Re: [Freeswitch-users] voicemail webapi returns 403 Forbidden

It works. Thanks. But when I click the flash "Play", it shows undefined. When I click download, it shows another challenge box where I had to input the same user/pass. Is that a problem? On May 7, 2009, at 8:09 PM, Anthony Minessale wrote: in the inside or On Thu, May 7, 2009 at 5:

Re: [Freeswitch-users] Total noob question

Weird did you modify anything in modules.conf? /b On May 7, 2009, at 9:01 PM, Lars Zeb wrote: Sorry for the hijacking, too anxious. I typed only one of the two commands – both Unknown. 2009-05-07 17:52:52 [CONSOLE] switch_core.c:1322 switch_core_init_and_modload() FreeSWITCH Version 1.0.4

Re: [Freeswitch-users] Total noob question

Sorry for the hijacking, too anxious. I typed only one of the two commands - both Unknown. 2009-05-07 17:52:52 [CONSOLE] switch_core.c:1322 switch_core_init_and_modload() FreeSWITCH Version 1.0.4pre7 (13238M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL

Re: [Freeswitch-users] Total noob question

Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command', but it does > accept 'shutdown'? Maybe the mod_commands module wasn't loaded.

Re: [Freeswitch-users] TLS initialization failures

A quick update: I can still reproduce the profile startup failure under revision 13246, but I haven't hit the segfault again. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch

Re: [Freeswitch-users] Total noob question

Check out the wiki for more info. Wiki.FreeSWITCH.org On Thursday, May 7, 2009, Lars Zeb wrote: > I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. > > Why is it that after I launch freeswitch and type in either 'show' or > 'status' at the console, it responds with 'Unknown command

Re: [Freeswitch-users] Total noob question

Its just "status" Also please do not hijack threads.. Click new message and input the address freeswitch-users@lists.freeswitch.org Thanks, Brian On May 7, 2009, at 8:13 PM, Lars Zeb wrote: I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. Why is it that after I launch frees

[Freeswitch-users] Total noob question

I have installed from the 1.0.4pre7 tarball on a openSuse 11.1. Why is it that after I launch freeswitch and type in either 'show' or 'status' at the console, it responds with 'Unknown command', but it does accept 'shutdown'? Thanks, Lars ___ Freeswit

Re: [Freeswitch-users] Re-2: Ruby and ESL help

Diego Viola wrote: > Hi guys, > > Nevermind with the ESL and EM thing. > > I was wondering what the getBody() getHeader() and other ESL stuff > does behind the scenes, in raw socket, do you know? Why not read the source code? This is free software and open-source, after all. _

Re: [Freeswitch-users] Re-2: Ruby and ESL help

Hi guys, Nevermind with the ESL and EM thing. I was wondering what the getBody() getHeader() and other ESL stuff does behind the scenes, in raw socket, do you know? Thanks, Diego On Thu, May 7, 2009 at 4:44 PM, Diego Viola wrote: > It seems like EM (EventMachine) can't be used with ESL. > > 1

Re: [Freeswitch-users] DTMF recognition flaky

Anthony Minessale wrote: > you may have a sonus infection > > try some of the stuff from here under DTMF > > http://wiki.freeswitch.org/wiki/RTP_Issues Thank you for the suggestion. I tried both the Sonus and Cisco settings in the external profile (running sofia profile external restart reload

Re: [Freeswitch-users] Double Re-Register problem

Peter, Fix your email server I'm tired of my messages to you getting bounced back. Also upload cleaned up configs without all the comments and DO NOT attach tar.gz files. /b ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswit

Re: [Freeswitch-users] Originate - using own uuid

it's up to you what you do, break it you buy it =D On Thu, May 7, 2009 at 4:26 PM, Brian West wrote: > How about something like this : > {origination_caller_id_number=16041234567,originate_timeout=60,origination_uuid=${create_uuid()}}sofia/gateway/icall/16041234567 > &park > > You can call an

Re: [Freeswitch-users] Double Re-Register problem

Also fix your evil mail server cuz I tried tried to email you and it bounced back. You have failed to detail your scenario more... options used .. how you connect things.. every detail counts no matter how small. Then collect it up and put it on Jira. /b On May 7, 2009, at 8:37 AM, Peter

