Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread David Knell
kokoska rokoska wrote: David Knell napsal(a): Nowadays, if I want to withhold my CLI, I don't send anything. That works pretty well. Don't put junk in - I know that I can't call various destinations over BT if I use something bogus (like na, or thiscliisaloadofoldbollocks[1]); I get a 40

Re: [Freeswitch-users] FS benchmark test

2008-06-20 Thread Brian West
Also forgot to ask if this was a 64bit OS? /b On Jun 20, 2008, at 5:27 PM, Santiago Gimenez Ocano wrote: FS1: Virtualized Processor: Xeon @ 3.00GHz OS: Debian Kernel version: 2.6.18-6-686 FS version: 1.0 Preemption: disabled FS2: Not virtualized

Re: [Freeswitch-users] FS benchmark test

2008-06-20 Thread Brian West
Also can you pastebin this.. you seem to be missing some bits of it. /b On Jun 20, 2008, at 5:27 PM, Santiago Gimenez Ocano wrote: expression="^(.*)"> data="accountcode2="/> d

Re: [Freeswitch-users] FS benchmark test

2008-06-20 Thread Brian West
Sorry to burst your bubble but JS IS the problem. If you setup the same scenario with calling extension that just plays music you'll not see this issue. /b On Jun 20, 2008, at 5:27 PM, Santiago Gimenez Ocano wrote: JS is not the problem _

[Freeswitch-users] FS benchmark test

2008-06-20 Thread Santiago Gimenez Ocano
We have been doing tests in order to establish benchmarks. This is the scenario: SIPp ---> FreeSWITCH - > Destination SIPp: SIPp is using the uac scenario with a delay of 15000ms. INVITE --> 100 <-- 180 <--

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Good to know it snapped into place now! :P /b On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote: > This is all making sense to me now. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
Ah, I put the "apply-inbound-acl" in the wrong XML file. I put it in the external.xml. When I instead put it in the internal.xml, I got it working. That is of course the correct place, because it is bound to port 5060, and the inbound from Voxbone comes in on port 5060, as you correctly guess

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread kokoska rokoska
David Knell napsal(a): > Brian West wrote: >> It will always contain the number. Its the far ends responsibility to >> honor the privacy flags. The same happens on a PRI as far as I have >> seen. Nobody should have the ability to withhold that info from your >> "switch". >> > That's n

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread kokoska rokoska
Many tanks Mike & Brian for your help! I know I could explicitly set the effective caller id, but I hope there is other and a little bit comfortable way :-) Mainly because I have to record valid CLIP in CDR (and show it if somebody interested - like police etc.) and this force me to use anothe

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread David Knell
Brian West wrote: It will always contain the number. Its the far ends responsibility to honor the privacy flags. The same happens on a PRI as far as I have seen. Nobody should have the ability to withhold that info from your "switch". That's not quite the case. In the good old world,

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread Brian West
That is true... the remote carrier might or might not honor the flags. Love this stuff eh? /b On Jun 20, 2008, at 12:24 PM, Michael Jerris wrote: > If you want to actually not send to them you need to set the effective > caller id name/number. > > Mike > > On Jun 20, 2008, at 1:10 PM, Brian

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread Michael Jerris
If you want to actually not send to them you need to set the effective caller id name/number. Mike On Jun 20, 2008, at 1:10 PM, Brian West wrote: > It will always contain the number. Its the far ends responsibility to > honor the privacy flags. The same happens on a PRI as far as I have > se

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Let me guess they only send to port 5060? If you have your ACL's setup correctly those two IP's will be let in without auth. If you have the profile that runs on 5060 on your FreeSWITCH box with auth-calls=true Add this to that profile This should allow them thru without auth. If y

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread Brian West
It will always contain the number. Its the far ends responsibility to honor the privacy flags. The same happens on a PRI as far as I have seen. Nobody should have the ability to withhold that info from your "switch". /b On Jun 20, 2008, at 11:36 AM, kokoska rokoska wrote: > > It is base

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread kokoska rokoska
It is based on what I put as data into "privacy" directive: => Remote-Party-ID: "name" ;screen=yes;privacy=full => Remote-Party-ID: "name" ;screen=yes;privacy=number ... And, like I wrote before, "From:" header is unttached, thus always contains: From: "name" ;tag=Ky5yFavK7gUNB Best regards

