- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Network Card
by Anthony Minessale 14 Aug '14

14 Aug '14

Ok it isn't obscene. Thread closed. On Aug 14, 2014 7:27 PM, "aademattia(a)comcast.net" <aademattia(a)comcast.net> wrote: > Why is 1000 cps obscene? > Denovo lab uses nokia stack and we get over 1k cps in a server that cost > 2k. > > > > Sent from my HTC > > ----- Reply message ----- > From: "Anthony Minessale" <anthony.minessale(a)gmail.com> > To: "Freeswitch-users" <freeswitch-users(a)lists.freeswitch.org> > Subject: [Freeswitch-users] Network Card > Date: Thu, Aug 14, 2014 6:16 PM > > We don't give advice on load related questions outside what you have > already seen in this thread. > > 1000cps is obscene. If you are actually getting paid for 1000cps you can > surely afford to invest in hardware and professional help. > On Aug 14, 2014 3:01 PM, "Andre" <andretodd(a)verizon.net> wrote: > >> Just testing for now, no production traffic. Trying to find the breaking >> point. >> >> >> >> The reason for the questions are because I’m testing FS and after a >> certain point I am getting very high PDD, more than 6 seconds. Per the >> CDR’s it takes about 6 seconds from start time in CDR Inbound to CDR >> outbound. >> >> I have tried 1 to 10 profiles to see if that makes a difference but it >> doesn’t and 1 to 4 consoles running one profile each still no difference. >> >> >> >> Any idea why the CDR’s should show it would take 6 seconds to get from >> start of inbound to start of outbound? >> >> >> >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Ken Rice >> *Sent:* Thursday, August 14, 2014 2:06 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Network Card >> >> >> >> 1000 CPS the NIC is not the bottle neck this is actually relatively small >> amount of TCP/UDP traffic for just the SIP message. >> >> Now the RTP @ 1000 calls on the other hand assuming G711 at 20ms ptime. >> And assuming 1 leg calls originating from a dialer would generate >> 80megabits of RTP (at the IP layer even more on the wire), and approx 50K >> PPS. Assuming 2 leg’d calls bridging media 160megabit and 100K PPS of just >> RTP... >> >> But then we have to extrapolate peak concurrent from 1000 CPS... Assuming >> dialer traffic again, 75% ASR and 20 second ACD, and lets assume the 25% >> are just hard failed for the calculation (no early media on these) that >> gives us 750 new calls/sec and over the course of 20 seconds we accumulate >> 15,000 concurrent calls.. >> >> So lets take that and extrapolate bandwidth and PPS requirements for >> single leg’d calls of 1.43Gig/sec (on the wire) and 750K PPS so that’s >> going to exceed the bandwidth of a GIG-E connection so you’d have to break >> this into 2 RTP endpoints in sofia, and make sure you switches/routers >> could handle the PPS loads. >> >> Double that to 2 leg calls and you have 2.86G/sec of RTP (again on the >> wire including UDP, IP, ethernet framing over heads) and pushing 1.5million >> PPS. >> >> Here again this dictates at least 3 gig-e NICs on your FS box. Is this >> possible? I’m sure it is with the right hardware configuration... But lets >> step back a minute and really consider what is going on... >> >> You are now putting approx over $2000/hr of traffic on a very expensive >> box that can and will at some point fail. Why are you doing such a thing? >> You’re better off limiting rates to sub 1000 CPS, (say 500CPS) deploying 3 >> boxes to handle the load in n+1 configuration, and if you lose a box you >> arent losing $2000+ in revenue and cranky customers while you >> repair/replace the now dead machine. >> >> Lets face it, you can purchase a colo a nice machine for say 200 to >> 300$/month (assuming bandwidth costs are the same as above so that’s a wash >> instead of trying to cram it all on box that ends up costing you 1000/month >> by the time you pay for the hardware and colocation expenses for it. >> >> >> >> >> >> On 8/14/14 11:56 AM, "Russell Treleaven" <rtreleaven(a)bunnykick.ca> wrote: >> >> 1000 calls per second... no idea but I really doubt the network interface >> is going to be the bottleneck. >> Not sure why you make a distinction here between sessions and connected. >> To be clear you mean 5000 bridge calls correct? >> Since media is not an issue I think you can just start testing with sipp >> and bwm-ng to see what is what. >> >> 1000 calls per second is a very big number. >> Let us know how you make out. >> >> Sincerely, >> >> Russell Treleaven >> >> >> >> >> >> On Wed, Aug 13, 2014 at 5:40 PM, Andre <andretodd(a)verizon.net> wrote: >> >> 1000 calls per second and 10000 sessions with 4000 connected. What would >> be recommend? >> >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ >> mailto:freeswitch-users-bounces@lists.freeswitch.org >> <freeswitch-users-bounces(a)lists.freeswitch.org>] *On Behalf Of *Russell >> Treleaven >> *Sent:* Wednesday, August 13, 2014 5:25 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Network Card >> >> >> >> If the media is bypassing you then you only need bandwidth for the >> signalling. >> >> >> >> Depends primarily on the calls per second. >> >> >> >> bwm-ng will help you to see whats going on. >> >> >> >> >> >> >> >> >> >> On Wed, Aug 13, 2014 at 5:20 PM, Andre <andretodd(a)verizon.net> wrote: >> >> >> >> Hi, is there anything special with the network card to get 10,000 >> sessions in media bypass mode with FreeSwitch? Will a Gig E be enough >> bandwidth? >> >> What would it look like if the network card couldn’t keep up and how can >> I verify the network card is the bottleneck? >> Thanks >> Andre >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www..freeswitchsolutions.com <http://www.freeswitchsolutions.com> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)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.FreeSWITCH.org>http://www.ClueCon.com <http://www.ClueCon.com> >> http://www.OSTAG.org <http://www.OSTAG.org>*irc.freenode.net #freeswitch >> Twitter: @FreeSWITCH >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www..freeswitchsolutions.com <http://www.freeswitchsolutions.com> >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] Network Card
by aademattia＠comcast.net 14 Aug '14