Re: [Freeswitch-users] Originate - using own uuid

How about something like this : {origination_caller_id_number =16041234567,originate_timeout=60,origination_uuid=$ {create_uuid()}}sofia/gateway/icall/16041234567 &park You can call an api with ${apiname()} and the results will go into its place. /b On May 7, 2009, at 4:18 PM, Simon Tang

Re: [Freeswitch-users] Originate - using own uuid

Abcdefg was an example...I was suggesting that I can guarantee uniqueness forever and if I collide, it's my fault and I take the consequences :) Anyway, I will use create_uuid...I was exploring this option before because I thought I couldn't use create_uuid because I made some wrong assumptions

Re: [Freeswitch-users] Originate - using own uuid

Yah kinda gets messy when trying to track down something if the same uuid gets used over and over in a short period of time. :P /b On May 7, 2009, at 4:02 PM, Andrew Thompson wrote: You also need to be sure that FreeSWITCH isn't going to use it at any point in the past, present or future ;)

Re: [Freeswitch-users] Originate - using own uuid

On Thu, May 07, 2009 at 01:47:26PM -0700, Simon Tang wrote: > Hi there, > > I know there's a "feature" that allows us to create a uuid and then use > that uuid to do an originate (api create_uuid). Is there any badness if > I don't do the "create_uuid" and just do the originate using my own > uui

Re: [Freeswitch-users] Originate - using own uuid

I would use real UUID's not something as simple as abcdefg cuz FS will hang up on you if you collide. /b On May 7, 2009, at 3:47 PM, Simon Tang wrote: Hi there, I know there's a "feature" that allows us to create a uuid and then use that uuid to do an originate (api create_uuid). Is ther

[Freeswitch-users] Originate - using own uuid

Hi there, I know there's a "feature" that allows us to create a uuid and then use that uuid to do an originate (api create_uuid). Is there any badness if I don't do the "create_uuid" and just do the originate using my own uuid. I experimented and made the following originate call: api originat

Re: [Freeswitch-users] Re-2: Ruby and ESL help

It seems like EM (EventMachine) can't be used with ESL. 16:41 < diegoviola> thedonvaughn: i see, so ESL itself can't be used with EM? 16:41 < wyhaines> You can't just hand the socket from EM to ESL. 16:42 < thedonvaughn> prolly not 16:42 < thedonvaughn> and if tmm1 doesn't know how, then i'm going

Re: [Freeswitch-users] Re-2: Ruby and ESL help

Ok, this seems to work: require 'rubygems' require 'eventmachine' module CallingCard def post_init send_data "sendmsg\ncall-command: execute\nexecute-app-name: answer\n\n" send_data "sendmsg\ncall-command: execute\nexecute-app-name: playback\nexecute-app-ar

Re: [Freeswitch-users] Amazon EC2 no audio

I had this same problem but eventually overcame it. I modified the docs at http://wiki.freeswitch.org/wiki/Amazon_ec2 accordingly. I think the problem I had was the internal vs. external IP address, as you've alluded to at the bottom of your message. In addition to the changes you've made, I a

Re: [Freeswitch-users] FS + encryption

If I can find funding for travel presently I would. Anthony Minessale wrote: > Hey David! > > You should come by to this year's ClueCon! > We still have some speaking slots left. > > > On Thu, May 7, 2009 at 11:08 AM, David Sugar > wrote: > > SIP TLS will pr

Re: [Freeswitch-users] FS + encryption

Hey David! You should come by to this year's ClueCon! We still have some speaking slots left. On Thu, May 7, 2009 at 11:08 AM, David Sugar wrote: > SIP TLS will protect the SIP session information with static keys via a > certificate, assuming of course the call is direct between two peers. >

Re: [Freeswitch-users] Compact, fanless appliance?

Antonio Gallo wrote: > Alix cases are like 6/9 € from their shop site. I think its easy to find > someone who work with aluminium that can make for you custom boxes for > like like 6/20 € at pcs Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2) allow for a PCI slot, e

Re: [Freeswitch-users] How to install OpenVox PCI card?

