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Call dropping after 32 seconds
by Rutu Patel 29 Feb '16

29 Feb '16

Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>

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Question on FreeSWITCH-1.7.0-0a024c4ecb-64bit and service startup
by Shawn Wheeler 29 Feb '16

29 Feb '16

Parameters Virtual Machine running Windows Server 2012 R2 base machine. I have installed FreeSWITCH-1.7.0-0a024c4ecb-64bit I have run into two issues as noted below. Any help would be most appreciated. Item 1 When I tried to start the service I received the following error. “Windows could not start the FreeSWITCH Multi Protocol Switch service on Local Computer Error 1053: The service did not respond to start or control request in a timely fashion” Item 2 I also received this error when trying to start the CLI “The program can't stat because VCRUNTIME140.dll is missing from your computer. Try reinstalling the program to fix this problem.” I assume it was due to the service not starting. Thank you all in advance. Shawn

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ESL Client --> Multiple FS Servers
by SamyGo 29 Feb '16

29 Feb '16

Hi All, I'm looking for some guidance on connecting my ESL client program to multiple FS Servers at once. I've written a small code which connects to one FS ESL and based on events it generates some reports and also send back some actions to FS. Now I'm looking to expand that form one to multiple FS servers but the code should collect all events from every FS and be able to send action to any of them whatever the logic be. Any guides or hints will be much appreciated. Regards, Sammy

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Re: [Freeswitch-users] Call dropping after 32 seconds
by Ahmed Habiba 29 Feb '16

29 Feb '16

Hi, Where is the firewall, is it between user and freeswitch or freeswitch and opensips All cases freeswitch should send the firewall IP for the endpoint behind firewall so that the endpoint would be able to communicate back to freeswitch correct external IP. Thanks, Ahmed Habiba. From: Rutu Patel <rutu.patel(a)inextrix.com <mailto:rutu.patel@inextrix.com>> Subject: Re: [Freeswitch-users] Call dropping after 32 seconds Date: February 29, 2016 at 4:15:28 PM GMT+3 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Hi Jurijs, We have to consider the setup with freeswitch and opensips only. registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) And there is a firewall and we have added x.x.x.3 IP there. Is it possible there is some network related issue Or any sip profile parameter we need to set? -- Thanks, Rutu Patel On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo(a)gmail.com <mailto:jurij.ivo@gmail.com>> wrote: Hi, 1) You have very complex set-up and I doubt that you need it. 2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP. 3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty. With kind regards, Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel(a)inextrix.com <mailto:rutu.patel@inextrix.com>>: Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look? Thanks. -- Thanks, Rutu Patel <http://www.inextrix.com/> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi(a)avimarcus.net <mailto:avi@avimarcus.net>> wrote: 5 second response: 32 seconds is a timer/[network/NAT] issue. You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call. -Avi Marcus BestFone On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel(a)inextrix.com <mailto:rutu.patel@inextrix.com>> wrote: Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel(a)inextrix.com <mailto:rutu.patel@inextrix.com>>: Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look? Thanks. -- Thanks, Rutu Patel <http://www.inextrix.com/> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi(a)avimarcus.net <mailto:avi@avimarcus.net>> wrote: 5 second response: 32 seconds is a timer/[network/NAT] issue. You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call. -Avi Marcus BestFone On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel(a)inextrix.com <mailto:rutu.patel@inextrix.com>> wrote: Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>

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mod_distributor / gwlist down
by Dominique Jeannerod 29 Feb '16

29 Feb '16

Hello, i'm currently working with mod_distributor, to load balance calls to gateways. The load balancing works well, and it's rather simple to configure, no problem. I'm trying to automatically exclude down gateways from the load balancing, which seems quite simple, too, when reading the mod_distributor doc... but i can't make it work. I'm using freeswitch 1.6.6. I configured a profile internal_trk_1, and 2 gateways (svc-isr-fr-301, and non_existent_gw), with ping option. One of the gateways, called (non_existent_gw) is down, and detected down by sofia When I check the status of the gateways, the result is ok : sofia profile internal_trk_1 gwlist down non_existent_gw When I check with distributor command in the cli, it's also ok : expand distributor I_MUT_SIPGTW ${sofia profile internal_trk_1 gwlist down} svc-isr-fr-301 But I can't make it work inside the dialplan. I tested using the 2 syntaxes listed in https://freeswitch.org/confluence/display/FREESWITCH/mod_distributor : <action application="bridge" data="sofia/gateway/${distributor I_MUT_SIPGTW ${sofia internal_trk_1 gwlist down}}/${destination_number}"/> <action application="bridge" data="sofia/gateway\${expand(distributor I_MUT_SIPGTW \${sofia(internal_trk_1 gwlist down)})}/${destination_number}"/> The dead gateway still gets used by the load balancer. What I am doing wrong ? Best regards Dominique Jeannerod

