Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup

Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 20

Re: [Freeswitch-users] echo cancellation on PRI cards

Sharing my humble experience: in Brazil we usually need echo cancellation to have reliable DTMF detection _and_ voice quality over E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for sip/tdm gateway devices or IVR applications. Usually there's no need for echo cancellation on links fr

Re: [Freeswitch-users] TLS support in Debian build

Brian West wrote: > if you installed the ssl devel stuff AFTER you configured you'll need > to reconfigure. I'm reasonably sure it was installed already, unless it was pulled in recently by a package upgrade. The configure script needs to look in /usr/include/openssl for the headers. I'll have

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: David Knell wrote: Steve Underwood wrote: [whopping big snip] The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. I'm sorry if you find disagreement offensive; you might not wish

Re: [Freeswitch-users] TLS support in Debian build

if you installed the ssl devel stuff AFTER you configured you'll need to reconfigure. /b On Mar 17, 2009, at 8:46 PM, Jason White wrote: > I've just tried enabling TLS support, and the SIP profiles with TLS > enabled in > them won't load. > > According to the wiki, this is typically the resu

[Freeswitch-users] TLS support in Debian build

I've just tried enabling TLS support, and the SIP profiles with TLS enabled in them won't load. According to the wiki, this is typically the result of missing headers during the build process, with TLS having not been included. However, on my Debian system, I have header files under /usr/include/

Re: [Freeswitch-users] Nibblebill - DB Error while updating cash!

Hi there, The updates to the DB are working, but the error is still being thrown. I will try and fix this tonight. Diego also reported the same issue last week, I just haven't gotten around to it. My apologies. The bug is filed, I'll close it out when it's fixed. - Darren _

Re: [Freeswitch-users] echo cancellation on PRI cards

David Knell wrote: > Steve Underwood wrote: >> [whopping big snip] >> >>> The first bit of that's a tad patronising, isn't it, >>> >> You are the one who started out being offensive. >> > I'm sorry if you find disagreement offensive; you might not wish to > read beyond this > point if s

Re: [Freeswitch-users] echo cancellation on PRI cards

> I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up spending > large sums > of money on technology which they probably do not need. Anybody with years, perhaps decades, of DSP progra

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: [whopping big snip] The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. I'm sorry if you find disagreement offensive; you might not wish to read beyond this point if so. and, in the case of the decade-old Acu

Re: [Freeswitch-users] Newbie question: Why can't I dial?

On Tue, Mar 17, 2009 at 3:35 PM, Mark Thomas wrote: > Hello, everyone. > > I am new to Freeswitch, and telephony in general.  I am trying to set up a > Freeswitch system at work for a project, and I have hit a wall. > > I have a dedicated LD T1 from Qwest and a Sangoma A104 card.  I believe I >

Re: [Freeswitch-users] Newbie question: Why can't I dial?

Mark, You should join both #openzap and #freeswitch on irc.freenode.net there are way too many things to go over and the list would just be too slow. /b On Mar 17, 2009, at 5:35 PM, Mark Thomas wrote: Hello, everyone. I am new to Freeswitch, and telephony in general. I am trying to s

[Freeswitch-users] Newbie question: Why can't I dial?

Hello, everyone. I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall. I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to

Re: [Freeswitch-users] DTMF detection during bridge

Check out the bind_meta_app that exists in the default examples... I think thats what you want. /b On Mar 17, 2009, at 4:13 PM, Cristian Talle wrote: > Hi, > > Is there any easy way to get in FS the same behavior as when using the > "d" flag with asterisk's Dial command? > I need FS to jump to

[Freeswitch-users] DTMF detection during bridge

Hi, Is there any easy way to get in FS the same behavior as when using the "d" flag with asterisk's Dial command? I need FS to jump to a different extension if the caller presses a digit while waiting for the called party to answer. *"...d*: intercepts any dtmf while waiting for the call to be

[Freeswitch-users] Nibblebill - DB Error while updating cash!

Hi All, I have installed nibblebill and* it is able to bill the calls.* However, it is giving following error in FreeSwitch server. 2009-03-17 23:17:19 [DEBUG] mod_nibblebill.c:283 bill_event() Doing update query [UPDATE accounts SET cash=cash-0.045767 WHERE id="1"] 2009-03-17 23:17:19 [CRIT] mo

Re: [Freeswitch-users] echo cancellation on PRI cards

David Knell wrote: > Steve Underwood wrote: >> David Knell wrote: >> >>> Steve Underwood wrote: >>> > When there is Echo being generated from the far end, usually in a > bridged call. If you application is just an IVR, with no far end > connectivity, then you shouldn't need an

Re: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ?

I checked, I don't see sw1.freephonie.net in the logs trying to resolve it... and the SRV records are all correct as are the naptr records which is shocking ;) /b On Mar 17, 2009, at 1:16 PM, ludovic wrote: > I understood the example. > What I mean is that my DNS issue comes from sofia-sip a

Re: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ?

