response is appreciated. > -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/41aefb0e/attachment.bin From brian at freeswitch.org Mon Oct 28 16:20:23 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Oct 2013 08:20:23 -0500 Subject: [Freeswitch-users] FS + ZRTP + SAS In-Reply-To: References: Message-ID: I hear that c-sipsimple IS broken along with many other endpoints. Every time someone says this is broken, I spend an hour recompiling, testing and verifying it and not once has it been broken. I wrote it and tested it with Acrobits/Groundwire and it still works flawlessly with those endpoints? So if it doesn?t work with yours then chances are its your endpoint thats broken. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 25, 2013, at 10:40 AM, Yuriy Nasida wrote: > Hi guys! > > I have problem with transmission of SAS (Short Authentication String). > FS version 1.2.10 > > I enabled > > but SAS strings is still different. > > I also tried but this didn't help. > > I heard that it is fixed with 1.3.X branch. Can somebody please confirm or deny this? > > Please advice. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/406860c5/attachment.bin From brian at freeswitch.org Mon Oct 28 16:49:28 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Oct 2013 08:49:28 -0500 Subject: [Freeswitch-users] connect two freeswitch boxes In-Reply-To: <1382952747240-7595974.post@n2.nabble.com> References: <1382952747240-7595974.post@n2.nabble.com> Message-ID: From the pastebin it looks like you?re using port audio to place a call to a locally registered endpoint, Can you crank up the debug ?/log debug? at fs_cli and paste results? Or better yet join #freeswitch on irc.freenode.net https://wiki.freeswitch.org/wiki/Main_Page#Community_and_Support -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Oct 28, 2013, at 4:32 AM, satya wrote: > Am trying to connect 2 freeswitch boxes following the procedure given in this > link,http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes . I am > getting the following console log when I try to make a > call.http://pastebin.freeswitch.org/21577 . I am a newbie, please help. > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/connect-two-freeswitch-boxes-tp7595974.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 841 bytes Desc: Message signed with OpenPGP using GPGMail Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/82a38640/attachment.bin From 4orbit at gmail.com Mon Oct 28 18:21:07 2013 From: 4orbit at gmail.com (Sergey Zhuravlov) Date: Mon, 28 Oct 2013 19:21:07 +0400 Subject: [Freeswitch-users] raspberry pi B FS latest not start Message-ID: Hi Twice tried to run the latest - both without success Compiles fine Before and after compiling and successfully launched a stable version -- WBR, Sergey GTALK/JABBER:4orbit at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/47d8f70c/attachment.html From jason.holden at start.ca Mon Oct 28 20:38:30 2013 From: jason.holden at start.ca (Jason Holden) Date: Mon, 28 Oct 2013 13:38:30 -0400 Subject: [Freeswitch-users] problem with group call Message-ID: <5C3AF1E028F44B5292CBDEBBD9CA8513@bob> I currently have a freeswitch running 1.3.5 and am having the following problem. When bridging to a group only the first extension in the directory is rining. I've tried added the +A flag with no success. Any suggestions on how I can have all pointers in the group call ring? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/e740cd27/attachment-0001.html From yehavi.bourvine at gmail.com Mon Oct 28 21:29:54 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 28 Oct 2013 20:29:54 +0200 Subject: [Freeswitch-users] Sound files for mod_say_he In-Reply-To: <30ED17C4-1163-43CB-89EC-49CD57A91710@jerris.com> References: <00000141fd64194d-deefa96d-9cbe-44fa-95e2-c359d5490e63-000000@email.amazonses.com> <30ED17C4-1163-43CB-89EC-49CD57A91710@jerris.com> Message-ID: Hi, The original code and sound files have been created at the Hebrew University by me and a co-worker; some minor changes have been done at the code by others when it has been entered into the official code base. I am willing to add the sound files; how do we do that? Thanks, __Yehavi: 2013/10/28 Michael Jerris > If someone wants to fill out this sound set, we can distribute it along > side the others we already distribute w/ FreeSWITCH. Who originally > created this set? > > Mike > > On Oct 28, 2013, at 12:48 AM, Avi Marcus wrote: > > It seems the link on the bottom of that page is down. > The last hebrew.tar.gz that I have is now available here: > http://ge.tt/39aI4Sw/v/0?c > It doesn't seem to include the sound xml files, and I'm not sure if they > were modified much from the english ones. > > -Avi > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/6b7b06ec/attachment.html From olegstolyar at gmail.com Mon Oct 28 21:38:16 2013 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Mon, 28 Oct 2013 11:38:16 -0700 Subject: [Freeswitch-users] Adding headers to provisional responses Message-ID: Hi, I am trying to add a header to my responses according to this: http://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Response_Headers When I add a sip_rh_ header, it appears fine in my 200 OK response. However, when I try to add a sip_ph_ header, it does not appear in the 100 Trying response. Am I doing something wrong or is there a bug? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/e42bb668/attachment.html From mike at jerris.com Mon Oct 28 21:43:12 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Oct 2013 14:43:12 -0400 Subject: [Freeswitch-users] Sound files for mod_say_he In-Reply-To: References: <00000141fd64194d-deefa96d-9cbe-44fa-95e2-c359d5490e63-000000@email.amazonses.com> <30ED17C4-1163-43CB-89EC-49CD57A91710@jerris.com> Message-ID: Ken Rice should be able to coordinate getting those files into our build and packaging system with you. Mike On Oct 28, 2013, at 2:29 PM, Yehavi Bourvine wrote: > Hi, > > The original code and sound files have been created at the Hebrew University by me and a co-worker; some minor changes have been done at the code by others when it has been entered into the official code base. > > I am willing to add the sound files; how do we do that? > > Thanks, __Yehavi: > > > 2013/10/28 Michael Jerris > If someone wants to fill out this sound set, we can distribute it along side the others we already distribute w/ FreeSWITCH. Who originally created this set? > > Mike > > On Oct 28, 2013, at 12:48 AM, Avi Marcus wrote: > >> It seems the link on the bottom of that page is down. >> The last hebrew.tar.gz that I have is now available here: http://ge.tt/39aI4Sw/v/0?c >> It doesn't seem to include the sound xml files, and I'm not sure if they were modified much from the english ones. >> >> -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/04cc1c26/attachment.html From krice at freeswitch.org Mon Oct 28 21:47:49 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 28 Oct 2013 13:47:49 -0500 Subject: [Freeswitch-users] Sound files for mod_say_he In-Reply-To: Message-ID: If you have all the sound files we need to get a copy of those and get them into the same distribution format as the other languages sound files... Contact me off list and we?ll get them on the CDN so that its easy to distribute them K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch On 10/28/13 1:29 PM, "Yehavi Bourvine" wrote: > Hi, > ? > ?The original code and sound files have been created at the Hebrew University > by me and a co-worker; some minor changes have been done at the code by others > when it has been entered into the official code base. > ? > I am willing to add the sound files; how do we do that? > ? > ???????????????????? Thanks, __Yehavi: > > > 2013/10/28 Michael Jerris >> If someone wants to fill out this sound set, we can distribute it along side >> the others we already distribute w/ FreeSWITCH. ?Who originally created this >> set? >> >> Mike >> >> On Oct 28, 2013, at 12:48 AM, Avi Marcus wrote: >> >>> It seems the link on the bottom of that page is down. >>> The last hebrew.tar.gz that I have is now available here: >>> http://ge.tt/39aI4Sw/v/0?c >>> It doesn't seem to include the sound xml files, and I'm not sure if they >>> were modified much from the english ones. >>> >>> -Avi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/cb8ec800/attachment-0001.html From fs-list at communicatefreely.net Tue Oct 29 00:23:56 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 28 Oct 2013 17:23:56 -0400 Subject: [Freeswitch-users] Delete unused voice mail files Message-ID: <526ED5EC.8000405@communicatefreely.net> Hello, I'm trying to find a good solution for distributed voice mail storage. Multi-master database seems to work well for the metadata, but I have yet to find an NFS based storage system that can do replication between sites, with both systems in use. I'm thinking of having a script fire after a message is left that will replicated new messages to all other systems. Since the filenames are unique, there is no danger in replacing other files, but messages would never be deleted. Is there a way to have FreeSWITCH compare the list of messages in the database with the voice mail directory and delete any files that are not referenced, or is that better done externally by looking through the database? Likewise, if someone knows of some kind of NFS server that will automatically copy or delete files to a mirrored system, that would be a better solution. The catch seems to be that I want to be able to read or write to EITHER system in the pair. Thanks! -Tim From krice at freeswitch.org Tue Oct 29 00:56:56 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 28 Oct 2013 16:56:56 -0500 Subject: [Freeswitch-users] Delete unused voice mail files In-Reply-To: <526ED5EC.8000405@communicatefreely.net> Message-ID: Are you using master-master or master-standby... If the later, why not just use rsync On 10/28/13 4:23 PM, "Tim St. Pierre" wrote: > Hello, > > I'm trying to find a good solution for distributed voice mail storage. > > Multi-master database seems to work well for the metadata, but I have > yet to find an NFS based storage system that can do replication between > sites, with both systems in use. > > I'm thinking of having a script fire after a message is left that will > replicated new messages to all other systems. Since the filenames are > unique, there is no danger in replacing other files, but messages would > never be deleted. > > Is there a way to have FreeSWITCH compare the list of messages in the > database with the voice mail directory and delete any files that are not > referenced, or is that better done externally by looking through the > database? > > Likewise, if someone knows of some kind of NFS server that will > automatically copy or delete files to a mirrored system, that would be a > better solution. The catch seems to be that I want to be able to read > or write to EITHER system in the pair. > > Thanks! > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch From richard.mace at gmail.com Tue Oct 29 01:34:53 2013 From: richard.mace at gmail.com (Richard Mace) Date: Mon, 28 Oct 2013 22:34:53 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers Message-ID: Hi, Can anyone recommend any SIP trunk providers, ideally based in the UK? Thanks Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/70948dea/attachment.html From dwiyulianto.anto at gmail.com Tue Oct 29 01:41:32 2013 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Tue, 29 Oct 2013 05:41:32 +0700 Subject: [Freeswitch-users] anyone can help me to configure TLS + ZRTP Message-ID: i've try configure TLS and zrtp in freeswitch, but i've problem with TLS, i've using tutorial from this website wiki.freeswitch about SIP_TLS. when i activated in my softphone (i'm use phonerlite) it say TLS registered, but there is no sound feedback. i try using echo test by calling 9196 but there is no sound feedback. but, if i'm use TCP or UDP , there is no problem with that, i can hear voice feedback when try call 9196. someone told me to use ssldump about the result is in attahcment in this email is there any configuration when wanna try using TLS? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/14ddf3bb/attachment.html -------------- next part -------------- New TCP connection #2: freeswitch.org(58950) <-> HP-PC.local(5061) 2 1 0.0005 (0.0005) C>S Handshake ClientHello Version 3.2 cipher suites Unknown value 0xc014 Unknown value 0xc00a Unknown value 0xc022 Unknown value 0xc021 TLS_DHE_RSA_WITH_AES_256_CBC_SHA TLS_DHE_DSS_WITH_AES_256_CBC_SHA Unknown value 0x88 Unknown value 0x87 Unknown value 0xc019 Unknown value 0xc020 Unknown value 0xc00f Unknown value 0xc005 TLS_RSA_WITH_AES_256_CBC_SHA Unknown value 0x84 Unknown value 0xc012 Unknown value 0xc008 Unknown value 0xc01c Unknown value 0xc01b TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA Unknown value 0xc017 Unknown value 0xc01a Unknown value 0xc00d Unknown value 0xc003 TLS_RSA_WITH_3DES_EDE_CBC_SHA Unknown value 0xc013 Unknown value 0xc009 Unknown value 0xc01f Unknown value 0xc01e TLS_DHE_RSA_WITH_AES_128_CBC_SHA TLS_DHE_DSS_WITH_AES_128_CBC_SHA Unknown value 0x9a Unknown value 0x99 Unknown value 0x45 Unknown value 0x44 Unknown value 0xc018 Unknown value 0xc01d Unknown value 0xc00e Unknown value 0xc004 TLS_RSA_WITH_AES_128_CBC_SHA Unknown value 0x96 Unknown value 0x41 Unknown value 0xc011 Unknown value 0xc007 Unknown value 0xc016 Unknown value 0xc00c Unknown value 0xc002 TLS_RSA_WITH_RC4_128_SHA Unknown value 0xff compression methods NULL 2 2 0.0076 (0.0071) S>C Handshake ServerHello Version 3.2 session_id[0]= cipherSuite TLS_RSA_WITH_AES_256_CBC_SHA compressionMethod NULL 2 3 0.0084 (0.0008) S>C Handshake Certificate 2 4 0.0084 (0.0000) S>C Handshake ServerHelloDone 2 5 0.0130 (0.0046) C>S Handshake ClientKeyExchange 2 6 0.0130 (0.0000) C>S ChangeCipherSpec 2 7 0.0130 (0.0000) C>S Handshake 2 8 0.0194 (0.0063) S>C Handshake Segmentation fault (core dumped) From cal.leeming at simplicitymedialtd.co.uk Tue Oct 29 03:29:42 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Oct 2013 00:29:42 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: Hi Richard, This topic was discussed a few months back, you can read the thread here; http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none Since my last post, our chosen UK providers are Magrathea and Simwood. This inter-op was over a period of 6 months but based on low traffic (<=500 calls a month), so others with higher throughput may wish to chime in. Magrathea - Over a period of one month, I've not had a single problematic call, routing/quality is always good, and they seem to have almost every dialing code possible for DIDs. I have not dealt with their technical staff yet, but their sales people were friendly and knowledgeable. Simwood - Over a period of two months, I had 2 call routing issues to international destinations and they are missing at least one number range (Bournemouth). However it's worth noting that Simwood were very quick to help, they identified the routing problem within 5 minutes of me reporting it, and despite being a low profit customer they treated us as a high priority. I've also spoken to numerous staff there, they always come across as friendly and, more importantly, competent. I would have no problem recommending Simwood to others. For international routing - I'm still in the process of reviewing TelServ and Flowroute, I'd like to use Voxbone but their min spend is too high for us. I cannot choose between Sim/Mag, they are both extremely good companies - use both (redundancy is a good thing anyway!) Cal On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace wrote: > Hi, > Can anyone recommend any SIP trunk providers, ideally based in the UK? > > Thanks > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gmangudai at gmail.com Tue Oct 29 03:31:17 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Tue, 29 Oct 2013 08:31:17 +0800 Subject: [Freeswitch-users] SRTP protection failed with code 9 In-Reply-To: <29AEA2F5-D025-4E71-BE8D-4263215030A1@freeswitch.org> References: <29AEA2F5-D025-4E71-BE8D-4263215030A1@freeswitch.org> Message-ID: in-house softphone using osip 2013/10/28 Brian West > What endpoint are you using? > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > On Oct 28, 2013, at 5:18 AM, Vincent Xia wrote: > > > im using TLS + SRTP, when making calls, im getting "SRTP protection > failed with code 9" error at the FS console > > > > all the time these days, the call is normal, but no SRTP, why is this > happening? > > > > the log is pastebined at http://pastebin.freeswitch.org/21578 > > > > any response is appreciated. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/698e7f48/attachment.html From valter at fastway.com.br Tue Oct 29 04:30:46 2013 From: valter at fastway.com.br (Valter Nogueira) Date: Mon, 28 Oct 2013 23:30:46 -0200 Subject: [Freeswitch-users] dev environment Message-ID: I want to setup emacs and related tools in order to edit and mainly understand source code. Is there any tutorial about this? thanks Valter * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131028/43fcbfde/attachment-0001.html From dwiyulianto.anto at gmail.com Tue Oct 29 08:09:54 2013 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Tue, 29 Oct 2013 12:09:54 +0700 Subject: [Freeswitch-users] SRTP protection failed with code 9 In-Reply-To: References: <29AEA2F5-D025-4E71-BE8D-4263215030A1@freeswitch.org> Message-ID: can u tell me how to configure TLS? im using tutorial from wiki but that didnt work for me, im using softphone phonerlite. On 10/29/13, Vincent Xia wrote: > in-house softphone using osip > > > 2013/10/28 Brian West > >> What endpoint are you using? >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH_Wire , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> On Oct 28, 2013, at 5:18 AM, Vincent Xia wrote: >> >> > im using TLS + SRTP, when making calls, im getting "SRTP protection >> failed with code 9" error at the FS console >> > >> > all the time these days, the call is normal, but no SRTP, why is this >> happening? >> > >> > the log is pastebined at http://pastebin.freeswitch.org/21578 >> > >> > any response is appreciated. >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From harsimran2201 at gmail.com Tue Oct 29 09:07:04 2013 From: harsimran2201 at gmail.com (Harsimran singh) Date: Tue, 29 Oct 2013 11:37:04 +0530 Subject: [Freeswitch-users] Conference session doesn't get end in case of incoming conference. Message-ID: Hi, I am creating conference with dtmf facility . When session is created with outgoing call through freeswitch then conference session gets end when no member remains in the session or when no one picks the call , but the same scenario is not happening with the incoming conference session . I am using the same code for both the cases . Can anyone help me out in this ? With Regards Harsimran Singh +91-9711271158 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/f3e6d68e/attachment.html From 4orbit at gmail.com Tue Oct 29 10:05:14 2013 From: 4orbit at gmail.com (Sergey Zhuravlov) Date: Tue, 29 Oct 2013 11:05:14 +0400 Subject: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle Message-ID: Hi! Does somebody have a positive experience with Raspberry Pi, mod_gsmopen GSM dongle ??? I use Raspberry Pi B and e1550. It's a shame that with an asterisk all works fine -- RasPBX. http://www.raspberry-asterisk.org/ But I prefer the FreeSWITCH ;-) On the basis of experience, I thought that PS will work best for this modest hardware faster and require fewer resources. But it turns out it is not. Asterisk with a web server, MySQL database, and other, less load than FS. In this connection, other parameters may be used starting FS to save resources? And the main question! Silence an incoming call on 5000 (IVR) as well as an outgoing in gsm network. Those are not sound. -- WBR, Sergey GTALK/JABBER:4orbit at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/bdc5da9e/attachment.html From andrew at cassidywebservices.co.uk Tue Oct 29 11:05:24 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 29 Oct 2013 08:05:24 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: Magrathea are pretty much the industry standard for volume. Depending on your requirements, there are many others such as SipGate and Gradwell. There are also people from at least 3 UK trunk providers in this list, including SureVoip, NumberGroup and PowerTel. On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi Richard, > > This topic was discussed a few months back, you can read the thread here; > > > http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results > > http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none > > Since my last post, our chosen UK providers are Magrathea and Simwood. > > This inter-op was over a period of 6 months but based on low traffic > (<=500 calls a month), so others with higher throughput may wish to > chime in. > > Magrathea - Over a period of one month, I've not had a single > problematic call, routing/quality is always good, and they seem to > have almost every dialing code possible for DIDs. I have not dealt > with their technical staff yet, but their sales people were friendly > and knowledgeable. > > Simwood - Over a period of two months, I had 2 call routing issues to > international destinations and they are missing at least one number > range (Bournemouth). However it's worth noting that Simwood were very > quick to help, they identified the routing problem within 5 minutes of > me reporting it, and despite being a low profit customer they treated > us as a high priority. I've also spoken to numerous staff there, they > always come across as friendly and, more importantly, competent. I > would have no problem recommending Simwood to others. > > For international routing - I'm still in the process of reviewing > TelServ and Flowroute, I'd like to use Voxbone but their min spend is > too high for us. > > I cannot choose between Sim/Mag, they are both extremely good > companies - use both (redundancy is a good thing anyway!) > > Cal > > On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace > wrote: > > Hi, > > Can anyone recommend any SIP trunk providers, ideally based in the UK? > > > > Thanks > > > > Richard > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/0eab83df/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Oct 29 11:42:12 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Oct 2013 08:42:12 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: Let's be clear on definitions here; BT, Kingston - Top of the food chain, only financially viable for those with lots of cash and traffic Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many millions of minutes per month. SipGate, SureVoip, Gradwell, GBC - These providers are designed for consumers, only necessary if you are not running your own equipment NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be more blunt, an unnecessary middleman If someone is running their own switch, why would you want to go with a consumer grade service which not only limits your CLI capabilities, but also increases complexity, affects latency and inflates call rates. Likewise, why would you go through a middleman when you could just go direct to the supplier with no minimum spend on outbound traffic? Though typically inbound does require a small min spend of around ?50/month (which is tiny in comparison with VoxBone who charge a min spend of around ?600/month [needs clarification]), which is really the only valid reason (that I can think of right now) to go with any of these reseller aggregates. Cal On Tue, Oct 29, 2013 at 8:05 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Magrathea are pretty much the industry standard for volume. > > Depending on your requirements, there are many others such as SipGate and > Gradwell. There are also people from at least 3 UK trunk providers in this > list, including SureVoip, NumberGroup and PowerTel. > > > On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hi Richard, >> >> This topic was discussed a few months back, you can read the thread here; >> >> >> http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results >> >> http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none >> >> Since my last post, our chosen UK providers are Magrathea and Simwood. >> >> This inter-op was over a period of 6 months but based on low traffic >> (<=500 calls a month), so others with higher throughput may wish to >> chime in. >> >> Magrathea - Over a period of one month, I've not had a single >> problematic call, routing/quality is always good, and they seem to >> have almost every dialing code possible for DIDs. I have not dealt >> with their technical staff yet, but their sales people were friendly >> and knowledgeable. >> >> Simwood - Over a period of two months, I had 2 call routing issues to >> international destinations and they are missing at least one number >> range (Bournemouth). However it's worth noting that Simwood were very >> quick to help, they identified the routing problem within 5 minutes of >> me reporting it, and despite being a low profit customer they treated >> us as a high priority. I've also spoken to numerous staff there, they >> always come across as friendly and, more importantly, competent. I >> would have no problem recommending Simwood to others. >> >> For international routing - I'm still in the process of reviewing >> TelServ and Flowroute, I'd like to use Voxbone but their min spend is >> too high for us. >> >> I cannot choose between Sim/Mag, they are both extremely good >> companies - use both (redundancy is a good thing anyway!) >> >> Cal >> >> On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace >> wrote: >> > Hi, >> > Can anyone recommend any SIP trunk providers, ideally based in the UK? >> > >> > Thanks >> > >> > Richard >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/e045b1a9/attachment-0001.html From andrew at cassidywebservices.co.uk Tue Oct 29 12:39:18 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 29 Oct 2013 09:39:18 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: I think the point to be made here is that your choice should be determined by your requirements and budget. As we know neither of those, we can't give you specific recommendations. The majority of people I encounter use 'consumer' grade services, with or without their own equipment as necessary. That said these are largely small businesses. In all seriousness there's always a best tool for the job and no one service that'll meet all needs. On 29 October 2013 08:42, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Let's be clear on definitions here; > > BT, Kingston - Top of the food chain, only financially viable for those > with lots of cash and traffic > Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many > millions of minutes per month. > SipGate, SureVoip, Gradwell, GBC - These providers are designed for > consumers, only necessary if you are not running your own equipment > NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be > more blunt, an unnecessary middleman > > If someone is running their own switch, why would you want to go with a > consumer grade service which not only limits your CLI capabilities, but > also increases complexity, affects latency and inflates call rates. > Likewise, why would you go through a middleman when you could just go > direct to the supplier with no minimum spend on outbound traffic? Though > typically inbound does require a small min spend of around ?50/month (which > is tiny in comparison with VoxBone who charge a min spend of around > ?600/month [needs clarification]), which is really the only valid reason > (that I can think of right now) to go with any of these reseller aggregates. > > Cal > > > On Tue, Oct 29, 2013 at 8:05 AM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> Magrathea are pretty much the industry standard for volume. >> >> Depending on your requirements, there are many others such as SipGate and >> Gradwell. There are also people from at least 3 UK trunk providers in this >> list, including SureVoip, NumberGroup and PowerTel. >> >> >> On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Hi Richard, >>> >>> This topic was discussed a few months back, you can read the thread here; >>> >>> >>> http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results >>> >>> http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none >>> >>> Since my last post, our chosen UK providers are Magrathea and Simwood. >>> >>> This inter-op was over a period of 6 months but based on low traffic >>> (<=500 calls a month), so others with higher throughput may wish to >>> chime in. >>> >>> Magrathea - Over a period of one month, I've not had a single >>> problematic call, routing/quality is always good, and they seem to >>> have almost every dialing code possible for DIDs. I have not dealt >>> with their technical staff yet, but their sales people were friendly >>> and knowledgeable. >>> >>> Simwood - Over a period of two months, I had 2 call routing issues to >>> international destinations and they are missing at least one number >>> range (Bournemouth). However it's worth noting that Simwood were very >>> quick to help, they identified the routing problem within 5 minutes of >>> me