[Freeswitch-users] Park with Pre Answer

Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-user

Re: [Freeswitch-users] ACLs through proxy

Hey Metik, Thanks so much for your insights and your help. And yes, I was able to append the X-AUTH-IP header with no problem. But that didn't solve the issue. After some more research, it appears that the problem isn't with auth-calls at all. I disabled all auth-call directives in every sip

Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)

I've been asked to provide some screenshots, so here's some of the agent/supervisor interface: http://eagle.bsd.st/~andrew/cpxshots/ Hopefully the image names are self-explanatory. In the ringing picture, that URL pop is a configurable URL that can be used to integrate with a CRM, in my case our

[Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)

So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS fo

Re: [Freeswitch-users] packaging preference question

I think that sounds like a good idea. It would also keep permission management simple. -William King On Fri, 2009-12-18 at 17:22 -0500, Frank Carmickle wrote: > Hello all > > The packaging folk are interested in knowing if anyone has a problem with > having the install set up the user and gr

[Freeswitch-users] packaging preference question

Hello all The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX

Re: [Freeswitch-users] Ringing after call has been rejected

Actually my application returns 403 if it decides that it doesnt want to answer the call So I changed the response that my applicatgion gives to 486 and I now get the behavior that I wanted Thanks for the advice Brian Anthony Minessale-2 wrote: > > that will only work if you have not answe

Re: [Freeswitch-users] mod_conference scalability

Sounds like a plan. We will pursue it through the consult...@freeswith.org route. Thanks, Brian. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, December 18, 2009 3:30 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference

Re: [Freeswitch-users] SIP Re-invite

Thank you Mike for your suggestion on IRC. We did what you recommend and found out it's the iptables issue that we thought it was not there at the beginning since we saw the first 2 invites from the far end fine, but somehow it has something to do with the 3rd invite. I did close the Jira that

Re: [Freeswitch-users] mod_conference scalability

Hi Brian, Have a look at this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop - I took a quick look through the code and couldn't see any reason why you shouldn't have a bunch of eavesdroppers listening to a single caller. I'd be surprised if this didn't perform a lot better for y

Re: [Freeswitch-users] mod_conference scalability

I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consult...@freeswitch.org and provide

Re: [Freeswitch-users] Ringing after call has been rejected

that will only work if you have not answered yet. if you already have, you would need to indicate the tones inband like I mentioned. On Fri, Dec 18, 2009 at 2:16 PM, Yehavi Bourvine wrote: > Try the following: > > > I don't know whether it will work in your case, but her

Re: [Freeswitch-users] mod_conference scalability

Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with sho

Re: [Freeswitch-users] Ringing after call has been rejected

Try the following: I don't know whether it will work in your case, but here we use it to reject a call while we want to signal that the remote party is busy. Regards, __Yehavi: 2009/12/18 bcxml > > I have an incomming call being answered

Re: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header

could be possible with a code change, open a bounty on jira and someone may do it On Fri, Dec 18, 2009 at 12:35 PM, Jerry Richards wrote: > Is it possible to allow/deny REGISTER requests based on the User-Agent > header? I need to know/manage what devices are registering. > > Best Regards, > J

Re: [Freeswitch-users] mod_conference scalability

Brian, there was not one insulting word in anything I have said and as this is a community mailing list my replies are always voiced to address the public in general not you specifically, like I already mentioned in my last post. If you open a public forum on a FAQ be prepared to hear our policy.

Re: [Freeswitch-users] LUA and return variables

On 12/18/09 7:18 PM, Michael Collins wrote: Hi Michael, > One thing you can do is create an extension that does the enum look up > and then transfers the call back into the dialplan. You could set up a Cool idea, didn't think about that! > separate context that handles just the enum checking. Y

Re: [Freeswitch-users] mod_conference scalability

Brian, Now that you know the scale freeswotch scales to in you scenario, and having designed a mult-server solution can you not add more server to scale further? As freeswitch continues to improve retest and revise your architecture design. Sent from my iPhone On Dec 18, 2009, at 11:14

Re: [Freeswitch-users] mod_conference scalability

On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > I was evaluating the technologies available, and I thought you would be > interested in my results. However, almost every other reply I get from you > to my posts, rather than being helpful, has been hostile and insulting. > Thanks for your input.

Re: [Freeswitch-users] LUA and return variables

I use a similar method (transfer to XML dialplan based on the value of "${enum_route_1}") to determine if the SIP URI is native to a particular FS instance or if it needs to be sent off-net and it works well. -metik Michael Collins wrote: > > > On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij

Re: [Freeswitch-users] mod_conference scalability

I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of “having robots call the conference

[Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header

Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/

Re: [Freeswitch-users] SIP Re-invite

Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. _

Re: [Freeswitch-users] LUA and return variables

On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij wrote: > Hi guys (and girls)! > > I'm working on a little bit of ENUM trickery and I tried doing some > (illegal) nested conditions. :-) > > What I want to do is to first check enum with the ENUM application, > then depending on the answer do some stuf

Re: [Freeswitch-users] SIP Re-invite

Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. _

Re: [Freeswitch-users] mod_conference scalability

yes, I understand. My reply was to the thread in general not directed at you =p On Fri, Dec 18, 2009 at 11:41 AM, François Delawarde < fdelawa...@wirelessmundi.com> wrote: > It was of course just bad humor, I love both projects for what they are, > and I agree that both have their own advantages

Re: [Freeswitch-users] ACLs through proxy

Honestly, several years ago I accomplished this by mod'ing SER (which became OpenSER which was then forked to OpenSIPS and Kamalio) and using one cluster of proxies for subscriber endpoints and another for infrastructure (so that I could keep RTP flows optimized yet support double NAT when requ

