Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines

What do you think I shall do? Michael Jerris wrote: > > Looking closer, it looks like it ran the dialplan, executed some > actions to set vars and ran out of actions to do so it hung up. > > On Jul 3, 2009, at 1:25 AM, Edmar Cruz wrote: > >> >> The same issue... >> param name="enable-3pcc"

Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines

Looking closer, it looks like it ran the dialplan, executed some actions to set vars and ran out of actions to do so it hung up. On Jul 3, 2009, at 1:25 AM, Edmar Cruz wrote: > > The same issue... > param name="enable-3pcc" value="true"/> > > Michael Jerris wrote: >> >> Try enabling 3pcc in th

Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines

The same issue... Michael Jerris wrote: > > Try enabling 3pcc in the sip profile. > > On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: > >> >> I have a GSM gateway. The issue is sometimes the calls failed what >> is the >> cause of the error this is my logs? >> >> This is on my freeswitch lo

Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines

Try enabling 3pcc in the sip profile. On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: > > I have a GSM gateway. The issue is sometimes the calls failed what > is the > cause of the error this is my logs? > > This is on my freeswitch logs... > > > 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >

[Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines

I have a GSM gateway. The issue is sometimes the calls failed what is the cause of the error this is my logs? This is on my freeswitch logs... 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 sofia_glue_tech_set_codec() Set Codec sofia/internal/1...@116.541.23.11 PCMU/8000 20 ms 160 samples 2009-06-

Re: [Freeswitch-users] Language Handling: call for assistance

Guys, I don't know if I really get the problem here. I mean, I do get that the 2+2 model does not work not even for where I live. I really hate the fact that all spanish south american dialects (some within the same country) are put in the same bag as it wouldn't matter to ppl so I am with you Stev

[Freeswitch-users] BOUNTY - extended RTP stats - RFC3611

howdy all.. ive added some extra $$$ to this existing ( seemingly closed ) bounty http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support http://jira.freeswitch.org/browse/BOUNTY-4 it would be sweet if a) a Jira admin would re-open the bo

Re: [Freeswitch-users] Language Handling: call for assistance

On Jul 2, 2009, at 7:50 PM, Steve Underwood wrote: >> > If by "the usual way" you mean the standard 2 + 2 letter codes we are > used to on computers, that just doesn't work. As I said before, those > are for written languages, not spoken languages. There are no standard > codes for many spoken la

Re: [Freeswitch-users] Language Handling: call for assistance

Raul Fragoso wrote: > On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote: > >> Michael Collins wrote: >> >>> On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves >> > wrote: >>> >>> Hi, >>> >>> >>> Michael Collins wrote: >>> Hello all!

Re: [Freeswitch-users] How to remove the IP from the SIP caller id number

For that you now have to attend cluecon! ;) /b On Jul 2, 2009, at 2:26 PM, Mitchel Constantin wrote: > Thanks Brian, that worked like a charm. :) ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mai

Re: [Freeswitch-users] How to remove the IP from the SIP caller id number

Thanks Brian, that worked like a charm. :) On Thu, Jul 2, 2009 at 6:37 AM, Brian West wrote: > set the variable sip_invite_domain > > /b > > On Jul 2, 2009, at 12:56 AM, Jason White wrote: > > > I suspect the other end (whatever device you are calling from > > FreeSWITCH) is > > adding the IP add

[Freeswitch-users] seeking for a freeswitch expert

Hi, We are looking to deploy a secured sip environment whereas all traffic, signalling and rtp encrypted etc. according to the rfc's. We are looking for an expert with deep understanding in both fields the freeSwitch and crypto world so to config and compile this for us. you may contact me at tz

Re: [Freeswitch-users] Language Handling: call for assistance

On Thu, 2009-07-02 at 15:47 -0300, Raul Fragoso wrote: > On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote: > > Michael Collins wrote: > > > > > > > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves > > > wrote: > > > > > > Hi, > > > > > > > > > Michael Collins wr

Re: [Freeswitch-users] Language Handling: call for assistance

On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote: > Michael Collins wrote: > > > > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves > > wrote: > > > > Hi, > > > > > > Michael Collins wrote: > >> Hello all! > >> > >> There's been some discussion lately on

Re: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?

Hello Brian, ok, I got it. Any other idea why the UDP port is closed after the TLS packet? Best regards Peter Brian West schrieb: > If its TLS you don't need options packets in the first place. Your > client should do the keep alive NOT FreeSWITCH. TLS is over TCP and > Options over UDP...

Re: [Freeswitch-users] SIP re-invite / bypass_media

no problem On Thu, Jul 2, 2009 at 12:27 PM, Phillip Jones wrote: > Used: > > session.execute("set","bypass_media_after_bridge=true"); > in the confirm.js script and that works perfectly! > > Thank you for you help! > On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale < > anthony.miness...@gmail.