reporting it, and despite being a low profit customer they treated >>> us as a high priority. I've also spoken to numerous staff there, they >>> always come across as friendly and, more importantly, competent. I >>> would have no problem recommending Simwood to others. >>> >>> For international routing - I'm still in the process of reviewing >>> TelServ and Flowroute, I'd like to use Voxbone but their min spend is >>> too high for us. >>> >>> I cannot choose between Sim/Mag, they are both extremely good >>> companies - use both (redundancy is a good thing anyway!) >>> >>> Cal >>> >>> On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace >>> wrote: >>> > Hi, >>> > Can anyone recommend any SIP trunk providers, ideally based in the UK? >>> > >>> > Thanks >>> > >>> > Richard >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/afaac525/attachment.html From ben at langfeld.co.uk Tue Oct 29 13:59:57 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 29 Oct 2013 08:59:57 -0200 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: Just curious where you guys see AQL fitting in that categorisation? I've used them in the past and they seem good. I also thought they were one of the bigger providers in the UK, but since no-one has mentioned them... On 29 October 2013 07:39, Andrew Cassidy wrote: > I think the point to be made here is that your choice should be determined > by your requirements and budget. As we know neither of those, we can't give > you specific recommendations. > > The majority of people I encounter use 'consumer' grade services, with or > without their own equipment as necessary. That said these are largely small > businesses. > > In all seriousness there's always a best tool for the job and no one > service that'll meet all needs. > > > On 29 October 2013 08:42, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Let's be clear on definitions here; >> >> BT, Kingston - Top of the food chain, only financially viable for those >> with lots of cash and traffic >> Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many >> millions of minutes per month. >> SipGate, SureVoip, Gradwell, GBC - These providers are designed for >> consumers, only necessary if you are not running your own equipment >> NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be >> more blunt, an unnecessary middleman >> >> If someone is running their own switch, why would you want to go with a >> consumer grade service which not only limits your CLI capabilities, but >> also increases complexity, affects latency and inflates call rates. >> Likewise, why would you go through a middleman when you could just go >> direct to the supplier with no minimum spend on outbound traffic? Though >> typically inbound does require a small min spend of around ?50/month (which >> is tiny in comparison with VoxBone who charge a min spend of around >> ?600/month [needs clarification]), which is really the only valid reason >> (that I can think of right now) to go with any of these reseller aggregates. >> >> Cal >> >> >> On Tue, Oct 29, 2013 at 8:05 AM, Andrew Cassidy < >> andrew at cassidywebservices.co.uk> wrote: >> >>> Magrathea are pretty much the industry standard for volume. >>> >>> Depending on your requirements, there are many others such as SipGate >>> and Gradwell. There are also people from at least 3 UK trunk providers in >>> this list, including SureVoip, NumberGroup and PowerTel. >>> >>> >>> On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Hi Richard, >>>> >>>> This topic was discussed a few months back, you can read the thread >>>> here; >>>> >>>> >>>> http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results >>>> >>>> http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none >>>> >>>> Since my last post, our chosen UK providers are Magrathea and Simwood. >>>> >>>> This inter-op was over a period of 6 months but based on low traffic >>>> (<=500 calls a month), so others with higher throughput may wish to >>>> chime in. >>>> >>>> Magrathea - Over a period of one month, I've not had a single >>>> problematic call, routing/quality is always good, and they seem to >>>> have almost every dialing code possible for DIDs. I have not dealt >>>> with their technical staff yet, but their sales people were friendly >>>> and knowledgeable. >>>> >>>> Simwood - Over a period of two months, I had 2 call routing issues to >>>> international destinations and they are missing at least one number >>>> range (Bournemouth). However it's worth noting that Simwood were very >>>> quick to help, they identified the routing problem within 5 minutes of >>>> me reporting it, and despite being a low profit customer they treated >>>> us as a high priority. I've also spoken to numerous staff there, they >>>> always come across as friendly and, more importantly, competent. I >>>> would have no problem recommending Simwood to others. >>>> >>>> For international routing - I'm still in the process of reviewing >>>> TelServ and Flowroute, I'd like to use Voxbone but their min spend is >>>> too high for us. >>>> >>>> I cannot choose between Sim/Mag, they are both extremely good >>>> companies - use both (redundancy is a good thing anyway!) >>>> >>>> Cal >>>> >>>> On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace >>>> wrote: >>>> > Hi, >>>> > Can anyone recommend any SIP trunk providers, ideally based in the UK? >>>> > >>>> > Thanks >>>> > >>>> > Richard >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>> Managing Director >>> >>> >>> *T *03300 100 960 *F >>> *03300 100 961 >>> *E *andrew at cassidywebservices.co.uk >>> *W *www.cassidywebservices.co.uk >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/73a1f205/attachment-0001.html From dwiyulianto.anto at gmail.com Tue Oct 29 15:00:00 2013 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Tue, 29 Oct 2013 19:00:00 +0700 Subject: [Freeswitch-users] i've problem with TLS Message-ID: i've folowed tutorial from http://wiki.freeswitch.org/wiki/SIP_TLS but when im use phonerlite as softphone i've problem when enable TLS in phonerlite, and when i see in FS_CLI from my freeswitch server i got this error tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0xb67215c8): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) anyone know the solution from that error? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/cfe4164a/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Oct 29 15:34:52 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Oct 2013 12:34:52 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: I would advise against AQL for wholesale services, they are just another reseller aggregate but with *significantly* higher rates (26p/min to UK orange, what?!) Their technical sales team are also difficult to work with, the majority of people I spoke to there clearly did not understand the product they were selling. Avoid. Cal On Tue, Oct 29, 2013 at 10:59 AM, Ben Langfeld wrote: > Just curious where you guys see AQL fitting in that categorisation? I've > used them in the past and they seem good. I also thought they were one of > the bigger providers in the UK, but since no-one has mentioned them... > > > On 29 October 2013 07:39, Andrew Cassidy wrote: > >> I think the point to be made here is that your choice should be >> determined by your requirements and budget. As we know neither of those, we >> can't give you specific recommendations. >> >> The majority of people I encounter use 'consumer' grade services, with or >> without their own equipment as necessary. That said these are largely small >> businesses. >> >> In all seriousness there's always a best tool for the job and no one >> service that'll meet all needs. >> >> >> On 29 October 2013 08:42, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Let's be clear on definitions here; >>> >>> BT, Kingston - Top of the food chain, only financially viable for those >>> with lots of cash and traffic >>> Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many >>> millions of minutes per month. >>> SipGate, SureVoip, Gradwell, GBC - These providers are designed for >>> consumers, only necessary if you are not running your own equipment >>> NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be >>> more blunt, an unnecessary middleman >>> >>> If someone is running their own switch, why would you want to go with a >>> consumer grade service which not only limits your CLI capabilities, but >>> also increases complexity, affects latency and inflates call rates. >>> Likewise, why would you go through a middleman when you could just go >>> direct to the supplier with no minimum spend on outbound traffic? Though >>> typically inbound does require a small min spend of around ?50/month (which >>> is tiny in comparison with VoxBone who charge a min spend of around >>> ?600/month [needs clarification]), which is really the only valid reason >>> (that I can think of right now) to go with any of these reseller aggregates. >>> >>> Cal >>> >>> >>> On Tue, Oct 29, 2013 at 8:05 AM, Andrew Cassidy < >>> andrew at cassidywebservices.co.uk> wrote: >>> >>>> Magrathea are pretty much the industry standard for volume. >>>> >>>> Depending on your requirements, there are many others such as SipGate >>>> and Gradwell. There are also people from at least 3 UK trunk providers in >>>> this list, including SureVoip, NumberGroup and PowerTel. >>>> >>>> >>>> On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> Hi Richard, >>>>> >>>>> This topic was discussed a few months back, you can read the thread >>>>> here; >>>>> >>>>> >>>>> http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results >>>>> >>>>> http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none >>>>> >>>>> Since my last post, our chosen UK providers are Magrathea and Simwood. >>>>> >>>>> This inter-op was over a period of 6 months but based on low traffic >>>>> (<=500 calls a month), so others with higher throughput may wish to >>>>> chime in. >>>>> >>>>> Magrathea - Over a period of one month, I've not had a single >>>>> problematic call, routing/quality is always good, and they seem to >>>>> have almost every dialing code possible for DIDs. I have not dealt >>>>> with their technical staff yet, but their sales people were friendly >>>>> and knowledgeable. >>>>> >>>>> Simwood - Over a period of two months, I had 2 call routing issues to >>>>> international destinations and they are missing at least one number >>>>> range (Bournemouth). However it's worth noting that Simwood were very >>>>> quick to help, they identified the routing problem within 5 minutes of >>>>> me reporting it, and despite being a low profit customer they treated >>>>> us as a high priority. I've also spoken to numerous staff there, they >>>>> always come across as friendly and, more importantly, competent. I >>>>> would have no problem recommending Simwood to others. >>>>> >>>>> For international routing - I'm still in the process of reviewing >>>>> TelServ and Flowroute, I'd like to use Voxbone but their min spend is >>>>> too high for us. >>>>> >>>>> I cannot choose between Sim/Mag, they are both extremely good >>>>> companies - use both (redundancy is a good thing anyway!) >>>>> >>>>> Cal >>>>> >>>>> On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace >>>>> wrote: >>>>> > Hi, >>>>> > Can anyone recommend any SIP trunk providers, ideally based in the >>>>> UK? >>>>> > >>>>> > Thanks >>>>> > >>>>> > Richard >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> *Andrew Cassidy BSc (Hons) MBCS SSCA* >>>> Managing Director >>>> >>>> >>>> *T *03300 100 960 *F >>>> *03300 100 961 >>>> *E *andrew at cassidywebservices.co.uk >>>> *W *www.cassidywebservices.co.uk >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/413dee06/attachment-0001.html From danb.lists at gmail.com Tue Oct 29 15:50:07 2013 From: danb.lists at gmail.com (DanB) Date: Tue, 29 Oct 2013 13:50:07 +0100 Subject: [Freeswitch-users] Best practices for parallel fork with redirect on one branch In-Reply-To: References: Message-ID: <526FAEFF.9090504@gmail.com> Hey Guys, Have met a situation where I need to do parallel call forking with bridge application and on one of the legs will receive a 302 which could mean another parallel fork there. Have tested the following scenarios: 1. Sip profile with manual-redirect=true. Here I fork the call using bridge with "," as separator and parallel fork works. The issue is that as soon as one of my forked legs receives 302, the other will be cancelled also and 302 will be handled further in it's context. Is this the expected behaviour from bridge? Question here: should it drop one branch of the parallel fork if other is receiving redirect? 