Re: [Freeswitch-users] mod_conference scalability

It was of course just bad humor, I love both projects for what they are, and I agree that both have their own advantages and inconvenients. For example, accessing that same conference from a dahdi card could be another goal where Asterisk would be at an advantage, as chan_dahdi is still superior (

[Freeswitch-users] Fwd: incoming call

Hi, i have up the freeswitch with domain(eg sipserver.domain.com) name instead of local ip, two clints are regitered with freeswitch using domain name(eg sipserver.domain.com), one client is making a call to other one, other clint receiving a invite request like this 173927 3120.658532 10.91.154.1

Re: [Freeswitch-users] SIP Re-invite

I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: > Mike, > > My latest traces that I captured were done within the FS box: > http://pastebin.freeswitch.org/11541 > > Thank you,

Re: [Freeswitch-users] Ringing after call has been rejected

not answering it would be the best way. if you want to generate fake congestion you can use tone_stream:// or gentones On Fri, Dec 18, 2009 at 5:16 AM, bcxml wrote: > > I have an incomming call being answered by FreeSwitch and passed to IVR > application which rejects the call. > > The call is

[Freeswitch-users] FreeSWITCH Weekly Conference Call Starting Shortly!

Hello everyone! Today's agenda is listed here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 Also, we are going to be giving away goodies on some of the upcoming conferences, so call in and see what we've got in store. :) For the first 15 minutes we'll let everyone mingle and then we'll g

Re: [Freeswitch-users] mod_conference scalability

Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at dif

Re: [Freeswitch-users] mod_conference scalability

Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: "FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS." Anyway, you

Re: [Freeswitch-users] Voicemail->Email

oh really, sendmail segfaults? if another application is crashing you need to figure that out, whatever used to work doesnt now so you need to figure out what it was and let us know. On Fri, Dec 18, 2009 at 3:51 AM, François Legal wrote: > I get the same result with sendmail. This used to work

Re: [Freeswitch-users] ACLs through proxy

You need to add that header manually in your OpenSIPS config, FreeSWITCH wont look in record-route/via to try to guess it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 18-Dec-09, at 10:53 AM, Bill W wrote: > Hello Mathi

Re: [Freeswitch-users] ACLs through proxy

Hello Mathieu, I assumed that apply-proxy-acl was a modifier of auth-calls, so in my quick tests I just hard-coded the UA IP in the profile. And I get: 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.97.83/32 Where 64.135.119.105 is the

Re: [Freeswitch-users] mod_conference scalability

What is your dialplan on the secondary box? On Dec 18, 2009, at 9:08 AM, Brian wrote: I’ve got FS running on a 64 bit OS, and here is more info on the tes t procedure. I’ve got one server (primary) that hosts the speaker call (this is m eant to be a primary conference with a few speakers,

Re: [Freeswitch-users] mod_xml_ldap compile issue.

On 12/18/2009 02:13 PM, Keith Laaks wrote: > Hi, > > I am having an issue getting mod_xml_ldap to compile properly > > > making all mod_xml_cdr > > making all mod_xml_ldap > Creating mod_xml_ldap.la... > /usr/bin/ld: > /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/l

Re: [Freeswitch-users] Presence across several networked FSs

I have found some ways to get presence, or rather BLF functions to work on Snom telephones in a distributed network with several FSs. I'll post a solution on the wiki when I have tested it further. Anyhow, I'm using the mod_event_multicast module with the following configuration:

Re: [Freeswitch-users] How can I join two freeswitch on two servers?

Hi, As far as I know, there are two ways to connect two freeswitch, by using ACL or using authentication. Also from this email history discussion, another solution is to create user in FS B directory,then treat server B as normal gateway by adding gateway definiton in FS A. So my question is ho

Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN

That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. You also fail to mention if the progress is a 180 or a 183 with sdp and

Re: [Freeswitch-users] mod_conference scalability

I've got FS running on a 64 bit OS, and here is more info on the test procedure. I've got one server (primary) that hosts the speaker call (this is meant to be a primary conference with a few speakers, but my test simplifies this to just one speaker). I've got a second server (secondary) that h

[Freeswitch-users] mod_xml_ldap compile issue.

Hi, I am having an issue getting mod_xml_ldap to compile properly making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.ro

Re: [Freeswitch-users] mod_conference scalability

Brian, You haven't said what codecs are being used yet. Are the listeners using a different codec to the speaker? If so, you're potentially doing transcoding on every single channel, which would make CPU usage skyrocket. -Steve 2009/12/17 Anthony Minessale : > What exactly is your test process?

[Freeswitch-users] Ringing after call has been rejected

I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejecti

[Freeswitch-users] LUA and return variables

Hi guys (and girls)! I'm working on a little bit of ENUM trickery and I tried doing some (illegal) nested conditions. :-) What I want to do is to first check enum with the ENUM application, then depending on the answer do some stuff. Say that the domain part of the ENUM answer is robin.nl,

Re: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN

Should I open a JIRA for this? Best regards Peter Peter P GMX schrieb: > Hello, > > we have the following scenario: > A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For > the called FS user, call forwarding has been enabled to another PSTN > extension (B) . > Result: The cal

Re: [Freeswitch-users] Voicemail->Email

I get the same result with sendmail. This used to work in 1.0.3 , and after upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is still there. François On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Schönbeck wrote: Currently it is Version 1.0.trunk (15982) VON: freesw

Re: [Freeswitch-users] SIP Re-invite

Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users]

Re: [Freeswitch-users] Destination Formats Expression

Thanks that will be a great help Jason White-14 wrote: > > Edmar Cruz wrote: >> >> Is there a link or tutorial for the expressions format. > > Anything that describes Perl regular expressions should help, and for > reference, see the pcre(3) manual page, and use the pcretest program t