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Channel Hangup not received for Leg A
by Vishal Sharma 29 Feb '16

29 Feb '16

I used following call string to bridge legB to legA {ignore_early_media=true,originate_timeout=150,legA_UUID=36a59b97-32ea-42f5-9b4e-89e19a015f56,origination_caller_id_number=42994299,client_id_sys=1,client_meta_id_sys=None,ivr_refnum=820000222} [leg_timeout=30]freetdm/outgoing/r/09506360596:_:{ignore_early_media=true,originate_timeout=150,legA_UUID=36a59b97-32ea-42f5-9b4e-89e19a015f56,origination_caller_id_number=42994299,client_id_sys=1,client_meta_id_sys=None,ivr_refnum=820000222} [leg_timeout=30]freetdm/outgoing/r/08601224488 If legA disconnect before any of LegB is bridged, I get Cannel_hangup for both legBs, and channel_hangup_complete for legA but no channel_hangup for legA. I am using 1.6.5 Regards, Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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Getting 503 when starting automatically
by freeswitch.opencode＠spamgourmet.com 28 Feb '16

28 Feb '16

On Thu, 2016-02-25 at 21:26:09 +0300, Bote wrote: > A 503 error, huh? Weird. Check for permissions problems. > My guess is that you are building as "root" so when you first ran it > manually all the databases and log files were created with root as owner. > But when it boots systemd is starting it as root and then dropping > privileges to the freeswitch user as it should. After dropping privileges it > can't access its config files or open ports necessary to function. Thanks for the suggestion. I compiled as a normal user using Arch Linux's makepkg tool and installed using its pacman tool, so I don't think that's the problem. I'm also letting system start freeswitch.service with its default user setting (as root). The weirdest thing is that it does start and connect fine using systemd when make a request after boot to have systemd start it (systemctl start freeswitch or systemctl restart freeswitch). In both cases, systemd is controlling process identity, so I don't think it's a permissions issue. My best guess is there's some kind of race condition at boot and if FreeSWITCH starts too early, it gets a cached but invalid handle of some sort that doesn't allow communicating in or out. This is just a guess though. Thanks, David

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Recording and mono directional webrtc video
by Mario Giammarco 28 Feb '16

28 Feb '16

Hello, I am evaluating freeswitch for this purpose: I need a bidirectional audio call with one direction video only from caller. And I need to record audio and video from caller. I need to use webrtc protocol. I have seen old threads that show that FreeSwitch has support for these things but I would like to be sure that the implementation is now on main tree and it is really supported. Thanks, Mario

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Re: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?
by Brian Chow 28 Feb '16