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
I do not have outbound registation to Voxbone, because Voxbone is only incoming. I am not registrating Voxbone at all. In their FAQ, they have how to configure for Asterisk: [81.201.82.20] host = 81.201.82.20 type = friend insecure = very context = your-context canreinvite=no [81.201.82.21] ho

Re: [Freeswitch-users] Key System Emulation

2008-06-20 Thread Brian West
On Jun 20, 2008, at 2:11 AM, Matt Darnell wrote: > It must be hard to do, or everyone would be doing it. Its a matter of nobody can agree on how to do it. > > > If Freeswitch could offer the kind of functionality it would take > over: > A call would come in, via SIP, Zaptel, etc. > User answe

Re: [Freeswitch-users] Key System Emulation

2008-06-20 Thread Brian Snipes
At least for Snom phones that is possible ( and used daily with FS at my office ). In the sample dialplan is a setting for Snom's Park+Orbit button option. If you set the same Park+Orbit # on a button on all the phones you can push the button when in a call, the call is parked and the light on

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Ivan, Do you have your outbound registration to voxbone on the default/ internal profile? If so then your acl's might be wrong. The one sure fire way to do this is to just setup another profile without auth on a different port and run with that. /b On Jun 20, 2008, at 3:20 AM, Ivan

Re: [Freeswitch-users] CLIR on SIP

2008-06-20 Thread Brian West
What does the remote-party-id header say? /b On Jun 20, 2008, at 8:39 AM, kokoska rokoska wrote: > > Hi all! > > I have (may be) stupid question: How to hide caller identity in > Freeswitch, i.e. provide CLIR? > > I try following (I found it somewhere in the wiki): > > > > > where xxx I repla

[Freeswitch-users] CLIR on SIP

2008-06-20 Thread kokoska rokoska
Hi all! I have (may be) stupid question: How to hide caller identity in Freeswitch, i.e. provide CLIR? I try following (I found it somewhere in the wiki): where xxx I replace with all mentioned values - no|yes|name|full|number I try all combinations of above methods, but "From:" header sti

Re: [Freeswitch-users] outbound proxy

2008-06-20 Thread Bernhard Suttner
Thx, I will try that! Regards, Bernhard Am Freitag, den 20.06.2008, 13:51 +0100 schrieb David Knell: > Hi Berhnard, > > If I've understood the question - you want to send the INVITE to a > specified IP > address - then you can do this in the dialplan: > > data="sofia/gatewa

Re: [Freeswitch-users] outbound proxy

2008-06-20 Thread David Knell
Hi Berhnard, If I've understood the question - you want to send the INVITE to a specified IP address - then you can do this in the dialplan: Cheers -- Dave > Hi, > > how can I setup a pure outbound proxy that the IP address of the SIP > INVITE is the outbound proxy but SIP-TO

[Freeswitch-users] outbound proxy

2008-06-20 Thread Bernhard Suttner
Hi, how can I setup a pure outbound proxy that the IP address of the SIP INVITE is the outbound proxy but SIP-TO is the address of the PBX (and not the outbound proxy address) In my understanding the outbound proxy should only change the IP destination address and nothing in the original SIP pack

Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
As I understand the ACL, it only controls which external machines with the IP address range given in the acl.conf.xml are allowed into FreeSwitch. So this doesn't help me much with the DID, as FreeSwitch sends a "Proxy Authentication Required" SIP message back to the Voxbone server. How can I

Re: [Freeswitch-users] SIP over TCP

2008-06-20 Thread Matt Darnell
On Thu, Jun 19, 2008 at 6:21 PM, Brian West <[EMAIL PROTECTED]> wrote: > Just an FYI > > http://wiki.freeswitch.org/wiki/Exchange_2007_UM > > Thats on the WIKI already. > > /b Brian, That is fantastic pageI will be testing it this weekend. Thanks for the pointer. -Matt

[Freeswitch-users] Key System Emulation

2008-06-20 Thread Matt Darnell
It seems like Freeswitch is a real player in the PC based PBX space. I think the first product in this space that can emulate a key system will take over the space in very short time. Metaswitch, Broadsoft, etc have shared line appearances but none of the smaller systems have it. It must be hard