2. Sip profile with manual-redirect=false, leaving mod_sofia doing the handling of the redirect. After I fork the call here with bridge, only the leg which receives 302 will be redirected which is the expected behaviour by me. The issue I have here is that I do not know how to influence mod_sofia to handle parallel forking out of 302 and not very obvious out of the documentation I have digged so far (I receive 2 contacts in redirect btw which I would like to fork also here). Question here: may I specify the type of fork mod_sofia should do in some way? Thanks in advance for any tip! DanB PS: Logs available at request if needed. From alex at alexkinch.com Tue Oct 29 16:18:46 2013 From: alex at alexkinch.com (Alex Kinch) Date: Tue, 29 Oct 2013 13:18:46 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers Message-ID: I don't often post on this list (which is why I'm replying to the digest - apologies if we've moved on since then), but I just wanted to comment on a few things in this thread. @Richard - when you talk about SIP trunks, it's probably best if you clarify whether you're looking for a 'retail' solution e.g. something you can use yourself in an office environment for a PBX, FS box, etc or resell to clients that want to do the same) or wholesale - whereby you're going to be aggregating traffic from multiple customers via SIP trunks, hosted PBX, etc etc. There'll be a few differences both technically and commercially between these two offerings - e.g. most wholesale providers will only deliver inbound traffic to a SIP URI, and will more often than not expect traffic to originate from one or more static IP addresses. @Cal not sure what AQL ratecard you're looking at that charges 26ppm for UK Mobile Orange, I'd hazard a guess you need to divide that by 10? Even when you do that's still a little on the expensive side for wholesale. Wholesale providers charge minimums as they are, as the name suggests, wholesale. They are high volume low margin operations, and often don't sell 'retail' services direct to the public - e.g. Magrathea, Simwood - and will (last time I checked) require you to sign a declaration that you qualify to be a wholesale customer. Gamma, BT, Kingston etc obviously do have retail offerings but also have separate wholesale divisions. Gamma and BT (and I'd imagine Kingston too) will also provide you retail services that you can white label and sell on to your customers under your own brand - no equipment required. Companies like Gradwell, AQL (who jointly own a third party called Telephony Services, who you might have come across), SureVoip, SipGate are (in my humblest opinion) retail providers. They may also have white label offerings like the above. Now... if after all that you fit into the wholesale customer category, you might want to split your requirements into UK and international, and again split inbound (numbering) from outbound. Why? Because as (I believe) Andrew said, there's no one service that'll meet all your needs. Example: Voxbone will happily provide you numbers in umpteen countries, but you'll find that they're i) expensive for UK numbering (actually they're expensive full stop, but that's just my personal opinion), ii) there is a minimum spend, and iii) they won't help you much with your outbound requirements (except for 999/112/911 etc traffic in a few countries). Likewise, Magrathea and Simwood wouldn't be a bad choice for UK numbers, but if you want numbering in the US, Australia, etc, they're probably not going to be able to help. As for BT IP Exchange, let's leave that discussion for another day, but suffice to say it's not a regulated interconnect product - so comparing a provider who hosts their numbering on BT IPX with one that leverages TDM connectivity to BT is like comparing apples to oranges both technically and commercially. I'm not going to tag on a plug for my company to this thread as it'd be bad etiquette - but ping me a mail off-list if you want to chat further. Cheers, Alex On 29 October 2013 12:35, wrote: > > ---------- Forwarded message ---------- > From: "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> > To: FreeSWITCH Users Help > Cc: > Date: Tue, 29 Oct 2013 12:34:52 +0000 > Subject: Re: [Freeswitch-users] UK SIP Trunk providers > I would advise against AQL for wholesale services, they are just another > reseller aggregate but with *significantly* higher rates (26p/min to UK > orange, what?!) > > Their technical sales team are also difficult to work with, the majority > of people I spoke to there clearly did not understand the product they were > selling. > > Avoid. > > Cal > > > On Tue, Oct 29, 2013 at 10:59 AM, Ben Langfeld wrote: > >> Just curious where you guys see AQL fitting in that categorisation? I've >> used them in the past and they seem good. I also thought they were one of >> the bigger providers in the UK, but since no-one has mentioned them... >> >> >> On 29 October 2013 07:39, Andrew Cassidy > > wrote: >> >>> I think the point to be made here is that your choice should be >>> determined by your requirements and budget. As we know neither of those, we >>> can't give you specific recommendations. >>> >>> The majority of people I encounter use 'consumer' grade services, with >>> or without their own equipment as necessary. That said these are largely >>> small businesses. >>> >>> In all seriousness there's always a best tool for the job and no one >>> service that'll meet all needs. >>> >>> >>> On 29 October 2013 08:42, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Let's be clear on definitions here; >>>> >>>> BT, Kingston - Top of the food chain, only financially viable for those >>>> with lots of cash and traffic >>>> Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many >>>> millions of minutes per month. >>>> SipGate, SureVoip, Gradwell, GBC - These providers are designed for >>>> consumers, only necessary if you are not running your own equipment >>>> NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be >>>> more blunt, an unnecessary middleman >>>> >>>> If someone is running their own switch, why would you want to go with a >>>> consumer grade service which not only limits your CLI capabilities, but >>>> also increases complexity, affects latency and inflates call rates. >>>> Likewise, why would you go through a middleman when you could just go >>>> direct to the supplier with no minimum spend on outbound traffic? Though >>>> typically inbound does require a small min spend of around ?50/month (which >>>> is tiny in comparison with VoxBone who charge a min spend of around >>>> ?600/month [needs clarification]), which is really the only valid reason >>>> (that I can think of right now) to go with any of these reseller aggregates. >>>> >>>> Cal >>>> >>>> >>>> On Tue, Oct 29, 2013 at 8:05 AM, Andrew Cassidy < >>>> andrew at cassidywebservices.co.uk> wrote: >>>> >>>>> Magrathea are pretty much the industry standard for volume. >>>>> >>>>> Depending on your requirements, there are many others such as SipGate >>>>> and Gradwell. There are also people from at least 3 UK trunk providers in >>>>> this list, including SureVoip, NumberGroup and PowerTel. >>>>> >>>>> >>>>> On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] < >>>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>>> >>>>>> Hi Richard, >>>>>> >>>>>> This topic was discussed a few months back, you can read the thread >>>>>> here; >>>>>> >>>>>> >>>>>> http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results >>>>>> >>>>>> http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none >>>>>> >>>>>> Since my last post, our chosen UK providers are Magrathea and Simwood. >>>>>> >>>>>> This inter-op was over a period of 6 months but based on low traffic >>>>>> (<=500 calls a month), so others with higher throughput may wish to >>>>>> chime in. >>>>>> >>>>>> Magrathea - Over a period of one month, I've not had a single >>>>>> problematic call, routing/quality is always good, and they seem to >>>>>> have almost every dialing code possible for DIDs. I have not dealt >>>>>> with their technical staff yet, but their sales people were friendly >>>>>> and knowledgeable. >>>>>> >>>>>> Simwood - Over a period of two months, I had 2 call routing issues to >>>>>> international destinations and they are missing at least one number >>>>>> range (Bournemouth). However it's worth noting that Simwood were very >>>>>> quick to help, they identified the routing problem within 5 minutes of >>>>>> me reporting it, and despite being a low profit customer they treated >>>>>> us as a high priority. I've also spoken to numerous staff there, they >>>>>> always come across as friendly and, more importantly, competent. I >>>>>> would have no problem recommending Simwood to others. >>>>>> >>>>>> For international routing - I'm still in the process of reviewing >>>>>> TelServ and Flowroute, I'd like to use Voxbone but their min spend is >>>>>> too high for us. >>>>>> >>>>>> I cannot choose between Sim/Mag, they are both extremely good >>>>>> companies - use both (redundancy is a good thing anyway!) >>>>>> >>>>>> Cal >>>>>> >>>>>> On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace < >>>>>> richard.mace at gmail.com> wrote: >>>>>> > Hi, >>>>>> > Can anyone recommend any SIP trunk providers, ideally based in the >>>>>> UK? >>>>>> > >>>>>> > Thanks >>>>>> > >>>>>> > Richard >>>>>> > >>>>>> >>>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/79d325f7/attachment.html From gavin.henry at gmail.com Tue Oct 29 17:09:49 2013 From: gavin.henry at gmail.com (Gavin Henry) Date: Tue, 29 Oct 2013 14:09:49 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: Hi Cal, Gavin here. I'm the MD of SureVoIP and would like to correct some of below: > Let's be clear on definitions here; > > BT, Kingston - Top of the food chain, only financially viable for those with lots of cash and traffic Correct. The UK Telecom Incumbents. But don't forgot the each have "Retail", "Business" and "Wholesale" divisions. > Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many millions of minutes per month. Correct. They each have their own SS7 platforms connected in the UK PSTN, SIP platforms and are IP network operators (have their own AS number, members of RIPE, members of LINX, Ofcom number allocations and other Internet Exchange Points). Same applies to the companies above. > SipGate, SureVoip, Gradwell, GBC - These providers are designed for consumers, only necessary if you are not running your own equipment These are not just for consumers. They may offer retail but are mainly for business, as SureVoIP is. When talking about consumers I would talk about Vonage etc. SureVoIP (us) have our own SIP platforms (FreeSWITCH, OpenSIPS etc.) and are IP network operators (AS199659) members of RIPE, members of LINX, Ofcom number allocation holders: https://www.linx.net/about/memberlist/letter/S We don't have hardware based SIP platforms like the level above us do (Magrathea, Simwood, Gamma) as we don't do as many minutes as them and aren't wholesale. In fact, the first two are two of our carriers as well as BT Wholesale. You can only get accounts with them if you are selling onwards to business or retail, so bear that in mind. > NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be more blunt, an unnecessary middleman Can't comment. Look at what I've described above to make your own mind up. > If someone is running their own switch, why would you want to go with a consumer grade service which not only limits your CLI capabilities, but also increases complexity, affects latency and inflates call rates. Likewise, why would you go through a middleman when you could just go direct to the supplier with no minimum spend on outbound traffic? Though typically inbound does require a small min spend of around ?50/month (which is tiny in comparison with VoxBone who charge a min spend of around ?600/month [needs clarification]), which is really the only valid reason (that I can think of right now) to go with any of these reseller aggregates. You'd do this because you are not selling on the minutes. If you are a business user you can't sign up to the wholesale guys. That's the point of wholesale. Make up your own mind and look for an API :-) Thanks, Gavin. -- http://www.surevoip.co.uk From cal.leeming at simplicitymedialtd.co.uk Tue Oct 29 18:02:58 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Oct 2013 15:02:58 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers In-Reply-To: References: Message-ID: On Tue, Oct 29, 2013 at 1:18 PM, Alex Kinch wrote: > I don't often post on this list (which is why I'm replying to the digest - > apologies if we've moved on since then), but I just wanted to comment on a > few things in this thread. > > @Richard - when you talk about SIP trunks, it's probably best if you clarify > whether you're looking for a 'retail' solution e.g. something you can use > yourself in an office environment for a PBX, FS box, etc or resell to > clients that want to do the same) or wholesale - whereby you're going to be > aggregating traffic from multiple customers via SIP trunks, hosted PBX, etc > etc. There'll be a few differences both technically and commercially between > these two offerings - e.g. most wholesale providers will only deliver > inbound traffic to a SIP URI, and will more often than not expect traffic to > originate from one or more static IP addresses. > > @Cal not sure what AQL ratecard you're looking at that charges 26ppm for UK > Mobile Orange, I'd hazard a guess you need to divide that by 10? Even when > you do that's still a little on the expensive side for wholesale. Take a look at http://www.aql.com/telecoms/site/aql_tariff.pdf Page 257, peak is ?0.128, off peak is ?0.269. I asked them about this before and they said "the majority of our customers make calls during the day, so we charge more during the evenings". I lol'd hard. > > Wholesale providers charge minimums as they are, as the name suggests, > wholesale. They are high volume low margin operations, and often don't sell > 'retail' services direct to the public - e.g. Magrathea, Simwood - and will > (last time I checked) require you to sign a declaration that you qualify to > be a wholesale customer. Correct. In hindsight I should have perhaps clarified with OP on whether his usage was going to be commercial or wholesale. > > Gamma, BT, Kingston etc obviously do have retail offerings but also have > separate wholesale divisions. Gamma and BT (and I'd imagine Kingston too) > will also provide you retail services that you can white label and sell on > to your customers under your own brand - no equipment required. > > Companies like Gradwell, AQL (who jointly own a third party called > Telephony Services, who you might have come across), SureVoip, SipGate are > (in my humblest opinion) retail providers. They may also have white label > offerings like the above. > > Now... if after all that you fit into the wholesale customer category, you > might want to split your requirements into UK and international, and again > split inbound (numbering) from outbound. Why? Because as (I believe) Andrew > said, there's no one service that'll meet all your needs. Agreed > Example: Voxbone will happily provide you numbers in umpteen countries, but > you'll find that they're i) expensive for UK numbering (actually they're > expensive full stop, but that's just my personal opinion), ii) there is a > minimum spend, and iii) they won't help you much with your outbound > requirements (except for 999/112/911 etc traffic in a few countries). > Likewise, Magrathea and Simwood wouldn't be a bad choice for UK numbers, but > if you want numbering in the US, Australia, etc, they're probably not going > to be able to help. Agreed > > As for BT IP Exchange, let's leave that discussion for another day, but > suffice to say it's not a regulated interconnect product - so comparing a > provider who hosts their numbering on BT IPX with one that leverages TDM > connectivity to BT is like comparing apples to oranges both technically and > commercially. > > I'm not going to tag on a plug for my company to this thread as it'd be bad > etiquette - but ping me a mail off-list if you want to chat further. > > Cheers, > Alex > > > > > > > > On 29 October 2013 12:35, > wrote: >> >> ---------- Forwarded message ---------- >> From: "Cal Leeming [Simplicity Media Ltd]" >> >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 29 Oct 2013 12:34:52 +0000 >> Subject: Re: [Freeswitch-users] UK SIP Trunk providers >> I would advise against AQL for wholesale services, they are just another >> reseller aggregate but with *significantly* higher rates (26p/min to UK >> orange, what?!) >> >> Their