On Thu, May 7, 2009 at 9:26 AM, Fred-145 wrote: > > > mercutioviz wrote: > > If I understand correctly the OpenVox card is a zaptel clone, which means > > you'd need to download the drivers, install them and then do the OpenZAP > > install. See the OpenZAP page on the wiki. > > Thanks for the tip

Re: [Freeswitch-users] How to install OpenVox PCI card?

mercutioviz wrote: > If I understand correctly the OpenVox card is a zaptel clone, which means > you'd need to download the drivers, install them and then do the OpenZAP > install. See the OpenZAP page on the wiki. Thanks for the tip. I'll give it a shot. In the mean time, I'm interested in feed

Re: [Freeswitch-users] FS + encryption

SIP TLS will protect the SIP session information with static keys via a certificate, assuming of course the call is direct between two peers. It will do nothing for the actual voice channel. There is SRTP, which can be used to create a cryptographic context over RTP. However, the key question is

Re: [Freeswitch-users] Re-2: Ruby and ESL help

I see, but it should work with ESL too right? Diego On Thu, May 7, 2009 at 7:55 AM, Mikael Aleksander Bjerkeland wrote: > EventMachine is very different to TCPSocket and is definitely not a > drop-in replacement. Take a look at FreeSWITCHeR > (http://code.rubyists.com/projects/fs/repository) and

Re: [Freeswitch-users] How to install OpenVox PCI card?

If I understand correctly the OpenVox card is a zaptel clone, which means you'd need to download the drivers, install them and then do the OpenZAP install. See the OpenZAP page on the wiki. -MC On Thu, May 7, 2009 at 5:46 AM, Fred-145 wrote: > > Hello > > I browsed the wiki + archives of this li

Re: [Freeswitch-users] FS + encryption

You're right. Digital circuits are not so easy to tap into as opposed to POTS. I'm thinking about this issue because I'm wondering how the US Govt. sets their DRSN lines up to be secure. From what I read, only Raytheon and Telecore have DRSN-use JITC-approved switches. It seems that list may be

Re: [Freeswitch-users] FS + encryption

Well its not so easy to take a lineman's handset and eavesdrop on a T1/ BRI/PRI/DSS1 circuit.. that takes way more hardware but POTS you just need a lineman's handset. But yes true secure will need to be end to end. /b On May 7, 2009, at 10:35 AM, Paul wrote: Yes, I've seen this http

[Freeswitch-users] FS + encryption

Yes, I've seen this http://wiki.freeswitch.org/wiki/SIP_TLS. I was just curious if the only way to have true end to end secure communications with FS would have to be a SIP trunk from one FS system to another encrypted SIP system on the other with no POTS/PRI/BRI circuits used in transit. I'm a

Re: [Freeswitch-users] SRTP Error "auth check failed"

Hello Helmut, I also have problems with my Snom300s and Snom320s and G711 and SRTP. They may be related to this problem, but I am not sure. The phones disconnect the media stream after a while (2..10 minutes) because the Snom media port is blocked all of a sudden. I have opened a bug report at Sno

[Freeswitch-users] SRTP Error "auth check failed"

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, today a colleague of mine told me that sometimes calls were disconnected without any obvious reasons. In FS's log I found this: 2009-05-07 15:52:22 [ERR] switch_rtp.c:1656 rtp_common_read() Error: SRTP unprotect failed with code 7 (auth check

Re: [Freeswitch-users] SRTP Error "auth check failed"

Were they on hold for a bit? /b On May 7, 2009, at 10:04 AM, Helmut Kuper wrote: I scanned my recent FS log files for that message and found that this error accours a few times a week. I use Snom Phones all with G722 and SRTP. Any ideas what this could be caused by? Brian West br...@freeswit

Re: [Freeswitch-users] Sip Register Problem

Sounds like the device doesn't challenge correctly. A sip trace would be nice to see. /b On May 7, 2009, at 8:50 AM, Nameer Kazzaz wrote: Hi all, Please if anybody can help me with a problem I'm having. I have 2 gateways Quintum AX and OneAccess 100D both are setup to register with Freeswi