28 Feb '16

Ok, in that case, what do I do next to figure out why inbound calls go directly to voicemail instead of ringing the extension? On Feb 25, 2016 7:57 AM, "Brian Chow" <brianchow(a)digitaldescent.net> wrote: > Ok, in that case, what do I look at next to see why it goes straight to > voicemail instead of ringing my extension? > On Feb 25, 2016 5:52 AM, "Brian West" <brian(a)freeswitch.org> wrote: > >> Its just the channel name so don't get tripped up by that, its just named >> with some default data about the session. >> >> On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow < >> freeswitch(a)digitaldescent.net> wrote: >> >>> Debug console output: https://pastebin.freeswitch.org/24572 >>> >>> My fqdn only resolves inside the network, but the external-ip is >>> configured in FS to use stun, which reports my external ip. >>> >>> >Is your endpoint registered? How often is it set to reregister? >>> >>> Yes, and the zoiper default registration expiry is 3600 >>> >>> >>> >>> >Does your router have a sip alg? Are the ports opened in your firewall? >>> >>> > ish, no GUI for it, but I did make sure to disable it from the command >>> line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but >>> allow bi-directional with existing states. I do get 2 way audio when >>> calling out, and I also can hear the system forwarding my call in to >>> voicemail; the voicemail message and prompts. I know asterisk is different >>> from FS, but my asterisk experience has proven that at least with asterisk, >>> port forwarding isn't needed. Also, unfortunately, I cannot get a static >>> ip. >>> >>> >>> Thanks! >>> >>> -Brian >>> >>> On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max(a)nysolutions.com> >>> wrote: >>> >>>> It is probably sending call to your extension@domainname, if your >>>> external ip is your domain name then you will see the external ip. >>>> >>>> Is your endpoint registered? How often is it set to reregister? >>>> >>>> Does your router have a sip alg? Are the ports opened in your firewall? >>>> >>>> Very hard to guess without a log of a call with debug enabled. >>>> >>>> sofia global sip trace on >>>> >>>> >>>> >>>> Thanks, >>>> >>>> >>>> >>>> Moishe Grunstein >>>> >>>> Tornado Computer Systems, Inc. >>>> >>>> 212.400.7650 888.IPPBX.US >>>> *Service Request Email: support(a)nysolutions.com >>>> <support(a)nysolutions.com> * >>>> >>>> [image: cid:image001.jpg@01C72F94.9EE45D60] >>>> <http://www.nysolutions.com/> >>>> >>>> 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(a)lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Brian >>>> Chow >>>> *Sent:* Wednesday, February 24, 2016 10:00 PM >>>> *To:* freeswitch-users(a)lists.freeswitch.org >>>> *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to >>>> internal extension with external IP? >>>> >>>> >>>> >>>> Hello all! >>>> >>>> I'm new to freeswitch, so I'm sure this is just a newbie >>>> configuration error. Sorry if it's been answered a million times, my >>>> google searches always just bring up the standard NAT configuration pages. >>>> I've already followed the confluence page on configuring NAT. >>>> >>>> >>>> >>>> My Setup: >>>> >>>> Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 >>>> instance. >>>> >>>> SIP Provider: Flowroute >>>> >>>> >>>> >>>> NAT: extension and freeswitch are on the same network, both of which >>>> are behind NAT and connecting to flowroute. I configured external sip and >>>> rtp to use stun entries. External profile is using $${external_rtp/sip_ip} >>>> for ext-rtp/sip respectively. >>>> >>>> >>>> >>>> I have one DID configured to go directly to my one extension. My >>>> extension can register just fine. My extension can dial out. When I call >>>> my cell, the call connects and I have 2 way audio. When I dial my DID from >>>> my cell, I can see the call hitting the FS server, but instead of ringing >>>> my extension, it goes straight to my extensions voicemail (which I can just >>>> fine). >>>> >>>> >>>> >>>> When I look the at the console, it appears (sorry if this is wrong, I'm >>>> only a day into free switch) that FS is attempting to route the call to my >>>> extension...@ my external ip? >>>> >>>> >>>> >>>> I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] >>>> switch_channel.c:1101 New Channel sofia/internal/<extension>@ >>>> *<externalip>*:46072 >>>> >>>> >>>> >>>> >>>> >>>> Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension> ? >>>> >>>> >>>> >>>> Sofia Status says: >>>> >>>> >>>> >>>> Name Type >>>> Data State >>>> >>>> >>>> ================================================================================================= >>>> >>>> external-ipv6 profile >>>> sip:mod_sofia@[::1]:5080 RUNNING (0) >>>> >>>> external profile >>>> sip:mod_sofia@<external_ip>:5080 RUNNING (0) >>>> >>>> external::<uuid> gateway >>>> sip:<flowrouteid>@sip.flowroute.com REGED >>>> >>>> internal-ipv6 profile >>>> sip:mod_sofia@[::1]:5060 RUNNING (0) >>>> >>>> internal profile >>>> sip:mod_sofia@<internal_ip>:5060 RUNNING (0) >>>> >>>> >>>> ================================================================================================= >>>> >>>> >>>> >>>> If I'm completely off base here, can anyone recommend where I can start >>>> looking to change/troubleshoot the issue? I feel like it's just me missing >>>> something, I just can't determine what that might be. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Brian >>>> >>>> >>>> _________________________________________________________________________ >>>> 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-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-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 >>> >> >> >> >> -- >> >> *Brian West* >> brian(a)freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: >> /r/freeswitch <https://www.reddit.com/r/freeswitch> >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> 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-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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Exception: ReferenceError: File is not defined (near: "var fd = new File(logfile);
by Shabbir abbasi 27 Feb '16

27 Feb '16

i am getting this error in javascript here is my code logToFile.js var logfile = "/usr/local/freeswitch/scripts/test.log"; var fd = new File(logfile); i have defined but error is coming logToFile.js:3 Exception: ReferenceError: File is not defined (near: "var fd = new File(logfile);") Dont know why ?

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