August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Simpl5[WebRTC] to PSTN codec changed during a call
by Adrian Andrei 25 Aug '14

25 Aug '14

Hello, In my follwing setup Sipml5 -> FS -> Cisco ->PSTN the codec is changed during the call. At the first glance the call evolution is: 1. Sipml5 Client codec pref is OPUS,ISAC, PCMU, PCMA , Cisco voice class codec PCMU, PCMA (g711u&a). After negociation the PCMU codec is set. 2014-08-25 19:03:05.412802 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2014-08-25 19:03:05.412802 [DEBUG] switch_core_media.c:3530 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match The call is set up, voice goes well in both ways. 2. After a while(after receiving the next SDP packet from WebRtc Client) it seems that it finds a new codec match in OPUS. 2014-08-25 19:04:11.292795 [DEBUG] switch_core_media.c:3475 Audio Codec Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1] 2014-08-25 19:04:11.292795 [DEBUG] switch_core_media.c:3530 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match FS will change the codec and the call will end. 2014-08-25 19:04:11.292795 [DEBUG] switch_core_media.c:2260 Changing Codec from PCMU@20ms@8000hz to opus@20ms@48000hz 2014-08-25 19:04:11.332798 [DEBUG] switch_core_media.c:2352 Set Codec sofia/internal/xxxx(a)5.6.7.8 opus/48000 20 ms 960 samples 0 bits 1 channels 2014-08-25 19:04:11.332798 [DEBUG] switch_core_codec.c:123 sofia/internal/ xxxx(a)5.6.7.8 Original read codec replaced with opus:116 2014-08-25 19:04:11.352807 [NOTICE] switch_core_io.c:1257 Activating write resampler 2014-08-25 19:04:11.752801 [NOTICE] switch_rtp.c:1144 Auto Changing stun/rtp/dtls port from 1.2.3.4:50594 to 1.2.3.4:19602 2014-08-25 19:04:12.072801 [ERR] mod_opus.c:402 Decoder Error: corrupted stream! 2014-08-25 19:04:12.072801 [ERR] switch_core_io.c:1310 Codec OPUS (STANDARD) decoder error! 2014-08-25 19:04:12.072801 [DEBUG] switch_ivr_bridge.c:579 sofia/external/ 5555555(a)9.9.9.9 ending bridge by request from write function I tried to force codec on both legs by adding: <action application="set" data="absolute_codec_string=PCMU,PCMA"/> but nothing change. Also inbound-late-negotiation is set to true in sofia-profiles. Any ideea how to fix this issue? Here is the full log: http://pastebin.freeswitch.org/23161 Thank you

1 0

mod_lua
by Joel White 25 Aug '14

25 Aug '14

Just a quick question..... How many Lua / Bindings can you define in a FreeSWITCH instance? e.x. dialplan, directory, etc...

1 0

LUA
by Joel White 25 Aug '14

25 Aug '14

I am trying to get Lua generation of the Directory so I can also tie this in with auto provisioning. I have created the table in the Postgres DB and have the lua script specified in the lua module config But I keep getting this error 2014-08-18 13:07:11.869818 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/gen_dir_user_xml.lua:5: attempt to index global 'params' (a nil value) I am sure after playing with it for a while I will get everything working properly, but would anyone have some information to point me in the right direction? Thank you in advance

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Registration Failed with status Method Not Allowed [405]
by Ave ebock 25 Aug '14

25 Aug '14

Hi, Please im trying to register a gateway that uses IP Authentication. I've added the IP address of my freeswitch to my service provider and i have this in my /sip_profiles/external/vb.xml <include> <gateway name="vb"> <param name="register" value="false"/> <param name="username" value="username"/> <param name="password" value="password"/> <param name="proxy" value="108.59.2.133"/> <param name="realm" value="108.59.2.133"/> <param name="expire-seconds" value="600"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> when i do sofia profile external register vb, i get an error vb Registration Failed with status Method Not Allowed [405] Please what might i be doing wrong I have other providers that do not use IP authentication and i've got no issue with them. Thank you

2 1

Fwd: mod_python always gives an error at first dialplan call
by Nuno Reis 25 Aug '14