Re: [Freeswitch-users] Language Handling: call for assistance

sound prefix should not be used for lang just "language" the sound_prefix is automatically built as ${base}/sounds/${language}/ each time you execute say. and restored to its previous val when the say is over. On Thu, Jul 2, 2009 at 12:07 PM, Michael Collins wrote: > > > On Thu, Jul 2, 200

Re: [Freeswitch-users] SIP re-invite / bypass_media

Used: session.execute("set","bypass_media_after_bridge=true"); in the confirm.js script and that works perfectly! Thank you for you help! On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale < anthony.miness...@gmail.com> wrote: > try setting bypass_media_after_bridge=true on the session in your co

Re: [Freeswitch-users] Language Handling: call for assistance

Michael Collins wrote: > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves > wrote: > > Hi, > > > Michael Collins wrote: >> Hello all! >> >> There's been some discussion lately on how to handle multiple >> languages, specifically with the *say* application. W

Re: [Freeswitch-users] Language Handling: call for assistance

On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves wrote: > Hi, > > Michael Collins wrote: > > Hello all! > > There's been some discussion lately on how to handle multiple languages, > specifically with the *say* application. We would like some input from the > community on how to handle multiple langua

Re: [Freeswitch-users] SIP re-invite / bypass_media

try setting bypass_media_after_bridge=true on the session in your confirm script On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote: > Thanks for responding and for your help. > > The xml and confirm.js are attached below. Basically trying to bypass_media > after the leg B presses 1 to accept

Re: [Freeswitch-users] SIP re-invite / bypass_media

Thanks for responding and for your help. The xml and confirm.js are attached below. Basically trying to bypass_media after the leg B presses 1 to accept the call. I tried, using bypass_media_after_bridge=true, but the re-invite appears to be done before the confirm.js, So the media is successfull

Re: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention

o it and is ready / Online now i need to call / recev call from / to GTalk (Jingle) cool anyone give me a sample Dialplan Extention? thanks __ Information from ESET NOD32 Antivirus, version of virus signature database 4209 (20090702) __ The message was checked by ESET NOD32 Ant

Re: [Freeswitch-users] SIP re-invite / bypass_media

I would need to know more details about what you are doing. you could set the variable bypass_media_after_bridge=true on the a leg before you call the b leg and use the group_confirm feature to get the caller to press the key. On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote: > Thanks for

[Freeswitch-users] Baby Update!

FreeSWITCHers, Kaiden Anthony Chandler will arrive sometime Friday July 3rd 2009!!! So to help out with any last minute expenses and help ease things up for Ray and Samantha and remove some of the worry I'm going to donate $100 dollars myself to the cause... never know diapers and v

Re: [Freeswitch-users] SIP re-invite / bypass_media

Thanks for that. That seems to successfully re-invite and re-route the the B leg - but does not reinvite the A leg and then immediately issues a "bye" on both legs. Do I have to do something to reinvite that A leg? On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < anthony.miness...@gmail.com>

Re: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention

__ Information from ESET NOD32 Antivirus, version of virus > signature database 4209 (20090702) __ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > ___ > Freeswitch-us

Re: [Freeswitch-users] How to remove the IP from the SIP caller id number

set the variable sip_invite_domain /b On Jul 2, 2009, at 12:56 AM, Jason White wrote: > I suspect the other end (whatever device you are calling from > FreeSWITCH) is > adding the IP address to the caller id. However, I am no SIP expert > and may be > wrong, but you can confirm this by doing

Re: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?

If its TLS you don't need options packets in the first place. Your client should do the keep alive NOT FreeSWITCH. TLS is over TCP and Options over UDP... doesn't make much sense. /b On Jul 2, 2009, at 6:11 AM, Peter P GMX wrote: > Hello, > > I have the following problem: Every call stops

[Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention

, version of virus signature database 4209 (20090702) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo

Re: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?

Some additions: TLS/RTP instead of SRTP does also not work. There are no logs on the debug console except the message that the call is being terminated 2009-07-02 12:06:45.252177 [DEBUG] sofia.c:3100 Channel sofia/internal/835...@sip.mydomain.de entering state [terminating][0] and later cause: NOR

[Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS?

Hello, I have the following problem: Every call stops after 30 seconds when TLS is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. The phones are behind NAT. So I expect, that every 30 seconds an Options request is sent. Wiresharking the traffic I can see * that there are ongoing UDP p

[Freeswitch-users] Connecting to FS through text or SMS?

Hi everyone, Can you give me some proper instructions what I will do to enable texting or SMS through freeswitch using a GSM gateway? Thanks for your cooperation, Edmar -- View this message in context: http://www.nabble.com/Connecting-to-FS-through-text-or-SMS--tp24304261p24304261.ht

Re: [Freeswitch-users] Language Handling: call for assistance

Hi, Michael Collins wrote: > Hello all! > > There's been some discussion lately on how to handle multiple > languages, specifically with the *say* application. We would like some > input from the community on how to handle multiple languages and sound > files. Anthony notes that the say applicatio