14 Aug '14

Why is 1000 cps obscene? Denovo lab uses nokia stack and we get over 1k cps in a server that cost 2k. Sent from my HTC ----- Reply message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: "Freeswitch-users" <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] Network Card Date: Thu, Aug 14, 2014 6:16 PM We don't give advice on load related questions outside what you have already seen in this thread. 1000cps is obscene. If you are actually getting paid for 1000cps you can surely afford to invest in hardware and professional help. On Aug 14, 2014 3:01 PM, "Andre" <andretodd(a)verizon.net> wrote: Just testing for now, no production traffic. Trying to find the breaking point. The reason for the questions are because I’m testing FS and after a certain point I am getting very high PDD, more than 6 seconds. Per the CDR’s it takes about 6 seconds from start time in CDR Inbound to CDR outbound. I have tried 1 to 10 profiles to see if that makes a difference but it doesn’t and 1 to 4 consoles running one profile each still no difference. Any idea why the CDR’s should show it would take 6 seconds to get from start of inbound to start of outbound? From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, August 14, 2014 2:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Network Card 1000 CPS the NIC is not the bottle neck this is actually relatively small amount of TCP/UDP traffic for just the SIP message. Now the RTP @ 1000 calls on the other hand assuming G711 at 20ms ptime. And assuming 1 leg calls originating from a dialer would generate 80megabits of RTP (at the IP layer even more on the wire), and approx 50K PPS. Assuming 2 leg’d calls bridging media 160megabit and 100K PPS of just RTP... But then we have to extrapolate peak concurrent from 1000 CPS... Assuming dialer traffic again, 75% ASR and 20 second ACD, and lets assume the 25% are just hard failed for the calculation (no early media on these) that gives us 750 new calls/sec and over the course of 20 seconds we accumulate 15,000 concurrent calls.. So lets take that and extrapolate bandwidth and PPS requirements for single leg’d calls of 1.43Gig/sec (on the wire) and 750K PPS so that’s going to exceed the bandwidth of a GIG-E connection so you’d have to break this into 2 RTP endpoints in sofia, and make sure you switches/routers could handle the PPS loads. Double that to 2 leg calls and you have 2.86G/sec of RTP (again on the wire including UDP, IP, ethernet framing over heads) and pushing 1.5million PPS. Here again this dictates at least 3 gig-e NICs on your FS box. Is this possible? I’m sure it is with the right hardware configuration... But lets step back a minute and really consider what is going on... You are now putting approx over $2000/hr of traffic on a very expensive box that can and will at some point fail. Why are you doing such a thing? You’re better off limiting rates to sub 1000 CPS, (say 500CPS) deploying 3 boxes to handle the load in n+1 configuration, and if you lose a box you arent losing $2000+ in revenue and cranky customers while you repair/replace the now dead machine. Lets face it, you can purchase a colo a nice machine for say 200 to 300$/month (assuming bandwidth costs are the same as above so that’s a wash instead of trying to cram it all on box that ends up costing you 1000/month by the time you pay for the hardware and colocation expenses for it. On 8/14/14 11:56 AM, "Russell Treleaven" <rtreleaven(a)bunnykick.ca> wrote: 1000 calls per second... no idea but I really doubt the network interface is going to be the bottleneck. Not sure why you make a distinction here between sessions and connected. To be clear you mean 5000 bridge calls correct? Since media is not an issue I think you can just start testing with sipp and bwm-ng to see what is what. 1000 calls per second is a very big number. Let us know how you make out. Sincerely, Russell Treleaven On Wed, Aug 13, 2014 at 5:40 PM, Andre <andretodd(a)verizon.net> wrote: 1000 calls per second and 10000 sessions with 4000 connected. What would be recommend? From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Treleaven Sent: Wednesday, August 13, 2014 5:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Network Card If the media is bypassing you then you only need bandwidth for the signalling. Depends primarily on the calls per second. bwm-ng will help you to see whats going on. On Wed, Aug 13, 2014 at 5:20 PM, Andre <andretodd(a)verizon.net> wrote: Hi, is there anything special with the network card to get 10,000 sessions in media bypass mode with FreeSwitch? Will a Gig E be enough bandwidth? What would it look like if the network card couldn’t keep up and how can I verify the network card is the bottleneck? Thanks Andre _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www..freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)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 Twitter: @FreeSWITCH _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www..freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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How does DNS SRV and SIP OPTIONS logic combine for a gateway?
by Victor Chukalovskiy 14 Aug '14