25 Aug '14

Hi guys. I know I've already asked this on the freeswitch-dev mailing list but since i still didn't get an answer, i'm sending it again, this time to the freeswitch-users ML too. I'm experiencing a weird behavior with the new FS 1.4. Everytime i call a python script in the dialplan after a reboot i always got an error similar to this: 2014-08-18 07:03:26.341145 [ERR] mod_python.c:293 Error calling python script 2014-08-18 07:03:26.341145 [ERR] mod_python.c:164 Python Error by calling script "stable.disa": <type 'exceptions.AttributeError'> Message: 'PySwigObject' object has no attribute 'sleep' This only happens with the first call after a restart. After that everything works fine. I never saw this in FS 1.2 This seem to happen only the first time mod_python is invoked and *session* object is used, no mater the script.The other python scripts will work after one (anyone) is used. As said before, after *session* object is accessed once, after a restart (giving back the error), everything starts working again. Any ideas on why? Thanks. *Note:* I'm using FS Version 1.4.8 64bit -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> ---------- Forwarded message ---------- From: Nuno Reis <nreis(a)wavecom.pt> Date: Mon, Aug 18, 2014 at 7:12 AM Subject: mod_python always gives an error at first dialplan call To: freeswitch-dev(a)lists.freeswitch.org Hi guys. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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(no subject)
by Denis Papes 25 Aug '14

25 Aug '14

Hi, I installed clean Debian 7 and executed remote installation using RockSolidPBX. Installation failed to change DHCP assigned address to static, so I did it manually. After that, FreeSWITCH installation passed flawlessly. After successfully creating user with extension 100, I created dialplan that should ring extension 100 when dialing number 123, but XML file it produced had an error: <include> <extension name="123"> <condition field="destination_number" expression="^(123)$"> <action application="export" data="dialed_extension=$1"/> .............. <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> </condition> </extension> </include> It does not ring extension 100, but 123. Also, it would be great if it could create custom SIP gateways. Denis

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Re: [Freeswitch-users] freeswitch stream remote audio file
by Snabel Kabiya 25 Aug '14