[Freeswitch-users] Sip Register Problem

Hi all, Please if anybody can help me with a problem I'm having. I have 2 gateways Quintum AX and OneAccess 100D both are setup to register with Freeswitch 1.0.4pre7, when I first switch on the gateways all is well and then after about 5 min sip register starts falling with sip 401. I can send wha

Re: [Freeswitch-users] Double Re-Register problem

Looks like they are sending you a bye... not quite sure this is our problem. Can you provide a sip trace in pcap format for me to see? /b On May 7, 2009, at 8:37 AM, Peter P GMX wrote: Hello, I habe the following problem when re-registering to an external SIP provider during a call which r

[Freeswitch-users] Double Re-Register problem

Hello, I habe the following problem when re-registering to an external SIP provider during a call which results in immediate call-hangups. - FS re-registers with nonce - 2ms later FS re-registers without nonce - external SIP provider asks for credentials - FS re-registers with nonce - External pro

[Freeswitch-users] How to install OpenVox PCI card?

Hello I browsed the wiki + archives of this list, but didn't find an article on how to add a TDM card to Freeswitch. If I've overlooked it, thank you for pointing me to it. I have a one-FXO module OpenVox PCI card. It's correctly detected by Linux CentOS 5: # lspci -v 03:00.0 Communica

Re: [Freeswitch-users] rtpDir Freeswitch

To answer the question, "channel drivers" in FreeSWITCH are called "Endpoint Modules" and there is certainly a way to write them we have nearly a dozen in tree. On Thu, May 7, 2009 at 6:44 AM, Karl Vesterling wrote: > > > Roger on the previous post asking the question some time ago. > I don't

Re: [Freeswitch-users] DTMF recognition flaky

you may have a sonus infection try some of the stuff from here under DTMF http://wiki.freeswitch.org/wiki/RTP_Issues On Thu, May 7, 2009 at 5:16 AM, Jason White wrote: > Remko Kloosterman wrote: > > > > > > Did you make a wireshark trace yet? You should be able to find out > > exactly what's

Re: [Freeswitch-users] voicemail webapi returns 403 Forbidden

in the inside or On Thu, May 7, 2009 at 5:12 AM, seven wrote: > I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to > > http://192.168.1.27:8080/api/voicemail/web > > in the challenge box, no matter I input 1009, 1...@192.168.1.27, > 1...@localhost, 1...@default, I got Er

Re: [Freeswitch-users] Re-2: Ruby and ESL help

EventMachine is very different to TCPSocket and is definitely not a drop-in replacement. Take a look at FreeSWITCHeR (http://code.rubyists.com/projects/fs/repository) and see how they implemented EventMachine. More info about EventMachine and specifically #start_server is here: http://eventmachin

[Freeswitch-users] rtpDir Freeswitch

Roger on the previous post asking the question some time ago. I don't have time to get into this sort of thing, I've got too much on my plate right now as it is. Hopefully someone will take up the challenge. I've cross-posted this to the freeswitch forum. 73 de n2vqm Best Regards, Karl

Re: [Freeswitch-users] DTMF recognition flaky

Remko Kloosterman wrote: > > > Did you make a wireshark trace yet? You should be able to find out > exactly what's going on there, which protocol is used, etc. We've had > our share of problems with DTMF over SIP trunks as well. I've just discovered that I'm having a similar problem to the on

[Freeswitch-users] voicemail webapi returns 403 Forbidden

I loaded mod_xml_rpc on trunk version 13174, to get voicemail, I go to http://192.168.1.27:8080/api/voicemail/web in the challenge box, no matter I input 1009, 1...@192.168.1.27, 1...@localhost, 1...@default, I got Error 403. However I use freeswitch/works can get the voicemail interface. but

Re: [Freeswitch-users] Inboud Call Queue

Thanks Seven I'll try it very soon. _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of seven Sent: Thursday, May 07, 2009 5:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Inboud Call

Re: [Freeswitch-users] DTMF recognition flaky

Hi Jay, Did you make a wireshark trace yet? You should be able to find out exactly what's going on there, which protocol is used, etc. We've had our share of problems with DTMF over SIP trunks as well. Your problems could also be related to timing issues introduced by multiple gateways. Do you