14 Aug '14

Good day, Anyone know how exactly FS handles a Sofia gateway configured with the combination below? * proxy resolved through DNS SRV; returns 5 records of the same weight * registration disabled * SIP OPTIONS enabled Is it going to send OPTIONS to 5 IPs in round-robin? What if 2 out 5 fail OPTIONS but the rest remain active? Is FS going to intelligently round-robin calls between remaining 3 IPs or it will mark the entire gateway as "down"? Thank you, -Victor

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ClueCon Video Cameras for Sale!!
by Ken Rice 14 Aug '14

14 Aug '14

Hey FreeSWITCHers, Here¹s your chance to own a little Piece of ClueCon 2014. I have 2 Cannon Vixia HF R500 HD camcorder setups used only for Recording Video at ClueCon! ( http://www.amazon.com/gp/product/B00HLDFTRS/ref=oh_aui_detailpage_o05_s00?ie =UTF8&psc=1 ) Included with the Camera is a Tripod and 2 SD Cards (all were purchased brand new just to use @ ClueCon). Whats included with the camera? Battery, Charger, cables for HDMI and USB, 2 x 32G SD Cards (good for 4hours of 1080P Video per card), and a Tripod with a carrying bag for the tripod. All original Docs are included. Sorry the Original boxes were sacrificed in Chicago at the train station so I could make it home with the cameras. I am asking $350 for each Kit! (the camera alone is worth that!) Contact me off list if you have any questions or you want to add one to your list of toys from ClueCon. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_radius_cdr - unwanted connections to at&t ip
by Stefan Kainz 14 Aug '14

14 Aug '14

Hello, After a clean install of both, debian and freeswitch ( with mod_radius_cdr enabled ) i noticed a few accounting-packages sent to strange ip's on port 1813. 120.0.0.224 ( at&t ) 32.45.27.224 ( chinaunicom ) i'm pretty sure those packages originate from mod_radius_cdr. since we're not related with these companys in any way, i'm wondering why mod_radius_cdr is ( at least trying ) to sent cdr's to those ip's. here is a sample of a cdr sent to at&t: ------------------------------ /U *xxx*.*xxx*.*xxx*.*xxx*:59323 -> *120.0.0.224*:1813// //....r...`/..I....I..(.....,&0670776c-2388-11e4-aef1-8b582cc877c8......l...*xxxxxxxxxxx*....l.......l...*xxxxxxx*....l.............l.......l...default....l...*xxxxxxxxxxxx*....l...."..l...src/switch_ivr_originate.c.'..l..!2014-08-14T09:53:05.848983+0200......).......O..f/ ---------------------------- I also searched the source files of mod_radius_cdr, but sadly ( or luckily ) no match. Somebody cares to look into this? If i'm in the wrong place for this question, maybe some of you can point me in the right direction ;) have a nice day, regards,

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CDR XMLs don't get generated sometimes
by Siri MM 14 Aug '14

14 Aug '14

Hi, I am facing an unusual problem off late. My setup is as follows: I use my webserver with SQL DB, and have setup the url in xml_cdr.conf.xml for posting xml cdr information. It works all well in a normal scenario - xml cdrs are generated, and http post onto my url is also called. Now, when I try to add a new column to my database (which is by the way very large), and then restart freeswitch, I see that the cdr xmls are not getting created - http post does occur though. I don't see any CDR related error messages in the freeswitch logs, either during start up, or after a call - any pointers on how I could debug why xml cdrs are not created?