25 Aug '14

Hi, I used this encoding: URLEncoder.encode(url_parameters, "UTF-8"); to encode the URL parameters. here is the url i got: URL = http://storage.googleapis.com/myBucket/file.wav <http://storage.googleapis.com/myBucket/file.wav>? GoogleAccessId%3D703391026438d3tc85gd1jiu6f32mckumln3b2gc5qhk% 40developer.gserviceaccount.com%26 Expires%3D1408970638639%26 Signature%3DF3zM%2Fkr2mKqulgrdp1F6Q9Gdf5rKvy7GHrWOW5GMU xxxxxxxxxxxXxxxxxxXXXXXXXXjXtFFRBJp4a%2BfhrQJcrs4p%2FZhsSdB kyet8qVwf%2BRUN8I0ROA%2FNpJ%2B0T0eEBqP2PNFPHzI8GKQWm7wI6mmZDwKtgUT3A%2BVXGV74nUqHrNY%3D But now i get this exception: *2014-08-25 15:53:35.138408 [ERR] mod_httapi.c:2723 Error checking file cache (check permissions)* that's all i see in the logs...is this related to the file name size? is there a limitation or is it something else? Thanks, Snabel ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 25 Aug 2014 07:07:57 -0500 > Subject: Re: [Freeswitch-users] freeswitch stream remote audio file > Yes you should URL encode that. @ would become %40 > > > On Mon, Aug 25, 2014 at 2:02 AM, Snabel Kabiya <snabel(a)lexifone.com> > wrote: > >> Hi, >> >> i tried removing each parameter to see which one is causing the error. >> when i removed this parameter: >> >> GoogleAccessId=xxx-xxx(a)developer.gserviceaccount.com >> <http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…> >> >> freeswitch was able do download and play the file. are there characters >> in this parameter which are causing session:streamFile() failure? >> should i escape some characters? if so which and how? >> >> this parameter is mandatory so i can't just remove it from the link. any >> ideas how to solve this? >> >> Thanks, >> Snabel >> >> ---------- Forwarded message ---------- >> >>> From: Snabel Kabiya <snabel(a)lexifone.com> >>> To: freeswitch-users(a)lists.freeswitch.org >>> Cc: >>> Date: Mon, 25 Aug 2014 09:41:52 +0300 >>> Subject: Re: [Freeswitch-users] freeswitch stream remote audio file >>> Hi, >>> >>> i generated the link using the instructions here: >>> >>> https://developers.google.com/storage/docs/accesscontrol#Signed-URLs >>> >>> http://stackoverflow.com/questions/17508285/gae-application-needing-to-crea… >>> >>> But the link i generated is valid, I paste the signed URL into a web >>> browser and i downloaded the file. >>> that's why i think it's maybe a special characters issues. some >>> characters are causing the >>> session:streamFile() method to return this error. >>> >>> Thanks, >>> Snabel >>> >>> >>> //////////////////////////////////////////////////////////////////////////////////////// >>> >>> Hi, >>> >>> I'm saving the remote audio files on google cloud storage. i want to >>> play these files in freeswitch. >>> >>> when i use: mediaLink = "http://storage.googleapis.com/myBucket/file.wav"; >>> >>> >>> session:streamFile(mediaLink); >>> >>> *it works great.* >>> >>> But when i use signed urls >>> >>> mediaLink = " >>> http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…"; >>> session:streamFile(mediaLink); >>> >>> *i get this error:* >>> >>> *2014-08-24 20:42:48.770818 [ERR] mod_httapi.c:2696 File at url >>> [http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…] >>> <http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…]> is >>> unreachable!* >>> >>> >>> is there a special characters issue in the link? >>> >>> >>> Thanks, >>> Snabel >>> >>> ---------- Forwarded message ---------- >>> >>>> From: Yosi Taguri <yosi(a)taguri.com> >>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>> Cc: >>>> Date: Sun, 24 Aug 2014 21:29:27 +0300 >>>> Subject: Re: [Freeswitch-users] freeswitch stream remote audio file >>>> how did you generate the link? >>>> >>>> >>>> On Sun, Aug 24, 2014 at 9:00 PM, Snabel Kabiya <snabel(a)lexifone.com> >>>> wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm saving the remote audio files on google cloud storage. i want to >>>>> play these files in freeswitch. >>>>> >>>>> when i use: mediaLink = " >>>>> http://storage.googleapis.com/myBucket/file.wav"; >>>>> >>>>> session:streamFile(mediaLink); >>>>> >>>>> *it works great.* >>>>> >>>>> But when i use signed urls >>>>> >>>>> mediaLink = " >>>>> http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…"; >>>>> session:streamFile(mediaLink); >>>>> >>>>> *i get this error:* >>>>> >>>>> *2014-08-24 20:42:48.770818 [ERR] mod_httapi.c:2696 File at url >>>>> [http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…] >>>>> <http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…]> is >>>>> unreachable!* >>>>> >>>>> >>>>> is there a special characters issue in the link? >>>>> >>>>> >>>>> Thanks, >>>>> >>>>> Snabel >>>>> >>>>

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Re: [Freeswitch-users] freeswitch stream remote audio file
by Snabel Kabiya 25 Aug '14