technical sales team are also difficult to work with, the majority >> of people I spoke to there clearly did not understand the product they were >> selling. >> >> Avoid. >> >> Cal >> >> >> On Tue, Oct 29, 2013 at 10:59 AM, Ben Langfeld wrote: >>> >>> Just curious where you guys see AQL fitting in that categorisation? I've >>> used them in the past and they seem good. I also thought they were one of >>> the bigger providers in the UK, but since no-one has mentioned them... >>> >>> >>> On 29 October 2013 07:39, Andrew Cassidy >>> wrote: >>>> >>>> I think the point to be made here is that your choice should be >>>> determined by your requirements and budget. As we know neither of those, we >>>> can't give you specific recommendations. >>>> >>>> The majority of people I encounter use 'consumer' grade services, with >>>> or without their own equipment as necessary. That said these are largely >>>> small businesses. >>>> >>>> In all seriousness there's always a best tool for the job and no one >>>> service that'll meet all needs. >>>> >>>> >>>> On 29 October 2013 08:42, Cal Leeming [Simplicity Media Ltd] >>>> wrote: >>>>> >>>>> Let's be clear on definitions here; >>>>> >>>>> BT, Kingston - Top of the food chain, only financially viable for those >>>>> with lots of cash and traffic >>>>> Magrathea, Simwood, Gamma - Designed for wholesale, scalable to many >>>>> millions of minutes per month. >>>>> SipGate, SureVoip, Gradwell, GBC - These providers are designed for >>>>> consumers, only necessary if you are not running your own equipment >>>>> NumberGroup, Sol4 - they are essentially a reseller aggregate, or to be >>>>> more blunt, an unnecessary middleman >>>>> >>>>> If someone is running their own switch, why would you want to go with a >>>>> consumer grade service which not only limits your CLI capabilities, but also >>>>> increases complexity, affects latency and inflates call rates. Likewise, why >>>>> would you go through a middleman when you could just go direct to the >>>>> supplier with no minimum spend on outbound traffic? Though typically inbound >>>>> does require a small min spend of around ?50/month (which is tiny in >>>>> comparison with VoxBone who charge a min spend of around ?600/month [needs >>>>> clarification]), which is really the only valid reason (that I can think of >>>>> right now) to go with any of these reseller aggregates. >>>>> >>>>> Cal >>>>> >>>>> >>>>> On Tue, Oct 29, 2013 at 8:05 AM, Andrew Cassidy >>>>> wrote: >>>>>> >>>>>> Magrathea are pretty much the industry standard for volume. >>>>>> >>>>>> Depending on your requirements, there are many others such as SipGate >>>>>> and Gradwell. There are also people from at least 3 UK trunk providers in >>>>>> this list, including SureVoip, NumberGroup and PowerTel. >>>>>> >>>>>> >>>>>> On 29 October 2013 00:29, Cal Leeming [Simplicity Media Ltd] >>>>>> wrote: >>>>>>> >>>>>>> Hi Richard, >>>>>>> >>>>>>> This topic was discussed a few months back, you can read the thread >>>>>>> here; >>>>>>> >>>>>>> >>>>>>> http://markmail.org/message/xdjhmpkzd7txndji#query:+page:1+mid:ncuirf776l5hjovc+state:results >>>>>>> >>>>>>> http://freeswitch-users.2379917.n2.nabble.com/DID-providers-any-thoughts-jan-2013-tt7586851.html#none >>>>>>> >>>>>>> Since my last post, our chosen UK providers are Magrathea and >>>>>>> Simwood. >>>>>>> >>>>>>> This inter-op was over a period of 6 months but based on low traffic >>>>>>> (<=500 calls a month), so others with higher throughput may wish to >>>>>>> chime in. >>>>>>> >>>>>>> Magrathea - Over a period of one month, I've not had a single >>>>>>> problematic call, routing/quality is always good, and they seem to >>>>>>> have almost every dialing code possible for DIDs. I have not dealt >>>>>>> with their technical staff yet, but their sales people were friendly >>>>>>> and knowledgeable. >>>>>>> >>>>>>> Simwood - Over a period of two months, I had 2 call routing issues to >>>>>>> international destinations and they are missing at least one number >>>>>>> range (Bournemouth). However it's worth noting that Simwood were very >>>>>>> quick to help, they identified the routing problem within 5 minutes >>>>>>> of >>>>>>> me reporting it, and despite being a low profit customer they treated >>>>>>> us as a high priority. I've also spoken to numerous staff there, they >>>>>>> always come across as friendly and, more importantly, competent. I >>>>>>> would have no problem recommending Simwood to others. >>>>>>> >>>>>>> For international routing - I'm still in the process of reviewing >>>>>>> TelServ and Flowroute, I'd like to use Voxbone but their min spend is >>>>>>> too high for us. >>>>>>> >>>>>>> I cannot choose between Sim/Mag, they are both extremely good >>>>>>> companies - use both (redundancy is a good thing anyway!) >>>>>>> >>>>>>> Cal >>>>>>> >>>>>>> On Mon, Oct 28, 2013 at 10:34 PM, Richard Mace >>>>>>> wrote: >>>>>>> > Hi, >>>>>>> > Can anyone recommend any SIP trunk providers, ideally based in the >>>>>>> > UK? >>>>>>> > >>>>>>> > Thanks >>>>>>> > >>>>>>> > Richard >>>>>>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jeremyc at ssimicro.com Tue Oct 29 18:54:03 2013 From: jeremyc at ssimicro.com (Jeremy Childs) Date: Tue, 29 Oct 2013 09:54:03 -0600 Subject: [Freeswitch-users] UUID from Extension Number Message-ID: <526FDA1B.2060304@ssimicro.com> I'm trying to create a "works-anywhere" eavesdrop extension. I've got the demo application that uses the hash/spymap functionality working correctly, but you would need the caller-id of the incoming call in order to eavesdrop on incoming calls. This could be fixed by capturing the destination number instead of the caller-id number for incoming calls (and I've done this, it works fine). However, this breaks in the case where an attended transfer has happened (the UUID of the call that gets connected "belonged" to the receptionist). The most robust thing I can think of is to do a live lookup of the call UUID using the "final" extension number. Is there any way to do this lookup from the dialplan? To be clear, what I'd like to do is this: Incoming call is picked up by receptionist (ext 100). Receptionist transfers call to extension 101. The Boss (extension 200), dials *88101 which looks up the UUID of extension 101's current call, provides this UUID to the eavesdrop extension, and the eavesdrop proceeds. There doesn't appear to be any way to do this from the dialplan, however -- am I missing something? From alipey at gmail.com Tue Oct 29 19:17:57 2013 From: alipey at gmail.com (Ali Pey) Date: Tue, 29 Oct 2013 12:17:57 -0400 Subject: [Freeswitch-users] perlmod ESL RPM Message-ID: Hello, Is it possible to install perlmod for ESL using an RPM package? i.e. is there an RPM package that has perlmod for ESL? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/35603964/attachment.html From mario_fs at mgtech.com Tue Oct 29 21:40:25 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 29 Oct 2013 11:40:25 -0700 Subject: [Freeswitch-users] ERR] switch_rtp.c:2968 Error starting timer [soft], async RTP disabled Message-ID: <94F3DA7B-744C-49A4-99E6-665F40344E03@mgtech.com> Can anyone point me in the right direction on what could cause this message that occurs when paging using G722 and only when using G722? Traces and debug logs have no additional information than the normal log. ERR] switch_rtp.c:2968 Error starting timer [soft], async RTP disabled Mario G From anthony.minessale at gmail.com Tue Oct 29 22:36:11 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Oct 2013 14:36:11 -0500 Subject: [Freeswitch-users] ERR] switch_rtp.c:2968 Error starting timer [soft], async RTP disabled In-Reply-To: <94F3DA7B-744C-49A4-99E6-665F40344E03@mgtech.com> References: <94F3DA7B-744C-49A4-99E6-665F40344E03@mgtech.com> Message-ID: Update to latest to get more detailed error and post the trace with sip trace and debug to JIRA .... Its possibly a mis-negotiation of g722 due to some legacy interop issues with some devices. sofia global siptrace on sofia tracelevel alert console loglevel debug On Tue, Oct 29, 2013 at 1:40 PM, Mario G wrote: > Can anyone point me in the right direction on what could cause this > message that occurs when paging using G722 and only when using G722? Traces > and debug logs have no additional information than the normal log. > ERR] switch_rtp.c:2968 Error starting timer [soft], async RTP disabled > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/eff7de21/attachment.html From spencer at 5ninesolutions.com Tue Oct 29 22:47:17 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 29 Oct 2013 12:47:17 -0700 Subject: [Freeswitch-users] Prevent parking on top of another call Message-ID: <527010C5.10708@5ninesolutions.com> Hello, Does anyone have any suggestions as to prevent a blind transfer from parking on top of another call when using valet parking? Thanks! Spencer From william.king at quentustech.com Tue Oct 29 22:48:11 2013 From: william.king at quentustech.com (William King) Date: Tue, 29 Oct 2013 12:48:11 -0700 Subject: [Freeswitch-users] gateways testing DNS SRV records In-Reply-To: <526978FD.6060705@xtronics.com> References: <5269553E.2030806@xtronics.com> <52696BBF.5090800@freeswitch.org> <526978FD.6060705@xtronics.com> Message-ID: <527010FB.3090104@quentustech.com> At least some DNS servers will provide some of the additional record info in the additional section of the dns response. I have seen dns servers that send additional records, but not the top priority ones, which does mean that the client would need to do additional lookups. For instance in the sample case following, the 3 last SRV records include the A record lookups, but my sip clients do requests for the first ones: quentusrex at quentusrex-thinkpad:~$ dig _sip._udp.callcentric.com SRV ; <<>> DiG 9.8.4-rpz2+rl005.12-P1 <<>> _sip._udp.callcentric.com SRV ;; global options: +cmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 28623 ;; flags: qr rd ra; QUERY: 1, ANSWER: 8, AUTHORITY: 3, ADDITIONAL: 3 ;; QUESTION SECTION: ;_sip._udp.callcentric.com. IN SRV ;; ANSWER SECTION: _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha15.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha16.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha17.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha18.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha19.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha11.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha12.callcentric.com. _sip._udp.callcentric.com. 600 IN SRV 20 0 5080 alpha13.callcentric.com. ;; AUTHORITY SECTION: callcentric.com. 1200 IN NS ns1.telengy.net. callcentric.com. 1200 IN NS ns3.telengy.net. callcentric.com. 1200 IN NS ns4.telengy.net. ;; ADDITIONAL SECTION: alpha11.callcentric.com. 1200 IN A 204.11.192.159 alpha12.callcentric.com. 1200 IN A 204.11.192.160 alpha13.callcentric.com. 1200 IN A 204.11.192.161 ;; Query time: 277 msec ;; SERVER: 192.168.100.4#53(192.168.100.4) ;; WHEN: Tue Oct 29 12:45:07 2013 ;; MSG SIZE rcvd: 500 William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 10/24/2013 12:46 PM, Karl Schmidt wrote: > On 10/24/2013 01:49 PM, Raymond Chandler wrote: >> Yes, FreeSWITCH supports SRV records, and handles them correctly. When >> I use sofia_dig, I see the IP and the port that are returned from DNS >> lookups via dig. If you're not, then you or your provider must have >> something misconfigured. > > This is correct - I was able to see the details by setting sofia loglevel all 9 > > ( I must say that the SRV record syntax is poorly designed - and shouldn't have required multiple > lookups - needlessly complex. ) > > >> >> On 10/24/2013 01:13 PM, Karl Schmidt wrote: >>> I'm trying to figure out a way to see if a freeswitch gateway respects SRV records. >>> >>> I've been trying different sofia commands without finding the IP that it is using. >>> >>> >>> sofia_dig returns just the A records. >>> >>> >>> To be clear - let's say you are using example.com as your DID and have a gateway set up. >>> >>> The provider should have SRV records so that there can be graceful failures. >>> >>> To see the IPs to use - do >>> >>> dig SRV _sip._upd.example.com >>> >>> >>> So my precise question is how to see the IP freeswitch is using for a gateway. (I'm assuming that >>> freeswitch knows how to use SRV records?) >>> > > > -------------------------------------------------------------------------------- > Karl Schmidt EMail Karl at xtronics.com > Transtronics, Inc. WEB http://secure.transtronics.com > 3209 West 9th Street Ph (785) 841-3089 > Lawrence, KS 66049 FAX (785) 841-0434 > > Taking subsidy money from the government is immoral. > Realize that you are taking the money of people > that work at McDonald?s by threat of force. > kps > > -------------------------------------------------------------------------------- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alex at alexkinch.com Tue Oct 29 23:39:59 2013 From: alex at alexkinch.com (Alex Kinch) Date: Tue, 29 Oct 2013 20:39:59 +0000 Subject: [Freeswitch-users] UK SIP Trunk providers Message-ID: Hi Cal > @Cal not sure what AQL ratecard you're looking at that charges 26ppm for > UK > > Mobile Orange, I'd hazard a guess you need to divide that by 10? Even > when > > you do that's still a little on the expensive side for wholesale. > > Take a look at http://www.aql.com/telecoms/site/aql_tariff.pdf > Page 257, peak is ?0.128, off peak is ?0.269. > > I asked them about this before and they said "the majority of our > customers make calls during the day, so we charge more during the > evenings". > > I lol'd hard. > That has to be an old rate card from not a million years ago when off-peak was more expensive than peak for a while. Even now, that peak rate is quite expensive for retail. Then again, they've probably got customers paying it, so that's a nice healthy margin :) As for the rest of my points, wasn't aimed at you personally Cal - more to the thread in general. The point still stands though that there's no wrong or right solution, it all depends on the requirements. Cheers, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/2863a644/attachment.html From eidevm5 at gmail.com Wed Oct 30 01:22:09 2013 From: eidevm5 at gmail.com (Peter) Date: Wed, 30 Oct 2013 09:22:09 +1100 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: Have you imported the CA cert into your phone? On Tue, Oct 29, 2013 at 11:00 PM, dwi yulianto wrote: > i've folowed tutorial from http://wiki.freeswitch.org/wiki/SIP_TLS > > but when im use phonerlite as softphone i've problem when enable TLS in > phonerlite, > > and when i see in FS_CLI from my freeswitch server i got this error > > tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING > tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING > tport_tls.c:958 tls_connect() tls_connect(0xb67215c8): TLS setup failed > (error:00000001:lib(0):func(0):reason(1)) > > anyone know the solution from that error? > > thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/64596455/attachment.html From krice at freeswitch.org Wed Oct 30 01:33:36 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 29 Oct 2013 17:33:36 -0500 Subject: [Freeswitch-users] ClueCon Weekly News and Notes Message-ID: Hey Guys, We had a great ClueCon Weekly Last week with Cal Leeming. This week we will be following up on last weeks Information with new information! Join us Wed at 1PM Eastern, 10AM Pacific via sip:888 at conference.freeswitch.org or for other access see http://fs0.us/call888 And for all your social networking needs don?t forget to join our circles on G+, our Group on FaceBook, and Follow us on Twitter! G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE Have a Great Week! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131029/f57192c7/attachment.html From dwiyulianto.anto at gmail.com Wed Oct 30 03:39:39 2013 From: dwiyulianto.anto at gmail.com (dwi yulianto) Date: Wed, 30 Oct 2013 07:39:39 +0700 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: thanks peter thats solved. i'm using linphone in windows, and i replace roorca.pem from linphone with cacert.pem from freeswitch server, and finaly thats work. :) On Wed, Oct 30, 2013 at 5:22 AM, Peter wrote: > Have you imported the CA cert into your phone? > > > On Tue, Oct 29, 2013 at 11:00 PM, dwi yulianto > wrote: > >> i've folowed tutorial from http://wiki.freeswitch.org/wiki/SIP_TLS >> >> but when im use phonerlite as softphone i've problem when enable TLS in >> phonerlite, >> >> and when i see in FS_CLI from my freeswitch server i got this error >> >> tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING >> tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING >> tport_tls.c:958 tls_connect() tls_connect(0xb67215c8): TLS setup failed >> (error:00000001:lib(0):func(0):reason(1)) >> >> anyone know the solution from that error? >> >> thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/6698bb04/attachment.html From brian.wiese.freeswitch at gmail.com Wed Oct 30 05:49:26 2013 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Tue, 29 Oct 2013 21:49:26 -0500 Subject: [Freeswitch-users] Prevent parking on top of another call In-Reply-To: <527010C5.10708@5ninesolutions.com> References: <527010C5.10708@5ninesolutions.com> Message-ID: Spencer: The following dialplan is what I'm using for valet park. There are four extensions as follows: 1. park-in, 50100 - FreeSWITCH automatically assigns an available parking slot number when a call is transferred to this extension. 2. park-in-directed, 501XX - Allows the user to transfer a call to specific slot number 1XX (101 to 199). If a blind transfer is attempted to a slot that is already occupied, the call recalls to the extension performing the park; if an attended transfer, the user is prompted to enter another slot number. 3. park-recall, ParkRecall10XXX - If the caller is parked for more than two minutes they are transferred to this extension and sent back to the extension that parked the call. 4. park-out, 511XX - Retrieve the call parked in slot 1XX. This dialplan assumes extensions are five digits long and start with "10". Extension 10900 is the attendant. I've also posted this to the FreeSWITCH pastebin in case your mail client eats the XML: http://pastebin.freeswitch.org/21586 =================== =================== Hope this helps! ~Brian On Tue, Oct 29, 2013 at 2:47 PM, Spencer Thomason wrote: > > Hello, > Does anyone have any suggestions as to prevent a blind transfer from > parking on top of another call when using valet parking? > > Thanks! > Spencer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From eidevm5 at gmail.com Wed Oct 30 06:15:27 2013 From: eidevm5 at gmail.com (Peter) Date: Wed, 30 Oct 2013 14:15:27 +1100 Subject: [Freeswitch-users] i've problem with TLS In-Reply-To: References: Message-ID: Run into that problem before. Linphone uses its own private root store and ignores the system wide root store, which of course means you either need to compile Linphone with your own CA cert, or override it on a rooted phone. On Wed, Oct 30, 2013 at 11:39 AM, dwi yulianto wrote: > thanks peter thats solved. > > i'm using linphone in windows, and i replace roorca.pem from linphone with > cacert.pem from freeswitch server, and finaly thats work. :) > > > On Wed, Oct 30, 2013 at 5:22 AM, Peter wrote: > >> Have you imported the CA cert into your phone? >> >> >> On Tue, Oct 29, 2013 at 11:00 PM, dwi yulianto < >> dwiyulianto.anto at gmail.com> wrote: >> >>> i've folowed tutorial from http://wiki.freeswitch.org/wiki/SIP_TLS >>> >>> but when im use phonerlite as softphone i've problem when enable TLS in >>> phonerlite, >>> >>> and when i see in FS_CLI from my freeswitch server i got this error >>> >>> tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING >>> tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING >>> tport_tls.c:958 tls_connect() tls_connect(0xb67215c8): TLS setup failed >>> (error:00000001:lib(0):func(0):reason(1)) >>> >>> anyone know the solution from that error? >>> >>> thanks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/b5ebad24/attachment.html From steveayre at gmail.com Wed Oct 30 11:01:11 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 30 Oct 2013 08:01:11 +0000 Subject: [Freeswitch-users] UUID from Extension Number In-Reply-To: <526FDA1B.2060304@ssimicro.com> References: <526FDA1B.2060304@ssimicro.com> Message-ID: Use mod_db to store the current call's uuid and then again to look it up. On 29 October 2013 15:54, Jeremy Childs wrote: > I'm trying to create a "works-anywhere" eavesdrop extension. I've got > the demo application that uses the hash/spymap functionality working > correctly, but you would need the caller-id of the incoming call in > order to eavesdrop on incoming calls. This could be fixed by capturing > the destination number instead of the caller-id number for incoming > calls (and I've done this, it works fine). > > However, this breaks in the case where an attended transfer has happened > (the UUID of the call that gets connected "belonged" to the receptionist). > > The most robust thing I can think of is to do a live lookup of the call > UUID using the "final" extension number. Is there any way to do this > lookup from the dialplan? > > To be clear, what I'd like to do is this: > > Incoming call is picked up by receptionist (ext 100). Receptionist > transfers call to extension 101. The Boss (extension 200), dials *88101 > which looks up the UUID of extension 101's current call, provides this > UUID to the eavesdrop extension, and the eavesdrop proceeds. > > There doesn't appear to be any way to do this from the dialplan, however > -- am I missing something? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/79431e82/attachment.html From khorsmann at gmail.com Wed Oct 30 13:21:24 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 30 Oct 2013 11:21:24 +0100 Subject: [Freeswitch-users] FS 1.2.x mod_cdr_csv change creation mode Message-ID: Hello, the default creation mode of cdr csv files are 0600. Is there an configuration parameter to change that creation mode to 0640? -- Kind Regards *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/9e61da63/attachment.html From gmangudai at gmail.com Wed Oct 30 13:57:52 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Wed, 30 Oct 2013 18:57:52 +0800 Subject: [Freeswitch-users] IVR not available with iLBC Message-ID: i have configured FS to use iLBC for audio CODEC, it works perfect for any ordinary calls, but when making calls to IVR, the calls hangs up immediately after being established, the log on FS consloe is: [WARNING] mod_sofia.c:1279 Asynchronous PTIME not supported, changing our end from 20 to 30 [WARNING] switch_core_codec.c:704 Codec iLBC Exists but not at the desired implementation. 8000hz 30ms [ERR] sofia_glue.c:3062 Can't load codec? is it the problem that iLBC could not decode and play IVR sound files? so should i convert the .wav IVR sound files to iLBC format, or is there any better solutions since i may have to maintain two sets of sound files, one for iLBC and the other for other codecs as PCMU? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/5a429e5b/attachment.html From 4orbit at gmail.com Wed Oct 30 14:16:43 2013 From: 4orbit at gmail.com (Sergey Zhuravlov) Date: Wed, 30 Oct 2013 15:16:43 +0400 Subject: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle Message-ID: I am glad to tell you that: 1) mod_gsmopen and e1550 work on Raspberry Pi (incoming, outgoing SMS in both directions). 2) and the stable version FS and the latest FS compile on Raspberry Pi, but it's a long time. 3) The latest version does not run on my raspberries. Need advice on optimizing the FS for raspberry Pi, as low-productivity device. Can any startup options? Maybe someone knows what modules you can disable resource-intensive. Something else? Does somebody have a positive experience with Raspberry Pi, mod_gsmopen > GSM dongle ??? > I use Raspberry Pi B and e1550. > > It's a shame that with an asterisk all works fine -- RasPBX. http://www. > raspberry-asterisk.org/ > > But I prefer the FreeSWITCH ;-) > On the basis of experience, I thought that PS will work best for this > modest hardware faster and require fewer resources. > > But it turns out it is not. Asterisk with a web server, MySQL database, > and other, less load than FS. > In this connection, other parameters may be used starting FS to save > resources? > > And the main question! Silence an incoming call on 5000 (IVR) as well as an > outgoing in gsm network. Those are not sound. > > -- > WBR, Sergey > > GTALK/JABBER:4orbit at gmail.com > > -- WBR, Sergey GTALK/JABBER:4orbit at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/281c50bf/attachment-0001.html From smrdoshi at gmail.com Wed Oct 30 16:27:11 2013 From: smrdoshi at gmail.com (Samir Doshi) Date: Wed, 30 Oct 2013 06:27:11 -0700 (PDT) Subject: [Freeswitch-users] Code negotiation based on local priority Message-ID: <1383139631286-7595975.post@n2.nabble.com> Hi Guys, I have question about code negotiation priority. Here is the case, In fs server we have set local codec priority g729, g711, g723, but client will send us call in g711, g723, g729 priority. so here fs will negotiation g711 as that's client codec priority. But can i be able to handle priority based on my local fs server? ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Code-negotiation-based-on-local-priority-tp7595975.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cmrienzo at gmail.com Wed Oct 30 17:19:58 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 30 Oct 2013 10:19:58 -0400 Subject: [Freeswitch-users] Code negotiation based on local priority In-Reply-To: <1383139631286-7595975.post@n2.nabble.com> References: <1383139631286-7595975.post@n2.nabble.com> Message-ID: Set this param in your SIP profile. On Wed, Oct 30, 2013 at 9:27 AM, Samir Doshi wrote: > Hi Guys, > > I have question about code negotiation priority. > > Here is the case, > In fs server we have set local codec priority g729, g711, g723, > but client will send us call in g711, g723, g729 priority. > > so here fs will negotiation g711 as that's client codec priority. But can i > be able to handle priority based on my local fs server? > > > > > > > > ----- > Thanks, > Samir > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Code-negotiation-based-on-local-priority-tp7595975.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/3f685ae7/attachment.html From jeremyc at ssimicro.com Wed Oct 30 18:12:37 2013 From: jeremyc at ssimicro.com (Jeremy Childs) Date: Wed, 30 Oct 2013 09:12:37 -0600 Subject: [Freeswitch-users] UUID from Extension Number In-Reply-To: References: <526FDA1B.2060304@ssimicro.com> Message-ID: <527121E5.60409@ssimicro.com> Yes, that works fine, except in the case where the incoming call bounces off a receptionist. Receptionist takes call - UUID 1234 is mapped (via mod_db) to the receptionist's extension number. Receptionist begins an attended transfer - new UUID is generated (6789) - this UUID is mapped to the tranfer target (extension 101). Receptionist completes transfer - new UUID (6789) is discarded and UUID 1234 is now the UUID of the call. Eavesdropper wants to listen to whatever extension 101 is doing... no mapping for this. I have a patch for mod_dptools ready to submit that allows eavesdrop to lookup an extension number, I'm just checking to see if I've missed some other way to do it. On 10/30/13, 2:01 AM, Steven Ayre wrote: > Use mod_db to store the current call's uuid and then again to look it up. > > > On 29 October 2013 15:54, Jeremy Childs > wrote: > > I'm trying to create a "works-anywhere" eavesdrop extension. I've got > the demo application that uses the hash/spymap functionality working > correctly, but you would need the caller-id of the incoming > call in > order to eavesdrop on incoming calls. This could be fixed by capturing > the destination number instead of the caller-id number for incoming > calls (and I've done this, it works fine). > > However, this breaks in the case where an attended transfer has > happened > (the UUID of the call that gets connected "belonged" to the > receptionist). > > The most robust thing I can think of is to do a live lookup of the > call > UUID using the "final" extension number. Is there any way to do this > lookup from the dialplan? > > To be clear, what I'd like to do is this: > > Incoming call is picked up by receptionist (ext 100). Receptionist > transfers call to extension 101. The Boss (extension 200), dials > *88101 > which looks up the UUID of extension 101's current call, provides this > UUID to the eavesdrop extension, and the eavesdrop proceeds. > > There doesn't appear to be any way to do this from the dialplan, > however > -- am I missing something? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/bb1206a0/attachment.html From michel.brabants at gmail.com Wed Oct 30 19:58:29 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 30 Oct 2013 17:58:29 +0100 Subject: [Freeswitch-users] UUID from Extension Number In-Reply-To: <527121E5.60409@ssimicro.com> References: <526FDA1B.2060304@ssimicro.com> <527121E5.60409@ssimicro.com> Message-ID: I've written a patch for freeswitch that allows you to execute a dialplan-command/api from the dialplan when the transfer succeeds ... It has been lying around for some time in my local branch waiting to be uploaded as a jira-patch ... I think that this would solve you problem. Michel On Wed, Oct 30, 2013 at 4:12 PM, Jeremy Childs wrote: > Yes, that works fine, except in the case where the incoming call bounces > off a receptionist. > > Receptionist takes call - UUID 1234 is mapped (via mod_db) to the > receptionist's extension number. > Receptionist begins an attended transfer - new UUID is generated (6789) - > this UUID is mapped to the tranfer target (extension 101). > Receptionist completes transfer - new UUID (6789) is discarded and UUID > 1234 is now the UUID of the call. > Eavesdropper wants to listen to whatever extension 101 is doing... no > mapping for this. > > I have a patch for mod_dptools ready to submit that allows eavesdrop to > lookup an extension number, I'm just checking to see if I've missed some > other way to do it. > > > > On 10/30/13, 2:01 AM, Steven Ayre wrote: > > Use mod_db to store the current call's uuid and then again to look it up. > > > On 29 October 2013 15:54, Jeremy Childs wrote: > >> I'm trying to create a "works-anywhere" eavesdrop extension. I've got >> the demo application that uses the hash/spymap functionality working >> correctly, but you would need the caller-id of the incoming call in >> order to eavesdrop on incoming calls. This could be fixed by capturing >> the destination number instead of the caller-id number for incoming >> calls (and I've done this, it works fine). >> >> However, this breaks in the case where an attended transfer has happened >> (the UUID of the call that gets connected "belonged" to the receptionist). >> >> The most robust thing I can think of is to do a live lookup of the call >> UUID using the "final" extension number. Is there any way to do this >> lookup from the dialplan? >> >> To be clear, what I'd like to do is this: >> >> Incoming call is picked up by receptionist (ext 100). Receptionist >> transfers call to extension 101. The Boss (extension 200), dials *88101 >> which looks up the UUID of extension 101's current call, provides this >> UUID to the eavesdrop extension, and the eavesdrop proceeds. >> >> There doesn't appear to be any way to do this from the dialplan, however >> -- am I missing something? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/e756a116/attachment-0001.html From michel.brabants at gmail.com Wed Oct 30 19:59:44 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 30 Oct 2013 17:59:44 +0100 Subject: [Freeswitch-users] UUID from Extension Number In-Reply-To: References: <526FDA1B.2060304@ssimicro.com> <527121E5.60409@ssimicro.com> Message-ID: It's also a patch ofcourse and I think that an extension that lists the uuid's for an extension would be usefull anyway. Michel On Wed, Oct 30, 2013 at 5:58 PM, Michel Brabants wrote: > I've written a patch for freeswitch that allows you to execute a > dialplan-command/api from the dialplan when the transfer succeeds ... It > has been lying around for some time in my local branch waiting to be > uploaded as a jira-patch ... > > I think that this would solve you problem. > > Michel > > > On Wed, Oct 30, 2013 at 4:12 PM, Jeremy Childs wrote: > >> Yes, that works fine, except in the case where the incoming call >> bounces off a receptionist. >> >> Receptionist takes call - UUID 1234 is mapped (via mod_db) to the >> receptionist's extension number. >> Receptionist begins an attended transfer - new UUID is generated (6789) - >> this UUID is mapped to the tranfer target (extension 101). >> Receptionist completes transfer - new UUID (6789) is discarded and UUID >> 1234 is now the UUID of the call. >> Eavesdropper wants to listen to whatever extension 101 is doing... no >> mapping for this. >> >> I have a patch for mod_dptools ready to submit that allows eavesdrop to >> lookup an extension number, I'm just checking to see if I've missed some >> other way to do it. >> >> >> >> On 10/30/13, 2:01 AM, Steven Ayre wrote: >> >> Use mod_db to store the current call's uuid and then again to look it up. >> >> >> On 29 October 2013 15:54, Jeremy Childs wrote: >> >>> I'm trying to create a "works-anywhere" eavesdrop extension. I've got >>> the demo application that uses the hash/spymap functionality working >>> correctly, but you would need the caller-id of the incoming call in >>> order to eavesdrop on incoming calls. This could be fixed by capturing >>> the destination number instead of the caller-id number for incoming >>> calls (and I've done this, it works fine). >>> >>> However, this breaks in the case where an attended transfer has happened >>> (the UUID of the call that gets connected "belonged" to the >>> receptionist). >>> >>> The most robust thing I can think of is to do a live lookup of the call >>> UUID using the "final" extension number. Is there any way to do this >>> lookup from the dialplan? >>> >>> To be clear, what I'd like to do is this: >>> >>> Incoming call is picked up by receptionist (ext 100). Receptionist >>> transfers call to extension 101. The Boss (extension 200), dials *88101 >>> which looks up the UUID of extension 101's current call, provides this >>> UUID to the eavesdrop extension, and the eavesdrop proceeds. >>> >>> There doesn't appear to be any way to do this from the dialplan, however >>> -- am I missing something? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/afc0f4d1/attachment.html From Vladislav.Grishin at vts24.ru Wed Oct 30 21:36:31 2013 From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=) Date: Wed, 30 Oct 2013 22:36:31 +0400 Subject: [Freeswitch-users] ACK disappeared from SIP session, Help Message-ID: <527151AF.6070009@vts24.ru> the scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone FreeSWITCH IP is 191.21.19.12 Asterisk IP is 162.48.13.26 Call comes to FreeSWITCH from a Cisco 2811. SIP session between freeswitch and asterisk(1.8.20) |Time | 191.21.19.12 | | | | 162.48.13.26 | |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" (5060) | |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" (5060) | |21.065794| 100 Trying| |SIP Status | |(5065) <------------------ (5060) | |21.122406| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |21.143580| RTP (g711A) |RTP Num packets:50 Duration:0.922s SSRC:0xB5320A6 | |(21866) <------------------ (12772) | |22.282684| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |22.296762| 181 Call is being forwarded |SIP Status | |(5065) <------------------ (5060) | |27.558268| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |35.039883| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.307222| RTP (g711A) |RTP Num packets:1490 Duration:29.773s SSRC:0x530CAB37 | |(21866) <------------------ (12772) | |35.476614| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.532995| RTP (g711A) |RTP Num packets:2966 Duration:29.680s SSRC:0x308FC816 | |(21866) ------------------> (12772) | |36.436846| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |38.282686| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |42.025399| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |45.637280| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |49.343115| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |53.119940| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |56.907800| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |60.791453| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |64.606730| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |65.219392| BYE | |SIP Request | |(5065) <------------------ (5060) | |65.234354| 200 OK | |SIP Status | |(5065) ------------------> (5060) | |65.234359| 200 OK | |SIP Status | |(5065) ------------------> (5060) | It is visible that there is no package with ACK confirmation from a FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a freeswitch, and without having waited confirmations (ACK) after 23 seconds sends SIP BY on a freeswitch and on IP Phone. After that call stops. Before disconnection subscribers hear each other of 29 (+-) seconds In tcpdump between Cisco and FreeSWITCH it is visible that the SIP ACK packet from Cisco arrives on a freeswitch. In what place the SIP ACK in a FreeSWITCH disappeared? What to me to analyse in SIP session? Maybe someone solved a similar problem? please, Help! Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/00d65a38/attachment-0001.html From nreis at wavecom.pt Wed Oct 30 21:44:41 2013 From: nreis at wavecom.pt (Nuno Reis) Date: Wed, 30 Oct 2013 18:44:41 +0000 Subject: [Freeswitch-users] mod_cdr_mongodb: leg_a doesn't exist when calling between 2 local extensions neither on incoming calls on a tenant Message-ID: Hi guys. I'm changing my cdr database to mongo and i'm using mod_cdr_mongodb . Everything looked promising although i'm facing an issue: - When a call is placed between 2 local extension (on the same tenant), there is no leg_a on mongo's cdr table. - I also have the issue on incoming call (Ex: from a SIP gateway) I can only see everything as expected when placing outgoing calls (let's say on a SIP gateway). Can someone confirm this behavior. I'm using FS 1.2.12. Looking forward to hear from you. BR, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis at wavecom.pt WAVECOM-Solu??es R?dio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS| www.wavecom.pt** * [image: Description: Description: WavecomSignature] [image: Publicity] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/806b8f27/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 16423 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/806b8f27/attachment.png From Vladislav.Grishin at vts24.ru Wed Oct 30 21:57:30 2013 From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=) Date: Wed, 30 Oct 2013 22:57:30 +0400 Subject: [Freeswitch-users] ACK disappeared from SIP session, Help In-Reply-To: <527151AF.6070009@vts24.ru> References: <527151AF.6070009@vts24.ru> Message-ID: <5271569A.5080308@vts24.ru> the correct scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP----NAT----asterisk(1.8.20)---IP Phone Asterisk IP is 162.48.13.26 (External IP NAT ) 30.10.2013 22:36, "?????? ?.?." ?????: > > the scheme is PBX---E1--Cisco 2811-----VoIP > SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone > > FreeSWITCH IP is 191.21.19.12 > Asterisk IP is 162.48.13.26 > Call comes to FreeSWITCH from a Cisco 2811. > > SIP session between freeswitch and asterisk(1.8.20) > > > |Time | 191.21.19.12 | > | | | 162.48.13.26 | > |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" > | |(5065) ------------------> (5060) | > |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" > | |(5065) ------------------> (5060) | > |21.065794| 100 Trying| |SIP Status > | |(5065) <------------------ (5060) | > |21.122406| 180 Ringing |SIP Status > | |(5065) <------------------ (5060) | > |21.143580| RTP (g711A) |RTP Num packets:50 > Duration:0.922s SSRC:0xB5320A6 > | |(21866) <------------------ (12772) | > |22.282684| 180 Ringing |SIP Status > | |(5065) <------------------ (5060) | > |22.296762| 181 Call is being forwarded |SIP Status > | |(5065) <------------------ (5060) | > |27.558268| 180 Ringing |SIP Status > | |(5065) <------------------ (5060) | > |35.039883| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |35.307222| RTP (g711A) |RTP Num > packets:1490 Duration:29.773s SSRC:0x530CAB37 > | |(21866) <------------------ (12772) | > |35.476614| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |35.532995| RTP (g711A) |RTP Num > packets:2966 Duration:29.680s SSRC:0x308FC816 > | |(21866) ------------------> (12772) | > |36.436846| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |38.282686| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |42.025399| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |45.637280| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |49.343115| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |53.119940| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |56.907800| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |60.791453| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |64.606730| 200 OK SDP (g711A) |SIP Status > | |(5065) <------------------ (5060) | > |65.219392| BYE | |SIP Request > | |(5065) <------------------ (5060) | > |65.234354| 200 OK | |SIP Status > | |(5065) ------------------> (5060) | > |65.234359| 200 OK | |SIP Status > | |(5065) ------------------> (5060) | > > It is visible that there is no package with ACK confirmation from a > FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a > freeswitch, and without having waited confirmations (ACK) after 23 > seconds sends SIP BY on a freeswitch and on IP Phone. After that call > stops. Before disconnection subscribers hear each other of 29 (+-) seconds > > In tcpdump between Cisco and FreeSWITCH it is visible that the SIP ACK > packet from Cisco arrives on a freeswitch. > > In what place the SIP ACK in a FreeSWITCH disappeared? > What to me to analyse in SIP session? > Maybe someone solved a similar problem? > please, Help! > > Vladislav Grishin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/ae0274c4/attachment-0001.html From jason.holden at start.ca Wed Oct 30 22:14:37 2013 From: jason.holden at start.ca (Jason Holden) Date: Wed, 30 Oct 2013 15:14:37 -0400 Subject: [Freeswitch-users] core dump problems fs 1.2.14 oct 17 Message-ID: freeswitch v1.2.14 date october 17 Currently we are experiencing random crashes, can anyone provide advice why? I have a core dump I can send, not sure if the mail list will permit this file size? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/87058ad7/attachment.html From Vladislav.Grishin at vts24.ru Wed Oct 30 23:02:48 2013 From: Vladislav.Grishin at vts24.ru (=?UTF-8?B?ItCT0YDQuNGI0LjQvSDQki7QoS4i?