I understood the example. What I mean is that my DNS issue comes from sofia-sip and my sip provider (freephonie.net) name resolution which fails when calling whereas it is well resolved during the registration process. Here is a trace : 2009-03-17 18:47:28 [NOTICE] switch_channel.c:538 switch_c

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: David Knell wrote: Steve Underwood wrote: When there is Echo being generated from the far end, usually in a bridged call. If you application is just an IVR, with no far end connectivity, then you shouldn't need an echo can. If you are bridging calls, then at som

Re: [Freeswitch-users] [OpenZap] problem with TE220P

> any idea how i can fix this error ? > I believe this is a harmless warning. However, you might try to use ozmod_libpri, which uses the libpri PRI stack instead of the built-in OpenZAP PRI stack. More info here: http://wiki.freeswitch.org/wiki/OpenZAP#OpenZAP_Installation -MC __

Re: [Freeswitch-users] echo cancellation on PRI cards

Anthony Knight wrote: > Thanks for the feedback. > > I have plenty of experience with IVRs and Dialogic cards (starting > with D121/LSI120s and SS96s under DOS in the 90's all the way up to > Intel's DM/Vs) and didn't ever have a problem with DTMF collection > with ISDN PRI lines except occasi

Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

wow, now that was fast :-) Cheers for all replies, setting the caller-id-in-from-parameter was sufficient! regards Christian ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch

[Freeswitch-users] Digium TE220P problem.

Hi, I have problem with Digium TE220P. Everything works, i can call & talk, but everytime i have CRIT message: 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 When FS start show me ERR message: 2009-03-14 17:44:06 [ERR] Span:0 Q.921()

Re: [Freeswitch-users] echo cancellation on PRI cards

Thanks for the feedback. I have plenty of experience with IVRs and Dialogic cards (starting with D121/LSI120s and SS96s under DOS in the 90's all the way up to Intel's DM/Vs) and didn't ever have a problem with DTMF collection with ISDN PRI lines except occasionally with wireless and cell phones (B

Re: [Freeswitch-users] Possible memory / cpu leak

The crash on shutdown was an issue in mod_spidermonkey that was accidentally added if you update again it's gone. please run the valgrind command again then make several calls that fall in line with your normal usage pattern so the program can get an accurate trace of the memory usage. On Tue

Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

The From: header is not the correct place to place the caller id in SIP yet some providers assume it is. If you add this to your gateway xml config it should fix your problem On Wed, Mar 11, 2009 at 12:07 PM, Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for

Re: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup

there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong > I apologize if this is a double post to -dev. I'm not sure why I don't see > my message appearing, so I'm going to try again in the -user list (first > timer posting here ;). > > I have a situat

Re: [Freeswitch-users] Possible memory / cpu leak

You're not leaking... I wouldn't call 737 bytes a leak. /b On Mar 17, 2009, at 10:49 AM, Chris Fowler wrote: Thanks for the tip Brian. Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log Chris. ___ Freeswitch-users mailing li

Re: [Freeswitch-users] Possible memory / cpu leak

Thanks for the tip Brian. Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log Chris. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIB

Re: [Freeswitch-users] Problem dialing out via E1

On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi > again, using the same settings as we always did, and we can now make external > calls. I suspect that whatever BT did yesterday kicked the circuit back into > life.

Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

Maybe it can help by following this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012083.html On Mar 17, 2009, at 11:23 PM, Christian Benke wrote: > Hi! > > Is this not possible with registration at a gateway or is there a > other > reason why i didn't get any respo

Re: [Freeswitch-users] echo cancellation on PRI cards

David Knell wrote: > Steve Underwood wrote: >>> When there is Echo being generated from the far end, usually in a >>> bridged call. If you application is just an IVR, with no far end >>> connectivity, then you shouldn't need an echo can. If you are bridging >>> calls, then at some point you may

Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

gateways have their username in the from section, callerid is sent out as remote-party-id or p-asserted-identity. if you want the from part to have the user you need to set the "caller- id-in-from" param to "true" Math On 11-Mar-09, at 1:07 PM, Christian Benke wrote: > Hi! > > I've recently s

Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

Try export instead of "set" /b On Mar 17, 2009, at 10:23 AM, Christian Benke wrote: >> > data="effective_caller_id_number=+4312345678${caller_id_number}"/> ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitc

Re: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username"

Hi! Is this not possible with registration at a gateway or is there a other reason why i didn't get any responses on this question? Regards Christian On Wed, 11 Mar 2009 18:07:42 +0100 Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > an