25 Aug '14

Hi, i tried removing each parameter to see which one is causing the error. when i removed this parameter: GoogleAccessId=xxx-xxx(a)developer.gserviceaccount.com <http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…> freeswitch was able do download and play the file. are there characters in this parameter which are causing session:streamFile() failure? should i escape some characters? if so which and how? this parameter is mandatory so i can't just remove it from the link. any ideas how to solve this? Thanks, Snabel On Mon, Aug 25, 2014 at 9:42 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: freeswitch stream remote audio file (Snabel Kabiya) > > > ---------- Forwarded message ---------- > From: Snabel Kabiya <snabel(a)lexifone.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Mon, 25 Aug 2014 09:41:52 +0300 > Subject: Re: [Freeswitch-users] freeswitch stream remote audio file > Hi, > > i generated the link using the instructions here: > > https://developers.google.com/storage/docs/accesscontrol#Signed-URLs > > http://stackoverflow.com/questions/17508285/gae-application-needing-to-crea… > > But the link i generated is valid, I paste the signed URL into a web > browser and i downloaded the file. > that's why i think it's maybe a special characters issues. some characters > are causing the > session:streamFile() method to return this error. > > Thanks, > Snabel > > > //////////////////////////////////////////////////////////////////////////////////////// > > Hi, > > I'm saving the remote audio files on google cloud storage. i want to play > these files in freeswitch. > > when i use: mediaLink = "http://storage.googleapis.com/myBucket/file.wav"; > > > session:streamFile(mediaLink); > > *it works great.* > > But when i use signed urls > > mediaLink = " > http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…"; > session:streamFile(mediaLink); > > *i get this error:* > > *2014-08-24 20:42:48.770818 [ERR] mod_httapi.c:2696 File at url > [http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…] > <http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…]> is > unreachable!* > > > is there a special characters issue in the link? > > > Thanks, > Snabel > > > > On Sun, Aug 24, 2014 at 11:51 PM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Changing display name in From header ? (Du?an Dragi?) >> 2. Re: freeswitch stream remote audio file (Yosi Taguri) >> 3. Re: Problem Registering Gateway (Ave ebock) >> >> >> ---------- Forwarded message ---------- >> From: "Dušan Dragić" <dragic.dusan(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Sun, 24 Aug 2014 21:48:56 +0200 >> Subject: Re: [Freeswitch-users] Changing display name in From header ? >> Try with sip_from_display and sip_invite_from_uri instead of >> sip_from_uri, something like: >> sip_from_display=Anonymous >> sip_invite_from_uri=<sip:anonymous@anonymous.invalid> >> origination_caller_id_name=9094445555 >> origination_caller_id_number=9094445555 >> >> On 24 August 2014 19:02, Tim Smith <gb10hkzo-fs1(a)yahoo.co.uk> wrote: >> > Hi, >> > >> > What magic needs to be done to change the display name in the From >> header ? >> > >> > I've tried outbound_caller_name, effective_caller_id_name and all sorts >> of other combinations under the sun and yet I can't get the Freeswitch to >> comply !!!! >> > >> > Here's what the relevant section of my XML dial plan looks like: >> > >> > >> > <condition field="${privacy}" expression="yes" break="never"> >> > <action application="export" >> data="outbound_caller_name=Anonymous"/> >> > <action application="export" >> data="sip_from_uri=<sip:anonymous@anonymous.invalid>"/> >> > <action application="export" data="sip_cid_type=pid"/> >> > <action application="export" >> data="origination_caller_id_name=${outbound_caller_id_number}"/> >> > <action application="export" >> data="origination_caller_id_number=${outbound_caller_id_number}"/> >> > <anti-action application="set" data="sip_cid_type=none"/> >> > <anti-action application="set" >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > <anti-action application="set" >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > </condition> >> > >> > >> > Basically, I need the From header to look like : >> > From: "Anonymous" <sip:anonymous@anonymous.invalid> >> > >> > and PID to look like : >> > >> > P-Asserted-Identity: "9094445555" <sip:9094445555@1.2.3.4> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> > http://www.cudatel.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 >> >> >> >> -- >> Dušan Dragić >> >> >> >> >> ---------- Forwarded message ---------- >> From: Yosi Taguri <yosi(a)taguri.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Sun, 24 Aug 2014 21:29:27 +0300 >> Subject: Re: [Freeswitch-users] freeswitch stream remote audio file >> how did you generate the link? >> >> >> On Sun, Aug 24, 2014 at 9:00 PM, Snabel Kabiya <snabel(a)lexifone.com> >> wrote: >> >>> Hi, >>> >>> I'm saving the remote audio files on google cloud storage. i want to >>> play these files in freeswitch. >>> >>> when i use: mediaLink = "http://storage.googleapis.com/myBucket/file.wav"; >>> >>> >>> session:streamFile(mediaLink); >>> >>> *it works great.* >>> >>> But when i use signed urls >>> >>> mediaLink = " >>> http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…"; >>> session:streamFile(mediaLink); >>> >>> *i get this error:* >>> >>> *2014-08-24 20:42:48.770818 [ERR] mod_httapi.c:2696 File at url >>> [http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…] >>> <http://storage.googleapis.com/myBucket/file.wav?GoogleAccessId=xxx-xxx@deve…]> is >>> unreachable!* >>> >>> >>> is there a special characters issue in the link? >>> >>> >>> Thanks, >>> >>> Snabel >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.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 >>> >> >> >> >> -- >> >> Yosi Taguri | Founder & CTO | Yalo | http://yaloapp.com >> <http://ahhhpah.com> | Mobile: +972-54-4816020 | Google Voice US Mobile: +1 >> (415) 935-1126 | Email: yosi(a)taguri.com >> >> >> >> >> ---------- Forwarded message ---------- >> From: Ave ebock <babyphatt2k(a)yahoo.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Sun, 24 Aug 2014 13:39:06 -0700 >> Subject: Re: [Freeswitch-users] Problem Registering Gateway >> Hi Steven, >> Removed REGISTER, but im still getting the issue. >> The Folks at the Provider Company havent been of any help either >> >> Thank You >> >> >> On Sunday, August 24, 2014 3:27 PM, Steven Ayre <steveayre(a)gmail.com> >> wrote: >> >> >> Perhaps they don't have the REGISTER method enabled? In SIP you aren't >> required to register to make outgoing calls, only to tell the server where >> you receive incoming ones. >> >> >> On 24 August 2014 01:18, Ave ebock <babyphatt2k(a)yahoo.com> wrote: >> >> Hi, >> Please im trying to register a gateway that uses IP Authentication. I've >> added the IP address of my freeswitch to my service provider and i have >> this in my /sip_profiles/external/vb.xml >> <include> >> <gateway name="vb"> >> <param name="register" value="false"/> >> <param name="username" value="username"/> >> <param name="password" value="password"/> >> <param name="proxy" value="108.59.2.133"/> >> <param name="realm" value="108.59.2.133"/> >> <param name="expire-seconds" value="600"/> >> <param name="retry-seconds" value="30"/> >> <param name="ping" value="25"/> >> </gateway> >> </include> >> >> when i do sofia profile external register vb, i get an error >> >> vb Registration Failed with status Method Not Allowed [405] >> >> Please what might i be doing wrong >> >> I have other providers that do not use IP authentication and i've got no >> issue with them. >> >> Thank you >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.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 >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.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 >> >> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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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Action Transfer Fail to next execution
by bhavik patel 25 Aug '14