=) Date: Thu, 31 Oct 2013 00:02:48 +0400 Subject: [Freeswitch-users] ACK disappeared from SIP session, Help In-Reply-To: <5271569A.5080308@vts24.ru> References: <527151AF.6070009@vts24.ru> <5271569A.5080308@vts24.ru> Message-ID: <527165E8.9010901@vts24.ru> Call in direction from IP Phone to PBX works fine. 30.10.2013 22:57, "?????? ?.?." ?????: > > > the correct scheme is PBX---E1--Cisco 2811-----VoIP > SIP-----FreeSWITCH-----VoIP SIP----NAT----asterisk(1.8.20)---IP Phone > > Asterisk IP is 162.48.13.26 (External IP NAT ) > > > 30.10.2013 22:36, "?????? ?.?." ?????: >> >> the scheme is PBX---E1--Cisco 2811-----VoIP >> SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone >> >> FreeSWITCH IP is 191.21.19.12 >> Asterisk IP is 162.48.13.26 >> Call comes to FreeSWITCH from a Cisco 2811. >> >> SIP session between freeswitch and asterisk(1.8.20) >> >> >> |Time | 191.21.19.12 | >> | | | 162.48.13.26 | >> |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" >> > | |(5065) ------------------> (5060) | >> |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" >> > | |(5065) ------------------> (5060) | >> |21.065794| 100 Trying| |SIP Status >> | |(5065) <------------------ (5060) | >> |21.122406| 180 Ringing |SIP Status >> | |(5065) <------------------ (5060) | >> |21.143580| RTP (g711A) |RTP Num >> packets:50 Duration:0.922s SSRC:0xB5320A6 >> | |(21866) <------------------ (12772) | >> |22.282684| 180 Ringing |SIP Status >> | |(5065) <------------------ (5060) | >> |22.296762| 181 Call is being forwarded |SIP Status >> | |(5065) <------------------ (5060) | >> |27.558268| 180 Ringing |SIP Status >> | |(5065) <------------------ (5060) | >> |35.039883| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |35.307222| RTP (g711A) |RTP Num >> packets:1490 Duration:29.773s SSRC:0x530CAB37 >> | |(21866) <------------------ (12772) | >> |35.476614| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |35.532995| RTP (g711A) |RTP Num >> packets:2966 Duration:29.680s SSRC:0x308FC816 >> | |(21866) ------------------> (12772) | >> |36.436846| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |38.282686| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |42.025399| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |45.637280| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |49.343115| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |53.119940| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |56.907800| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |60.791453| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |64.606730| 200 OK SDP (g711A) |SIP Status >> | |(5065) <------------------ (5060) | >> |65.219392| BYE | |SIP Request >> | |(5065) <------------------ (5060) | >> |65.234354| 200 OK | |SIP Status >> | |(5065) ------------------> (5060) | >> |65.234359| 200 OK | |SIP Status >> | |(5065) ------------------> (5060) | >> >> It is visible that there is no package with ACK confirmation from a >> FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a >> freeswitch, and without having waited confirmations (ACK) after 23 >> seconds sends SIP BY on a freeswitch and on IP Phone. After that call >> stops. Before disconnection subscribers hear each other of 29 (+-) >> seconds >> >> In tcpdump between Cisco and FreeSWITCH it is visible that the SIP >> ACK packet from Cisco arrives on a freeswitch. >> >> In what place the SIP ACK in a FreeSWITCH disappeared? >> What to me to analyse in SIP session? >> Maybe someone solved a similar problem? >> please, Help! >> >> Vladislav Grishin >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > Vladislav Grishin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Vladislav Grishin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131031/0666425a/attachment-0001.html From khorsmann at gmail.com Wed Oct 30 23:33:49 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 30 Oct 2013 21:33:49 +0100 Subject: [Freeswitch-users] core dump problems fs 1.2.14 oct 17 In-Reply-To: References: Message-ID: Hi, if your FreeSWITCH coredumps and it seems to be a bug, then you better fill out an JIRA Bug-Report and attached an gzipped coredump and anything what JIRA form wants for the FS-Devs to investigate. 2013/10/30 Jason Holden > freeswitch v1.2.14 date october 17**** > > Currently we are experiencing random crashes, can anyone provide advice > why?**** > > I have a core dump I can send, not sure if the mail list will permit this > file size?**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Mit freundlichen Gr??en *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/ec1e47a4/attachment.html From krice at freeswitch.org Wed Oct 30 23:43:06 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 30 Oct 2013 15:43:06 -0500 Subject: [Freeswitch-users] core dump problems fs 1.2.14 oct 17 In-Reply-To: Message-ID: NO NO NO NO NO Do not attach a gzip?d coredump We can not do anything with a coredump... You need to get a backtrace... See http://wiki.freeswitch.org/wiki/Reporting_Bugs On 10/30/13 3:33 PM, "Karsten Horsmann" wrote: > Hi, > > if your FreeSWITCH coredumps and it seems to be a bug, then you better fill > out an JIRA Bug-Report and attached an gzipped coredump and anything what JIRA > form wants for the FS-Devs to investigate. > > > > 2013/10/30 Jason Holden >> ??????????? freeswitch v1.2.14 date october 17 >> Currently we are experiencing random crashes, can anyone provide advice why? >> I have a core dump I can send, not sure if the mail list will permit this >> file size? >> ? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131030/8fcd064a/attachment.html From arkaha at hotbox.ru Thu Oct 31 16:12:13 2013 From: arkaha at hotbox.ru (Limit) Date: Thu, 31 Oct 2013 06:12:13 -0700 (PDT) Subject: [Freeswitch-users] Video only call Message-ID: <1383225133515-7595976.post@n2.nabble.com> Good day! I'am experimenting with SIP video calls (using VP8 video codec). Ordinary audio+video calls works very well (audio recordered and video passed through). But if I try to make only video call, i.e. SDP contains only video media, than I got an error and channels hang up: 2013-10-31 17:00:38.494743 [DEBUG] sofia.c:5708 Channel sofia/internal/sip:1008 at 10.10.104.125:10018 entering state [ready][200] 2013-10-31 17:00:38.494743 [DEBUG] sofia_glue.c:5382 Video Codec Compare [VP8:99]/[H264:97] 2013-10-31 17:00:38.494743 [DEBUG] sofia_glue.c:5382 Video Codec Compare [VP8:99]/[VP8:99] 2013-10-31 17:00:38.494743 [ERR] sofia_glue.c:3025 No audio codec available 2013-10-31 17:00:38.494743 [DEBUG] sofia_glue.c:2975 Set VIDEO Codec sofia/internal/sip:1008 at 10.10.104.125:10018 VP8/90000 0 ms 2013-10-31 17:00:38.514745 [ERR] sofia.c:6464 RTP Error! 2013-10-31 17:00:38.514745 [NOTICE] sofia.c:6465 Hangup sofia/internal/sip:1008 at 10.10.104.125:10018 [CS_CONSUME_MEDIA] [DESTINATION_OUT_OF_ORDER] 2013-10-31 17:00:38.514745 [DEBUG] switch_channel.c:3130 Send signal sofia/internal/sip:1008 at 10.10.104.125:10018 [KILL] Incoming SDP information looks like this: v=0 o=- 1383224434 1 IN IP4 10.10.104.125 s=Test/2.0beta263 c=IN IP4 10.10.104.125 t=0 0 m=video 5000 RTP/AVP 31 34 110 109 102 103 96 108 107 b=AS:768 b=AS:768000 a=rtpmap:31 h261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:34 H263/90000 a=fmtp:34 F=1;CIF=1;CIF16=1;CIF4=1;maxbr=7680;QCIF=1;SQCIF=1 a=rtpmap:110 H263-1998/90000 a=fmtp:110 D=1;F=1;I=1;J=1;CIF=1;CIF16=1;CIF4=1;maxbr=7680;QCIF=1;SQCIF=1 a=rtpmap:109 H264/90000 a=fmtp:109 max-fs=9000;max-mbps=270000;profile-level-id=428028 a=rtpmap:102 H264/90000 a=fmtp:102 packetization-mode=1;max-fs=9000;max-mbps=270000;profile-level-id=428028 a=rtpmap:103 H264/90000 a=fmtp:103 packetization-mode=1;max-fs=9000;max-mbps=270000;profile-level-id=640028 a=rtpmap:96 raw/90000 a=fmtp:96 rate=90000;height=288;width=352;colorimetry=BT601-5;depth=8;sampling=YCbCr-4:2:0 a=rtpmap:108 X-MX-VP8/90000 a=fmtp:108 x-mx-max-size=1920x1200 a=rtpmap:107 VP8/90000 a=fmtp:107 max-fs=9000 a=rtcp-fb:* pli fir tmmbr tstr freeswitch local SDP: v=0 o=FreeSWITCH 1383203862 1383203863 IN IP4 10.10.104.125 s=FreeSWITCH c=IN IP4 10.10.104.125 t=0 0 m=video 32038 RTP/AVP 98 99 a=rtpmap:98 H264/90000 a=fmtp:98 packetization-mode=1;max-fs=9000;max-mbps=270000;profile-level-id=640028 a=rtpmap:99 VP8/90000 a=fmtp:99 max-fs=9000 and remote SDP: v=0 o=- 1383224434 1 IN IP4 10.10.104.125 s=SimpleOPAL/3.12.7 c=IN IP4 10.10.104.125 t=0 0 m=video 5002 RTP/AVP 99 b=AS:16000 b=AS:16000000 a=rtpmap:99 VP8/90000 a=fmtp:99 max-fs=9000 a=rtcp-fb:* pli fir tmmbr tstr -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-only-call-tp7595976.html Sent from the freeswitch-users mailing list archive at Nabble.com. From arkaha at hotbox.ru Thu Oct 31 16:35:00 2013 From: arkaha at hotbox.ru (Limit) Date: Thu, 31 Oct 2013 06:35:00 -0700 (PDT) Subject: [Freeswitch-users] Code negotiation based on local priority In-Reply-To: <1383139631286-7595975.post@n2.nabble.com> References: <1383139631286-7595975.post@n2.nabble.com> Message-ID: <1383226500826-7595977.post@n2.nabble.com> Put following line in your sip profile: in that case local FreeSWITCH codec preference list take precedence. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Code-negotiation-based-on-local-priority-tp7595975p7595977.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ashou at mobile-sphere.com Thu Oct 31 18:24:00 2013 From: ashou at mobile-sphere.com (Ashou Han) Date: Thu, 31 Oct 2013 11:24:00 -0400 Subject: [Freeswitch-users] max-sessions value small may cause crash Message-ID: <52727610.6020907@mobile-sphere.com> Hi, My FS is FreeSWITCH Version 1.2.12+git~20130910T181609Z~f5ba0bb7d4 (git f5ba0bb 2013-09-10 18:16:09Z) I set , in switch.conf.xml FS carshed very offen like this in messages log: Oct 26 19:10:21 fs kernel: [4592903.781042] freeswitch[29353]: segfault at 0 ip 00007f4e22e0c8cd sp 00007f4e0924e5b0 error 4 in libfreeswitch.so.1.0.0[7f4e22ce4000+229000] Oct 26 23:41:45 fs kernel: [4609190.441226] freeswitch[518]: segfault at 0 ip 00007ff71baf58cd sp 00007ff7068ae5b0 error 4 in libfreeswitch.so.1.0.0[7ff71b9cd000+229000] Oct 28 09:57:58 fs kernel: [89606.041406] freeswitch[23780]: segfault at 0 ip 00007f68651d78cd sp 00007f684ff205b0 error 4 in libfreeswitch.so.1.0.0[7f68650af000+229000] Oct 29 16:55:24 fs kernel: [201076.501294] freeswitch[16422]: segfault at 0 ip 00007f0b68fc48cd sp 00007f0b4f50d5b0 error 4 in libfreeswitch.so.1.0.0[7f0b68e9c000+229000] lots in freeswitch.log: 2013-10-25 15:25:24.651750 [CRIT] switch_core_session.c:2228 Over Session Limit! 20 2013-10-25 15:25:24.651750 [CRIT] mod_sofia.c:4869 Error Creating Session 2013-10-25 18:36:03.311729 [CRIT] switch_core_session.c:2228 Over Session Limit! 20 2013-10-25 18:36:03.311729 [CRIT] mod_sofia.c:4869 Error Creating Session ... 2013-10-30 15:01:46.501223 [CRIT] mod_sofia.c:4869 Error Creating Session 2013-10-30 15:01:46.521224 [CRIT] switch_time.c:1007 Over Session Rate of 10! May this cause the FS crash? I have another FS server with same Hardware and FS version, but and no crash issues. Thanks, Ashou Han From anthony.minessale at gmail.com Thu Oct 31 19:42:00 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Oct 2013 11:42:00 -0500 Subject: [Freeswitch-users] max-sessions value small may cause crash In-Reply-To: <52727610.6020907@mobile-sphere.com> References: <52727610.6020907@mobile-sphere.com> Message-ID: You would have to gather a backtrace and post it to jira. https://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace On Thu, Oct 31, 2013 at 10:24 AM, Ashou Han wrote: > Hi, > > My FS is FreeSWITCH Version 1.2.12+git~20130910T181609Z~f5ba0bb7d4 (git > f5ba0bb 2013-09-10 18:16:09Z) > > I set , name="sessions-per-second" value="10"/> in switch.conf.xml > > FS carshed very offen like this in messages log: > > Oct 26 19:10:21 fs kernel: [4592903.781042] freeswitch[29353]: segfault > at 0 ip 00007f4e22e0c8cd sp 00007f4e0924e5b0 error 4 in > libfreeswitch.so.1.0.0[7f4e22ce4000+229000] > Oct 26 23:41:45 fs kernel: [4609190.441226] freeswitch[518]: segfault at > 0 ip 00007ff71baf58cd sp 00007ff7068ae5b0 error 4 in > libfreeswitch.so.1.0.0[7ff71b9cd000+229000] > Oct 28 09:57:58 fs kernel: [89606.041406] freeswitch[23780]: segfault at > 0 ip 00007f68651d78cd sp 00007f684ff205b0 error 4 in > libfreeswitch.so.1.0.0[7f68650af000+229000] > Oct 29 16:55:24 fs kernel: [201076.501294] freeswitch[16422]: segfault > at 0 ip 00007f0b68fc48cd sp 00007f0b4f50d5b0 error 4 in > libfreeswitch.so.1.0.0[7f0b68e9c000+229000] > > lots in freeswitch.log: > > 2013-10-25 15:25:24.651750 [CRIT] switch_core_session.c:2228 Over > Session Limit! 20 > 2013-10-25 15:25:24.651750 [CRIT] mod_sofia.c:4869 Error Creating Session > 2013-10-25 18:36:03.311729 [CRIT] switch_core_session.c:2228 Over > Session Limit! 20 > 2013-10-25 18:36:03.311729 [CRIT] mod_sofia.c:4869 Error Creating Session > ... > 2013-10-30 15:01:46.501223 [CRIT] mod_sofia.c:4869 Error Creating Session > 2013-10-30 15:01:46.521224 [CRIT] switch_time.c:1007 Over Session Rate > of 10! > > May this cause the FS crash? > I have another FS server with same Hardware and FS version, but name="max-sessions" value="2000"/> and no crash issues. > > Thanks, > Ashou Han > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131031/2e1d38f8/attachment-0001.html From alipey at gmail.com Thu Oct 31 21:16:10 2013 From: alipey at gmail.com (Ali Pey) Date: Thu, 31 Oct 2013 14:16:10 -0400 Subject: [Freeswitch-users] bind_digit_action stops working when the call is sent to a socket Message-ID: Hello, As soon as the call is sent to a socket, bind digit actions stop working. I have tried setting bind_digit_action both before and after the call is sent to the socket. It doesn't work. Before call is sent to socket, I see: 2013-10-31 10:48:38.183166 [DEBUG] switch_rtp.c:3915 RTP RECV DTMF 5:3520 2013-10-31 10:48:38.183166 [DEBUG] mod_dptools.c:186 sofia/sip-trunks/ 6132952508 at qasip.onebox.com Digit match binding [exec:playback][ivr/ivr-please_enter_extension_followed_by_pound.wav] After is sent to the socket, it becomes: 2013-10-31 11:07:38.042566 [DEBUG] switch_rtp.c:3915 RTP RECV DTMF 5:4160 2013-10-31 11:07:38.042566 [DEBUG] switch_channel.c:419 RECV DTMF 5:4160 2013-10-31 11:07:38.042566 [DEBUG] mod_dptools.c:2049 Digit 5 Any ideas? Is this a bug? Is there a work around? Thanks, Ali Pey -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131031/b97f3b19/attachment.html From fvillarroel at yahoo.com Thu Oct 31 23:07:49 2013 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Thu, 31 Oct 2013 13:07:49 -0700 (PDT) Subject: [Freeswitch-users] mod_lcr and UnixODBC Error Message-ID: <1383250069.76134.YahooMailNeo@web162005.mail.bf1.yahoo.com> Dear all. I am trying to use mod_lcr, but in my tests i did received the follow error: 2013-10-31 16:35:03.916887 [ERR] switch_core_sqldb.c:1114 ERR: [SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS lcr_rate_field, ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? JOIN carrier_gateway cg ON c.id=cg.carrier_id ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND ( ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? (digits IN (12145551111, 1214555111, 121455511, 12145551, 1214555, 121455, 12145, 1214, 121, 12, 1) ? ? AND lrn = false) OR ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? (digits IN (12145551111, 1214555111, 121455511, 12145551, 1214555, 121455, 12145, 1214, 121, 12, 1) AND lrn = true)) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, ?rate, ?quality DESC, ?reliability DESC, random();] [STATE: 42883 CODE 7 ERROR: [unixODBC]ERROR: el operador no existe: prefix_range = bigint; Error while executing the query ] I am using PostgreSQL 8.4.11 and Debian Squezee How i can solve this issue? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131031/7ab2bd1a/attachment.html