Re: [Freeswitch-users] enable anonymous incomming calls

Thanks a lot dujinfang !! roberto 2009/3/17 dujinfang : > at default config, in conf/sip_profiles/external.xml > >     >     >     > > where $${external_sip_port} is a variable you can find in conf/ > vars.xml, normally it's 5080, make sure sipbroker route calls to that > port, and then you ca

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: When there is Echo being generated from the far end, usually in a bridged call. If you application is just an IVR, with no far end connectivity, then you shouldn't need an echo can. If you are bridging calls, then at some point you may need it, depending on what else is

Re: [Freeswitch-users] enable anonymous incomming calls

at default config, in conf/sip_profiles/external.xml where $${external_sip_port} is a variable you can find in conf/ vars.xml, normally it's 5080, make sure sipbroker route calls to that port, and then you can make a dialplan in conf/dialplan/public.xml turn on verbose log

[Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup

I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the ag

[Freeswitch-users] [OpenZap] problem with TE220P

Hi, I have problem with Digium TE220P. Everything works, i can call & talk, but everytime i have CRIT message: 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 When FS start show me ERR message: 2009-03-14 17:44:06 [ERR] Span:0 Q.921()

[Freeswitch-users] enable anonymous incomming calls

Hi all I'm freswitch newbie  and have a simple question. How can enable anonymous inbound calls ? I would like to use freeswitch to accept incomming calls from sipbroker DIDs Any hint ? Thank in advance for all freeswitch team !! roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.con

[Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup

I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the ag

Re: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ?

What I provided you was an example. I don't think you understood what I was talking about. In the settings for ext-sip-ip and ext-rtp-ip you'll have to use something like "host:yourdyndnshostname.blah.tld" Then set the sip-ip and rtp-ip to what ever is auto detected. /b On Mar 17, 2009, a

Re: [Freeswitch-users] echo cancellation on PRI cards

Wasim Baig wrote: > 2009/3/17 Anthony Knight > > > I'm thinking about doing a project that would use FreeSWITCH as an > IVR, with callers being routed in by both ISDN PRI, and also SIP > trunks, with occasional bridge calls between callers. > > I'm wonde

Re: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ?

Thanks. It seems that it comes from my sip provider. when using my_host as my hostname, reg fails when using my_host.com as my hostname, reg succeeds  (my_host.com does not exist as a domain internet) when using ip address, reg succeeds. Tested with version 1.0.3 Is it a way to force the IP

Re: [Freeswitch-users] Problem with shortened local extensions

I also had problems with not reaching local extensions some time. I solved it by adding: to /usr/local/freeswitch/conf/directory/default.xml Best regards Peter f...@xenpad.eu schrieb: > Hi, > > On Mon, 16 Mar 2009, Michael Collins wrote: > >>> I have a (probably dumb) question tha

Re: [Freeswitch-users] Problem dialing out via E1

Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. However placing an external call into FS isn't as successful

Re: [Freeswitch-users] Mod_limit stuck when hitting limit value

limit_hash uses a faster data structure then limit but works the same way for tne end-user. viens sur IRC si t'as des questions en francais =) Math On 17-Mar-09, at 3:06 AM, rod wrote: > Hi, > > not too hard :p > but it's just a bad habit when I write in my native language > (french). I >

Re: [Freeswitch-users] Problem dialing out via E1

Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -Original Messag

Re: [Freeswitch-users] Freeswitch and Kamailio (OpenSer) Integration

Yes it is possible but there is no documentation on how to do it. You will have to learn SIP and understand what you are doing. Forwarding the call to FS for nat may cause issue as FS will then not have direct connection to the phone and may not be able to always detect it is behind NAT. Have

Re: [Freeswitch-users] Windows-compatible FXO PCI card?

On Tue, Mar 17, 2009 at 2:02 PM, Gilles wrote: I don't know of FXO PCI cards to connect an XP/Vista host to a phone > line. Does someone know of such a thing? Sangoma makes a low cost 4FXO, 1FXS. http://sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html -- wasim h. baig |

[Freeswitch-users] Windows-compatible FXO PCI card?

Hello For SOHO users, getting a second, Linux-based computer just to run a small voice server is overkill, so I'm thinking of selling an application based on the Windows version of Freeswitch. Instead of the Sangoma USB connector, I'd really prefer to sell them a PCI card, because it's less me

[Freeswitch-users] sip redirect contact variable no more available in SVN 12638

Hi, running SVN r12638, I don't have access anymore to these 2 variables after a SIP 302 message, using info application: variable_sip_redirect_contact_user_0 variable_sip_redirect_contact_host_0 It was okay with SVN r12611. regards, rod ___ Freesw

Re: [Freeswitch-users] Mod_limit stuck when hitting limit value

Hi, not too hard :p but it's just a bad habit when I write in my native language (french). I guess that this spelling is not too common for english speaker. I'll do my best next time to write it correctly. @tamas you are right, we could use limit_hash the same way as limit when not specifying