25 Aug '14

Hi, We are trying to call one extension. If first call finish with NO_ANSFER after 10 sec ring timeout then we forward call to another extension using Action transfer. If transfer also fail then call will be send to Voicemail but my issue is call is not pass to voicemail after transfer fail. Is there any solution how to pass call on Voicemail once transfer fail. MY dialplan is some thing like below. ================================================================================== <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="demo dynamic"> <context name="default"> <extension name="1212456"> <condition field="destination_number" expression="1212456"> <action application="set" data="effective_destination_number=1212456"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=10"/> <action application="bridge" data="user/TESTHome@${domain_name}"/> <action application="set" data="call_timeout=20"/> <action application="transfer" data="12129873456 XML default"/> <action application="answer"/> <action application="voicemail" data="default $${domain_name} TESTHome"/> </condition> </extension> </context> </section> </document> ============================================================================== -- Thanks, Bhavik Patel

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Freeswitch rpm building
by Mujahid Ali 25 Aug '14

25 Aug '14

i'm building freeswitch rpm from spec file i download freeswitch-1.4.5.tar.bz2 <http://files.freeswitch.org/freeswitch-1.4.5.tar.bz2> from http://files.freeswitch.org/ but during rpm building there is alot of errors like error: Installed (but unpackaged) file(s) found: /etc/freeswitch/autoload_configs/format_cdr.conf.xml /etc/freeswitch/autoload_configs/opus.conf.xml /etc/freeswitch/autoload_configs/spidermonkey.conf.xml /etc/freeswitch/autoload_configs/v8.conf.xml /etc/freeswitch/lang/sv/sv.xml /etc/freeswitch/lang/sv/vm/sounds.xml RPM build errors: Installed (but unpackaged) file(s) found: /etc/freeswitch/autoload_configs/format_cdr.conf.xml /etc/freeswitch/autoload_configs/opus.conf.xml /etc/freeswitch/autoload_configs/spidermonkey.conf.xml /etc/freeswitch/autoload_configs/v8.conf.xml /etc/freeswitch/lang/sv/sv.xml /etc/freeswitch/lang/sv/vm/sounds.xml

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