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Freetdm libsng_isdn and Redirecting Number in SETUP - no rdnis
by Adam Kuśmirek 14 Aug '14

14 Aug '14

Hi All I have FS with libsng and Sangoma A108DE conected to PBX that is redirecting calls A -> B to C. Received SETUP message: 2014-08-13 15:33:21.148143 [DEBUG] ftmod_sangoma_isdn_trace.c:224 [SNGISDN Q931] wp2 FRAME INCOMING: Prot Disc:Q.931/I.451 (0x08) Call Ref:007D (Originating side) Type:SETUP (0x5) Sending complete: Bearer Capability:Coding:ITU-T(0) TransferCap:Speech(0) TransferRate:64 Kbit/s(16) L1Prot:G.711 A-Law(3) Channel Id:No:31 Type:B-chans(3) Exclusive/Implicit Calling Party Number:69214XXXX(l:9) plan:isdn(1) type:national(2)scr:user, passed(1) pres:allowed(0) Called Party Number:77544XXXX(l:9) plan:isdn(1) type:unknown(0) Redirecting Number:773XXXX(l:7) plan:isdn(1) type:national(2)scr:user, not screened(0) pres:allowed(0)reason:Call forwarding unconditional(15) I need Redirecting Number to make correct dialplan decisions, but I cannot get it from channel variables. I hoped, that rdnis channel variable will contain this number, but info application tells nothing about it. Is there any way to get acces to this Redirecting Number information in dialplan ? FreeSWITCH Version 1.2.24+git~20140617T204847Z~9dde4b1548~64bit (git 9dde4b1 2014-06-17 20:48:47Z 64bit) libsng_isdn-7.29.0.x86_64 Thank you for any help Adam

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Callgroup intercept when using multiple FS Servers
by Errol Samuels 13 Aug '14

13 Aug '14

Hello, I have a multi-tenant PBX setup with 2+ Freeswitch 1.4.x servers sharing the same PostgreSQL 9.3.x DB behind Kamailio for Round Robin LB generally working great however, there is an issue where Callgroup intercep <https://wiki.freeswitch.org/wiki/Callgroup_intercept>t '*8' feature only works if the call is on the same server that I am registered to. For example, if I am registered to Freeswitch-1 and the inbound call I am trying to pickup via *8 is on Freeswitch-2 then it doesn't work. Has anyone got any ideas or experience with this scenario? regards, Errol

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multiple sip gateway configuration
by Louie Liu 13 Aug '14

13 Aug '14

Hi, There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions: 1. Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that's case, what do I need to change in the sip profile external.xml file? 2. What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? 3. What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good. External.xml <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="1.2.3.4"/> <param name="ext-sip-ip" value="1.2.3.4"/> External/sip_provider_A.xml <gateway name="Sip_Provider_A"> <param name="username" value="xxx"/> <param name="password" value="xxx"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> External/sip_provider_B.xml <gateway name="Sip_Provider_B"> <param name="username" value="aaa"/> <param name="password" value="bbb"/> <param name="from-user" value="03xxxxxxx"/> <param name="from-domain" value="x.y.z"/> <param name="auth-username" value="yyyyy"/> <param name="realm" value="bwas02.voip.izzz.zz.au"/> <param name="proxy" value="sipconnect.voip.zzzz.zz.au"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> Cheers, Louie

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play back and moh
by 刘宁 13 Aug '14

13 Aug '14

Hi all Sorry to trouble you ,but I really came across a problem still remaining unsolved when using freeswitch to develop some features. Assume that Bob calls Alice, before Bob here the ringtone of Alice,may be a clip of music. When bob use voip phone, after he dials, maybe there will be a bit longer time before he receives the ringtone. I want to play some music in this period. Here is the diaplan config The path is :/usr/local/freeswitch/conf/dialplan/default/call.xml Here is the content: </extension> <extension name="testlua"> <condition field="destination_number" expression="^(\d{4}\d+)$"> <action application="playback" data="voicemail/qav_video_incoming.wav"/>  <action application="lua" data="gen_dialplan_xml.lua"/> </condition> </extension> </include> The lua scripts “gen_dialplan_xml.lua" have to set some arguments for the session and finally it calls freeswitch.bridge(session,newsession) Assume that the time when entering the bridge function is t1, and the time when the ringtone of Alice arrives is t2, thus a time of (t1— t2) exists. I want to play a music to Bob during this time. But the action of “playback” and “lua” seem to be applied sequentially, that means ,the music is not played during the (t1—t2)time. I also tried <extension name="testlua"> <condition field="destination_number" expression="^(\d{4}\d+)$">   <action application="set" data="hold_music=${sound_prefix}/voicemail/qav_video_incoming.wav"/>  <action application="set" data="tmp_hold_music=${sound_prefix}/voicemail/qav_video_incoming.wav"/>  <action application="lua" data="gen_dialplan_xml.lua"/> </condition> </extension> </include> And the music file does exist, but it still did not work. How could I complete this feature?Do I have to complete this in the lua scripts? Than you in advance,looking forward to your reply.

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