Re: [Freeswitch-users] Exchange 2007 UM - DTMF problem
post a trace of FS after pressing f8 from the cli detailing the entire call and we can have a look. On Fri, Sep 19, 2008 at 6:29 PM, Matt Darnell <[EMAIL PROTECTED]> wrote: > On Wed, Sep 17, 2008 at 10:14 AM, Matt Darnell <[EMAIL PROTECTED]> > wrote: > > On Fri, Jul 25, 2008 at 8:51 PM, UV <[EMAIL PROTECTED]> wrote: > >> Yes I did, but you might not even need that. > >> Try adding in your external > SIP > >> profile and see if it solves the problem. > >> > > > > I am still trying to get the DTMF 100%, I added the value but get this > > message in the debug log: > > [WARNING] mod_sofia.c:787 sofia_receive_message() Cannot pass 2833 on > > a transcoded call > > > > It does not appear that any transcoding is happening from the SIP > > setup messages. > > Steve, > > Thanks for the response, here is the architecture: > > SIP Provider <-> SIP UDP <-> Internet <-> Freeswitch <-> LAN <-> SIP > TCP <-> Exch 2007 > > It appears to be G711 throughout the entire call, is it transcoded > just because Freeswitch is bridging the call? > > -Matt > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Also what is "version" and make sure you're on svn trunk if possible. That [] had a bug in it at some point and I can't recall the exact rev we fixed it in. But I'm on svn trunk here working to nail down bugs so we can release 1.0.2. /b ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Jon, I did this originate at the CLI and each one has its own callerid: originate [origination_caller_id_number=6]sofia/x.x.x.x/1000, [origination_caller_id_number=9]sofia/x.x.x.x/1001 And this in the dialplan: This is confirmed working as is my previous email. You have to watch and make sure you're looking at the correct invite. Also remove the caller-id-in-from. My RPID looks like: Remote-Party-ID: "Extension 1005" ;screen=yes;privacy=off Remote-Party-ID: "Extension 1005" ;screen=yes;privacy=off I have even tried this out to the PSTN to my cellphone and it works. /b On Sep 19, 2008, at 4:44 PM, Jon Bruel wrote: > Yes I can, but somehow I start to repeat myself. An INVITE is sent > allright every time, but one of the headers differs: > Works: > Remote-Party-ID: > ;screen=yes;privacy=off > Doesn't work: > Remote-Party-ID: "Extension 1000" > ;screen=yes;privacy=off > I have tried with different variable names in the bridge data: > data="[effective_caller_id_number=45161061]sofia/gateway/ > 45161061/$1"/> > as well as: origination_caller_id_number > data="[origination_caller_id_number=45161061]sofia/gateway/ > 45161061/$1"/ >> > the latter combined with: value="true"/>. > But the only thing which works is setting the > effective_caller_id_number > in the dialplan before the bridge. That would be fine, unless I want > to > bridge to several destinations. Every time the call is rejected, the > Remote-Party-ID-header hasn't the right value. > I have tried to execute another dialplan at bridge, testing with other > variables, and the info application doesn't reveal the variable I > try to > include in square brackets in the bridge data string. So maybe its not > passed over to the B-leg. > If you need more data, let me know, and I'll package it during the > weekend. > Jon > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Wrong port on response
Yep just checked in my 55i that I have and no STUN nor RPORT. Great for the LAN silly for the WAN. the force-rport is your only option and its a global per profile setting. /b On Sep 19, 2008, at 6:08 PM, David Aldworth wrote: > User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ > v3.2.8.45. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Exchange 2007 UM - DTMF problem
On Wed, Sep 17, 2008 at 10:14 AM, Matt Darnell <[EMAIL PROTECTED]> wrote: > On Fri, Jul 25, 2008 at 8:51 PM, UV <[EMAIL PROTECTED]> wrote: >> Yes I did, but you might not even need that. >> Try adding in your external SIP >> profile and see if it solves the problem. >> > > I am still trying to get the DTMF 100%, I added the value but get this > message in the debug log: > [WARNING] mod_sofia.c:787 sofia_receive_message() Cannot pass 2833 on > a transcoded call > > It does not appear that any transcoding is happening from the SIP > setup messages. Steve, Thanks for the response, here is the architecture: SIP Provider <-> SIP UDP <-> Internet <-> Freeswitch <-> LAN <-> SIP TCP <-> Exch 2007 It appears to be G711 throughout the entire call, is it transcoded just because Freeswitch is bridging the call? -Matt ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Wrong port on response
On Sep 19, 2008, at 6:08 PM, David Aldworth wrote: Hello - Got an issue with Freeswitch not responding on the port that the initial request was made on. I'm not beyond believing that it is a NAT or router issue except that I can register a Cisco phone from another location or a softphone from the same location without any problem. This Aastra just won't work for some reason. We have connectile-dysfuntion turned on. Otherwise we are using the default profile settings. Auth is on (as you can see from the below). Basically, the Reg request comes from port 41450, but freeswitch responds on port 5060. Again, other UA's work fine, just one Cisco and one Aastra from this site do not. Meanwhile a soft phone from this site, and the same model cisco from another site do not work. SIP dump and external profile are below. Thank you for any help. David This isn't a bug. If you notice the phone explicitly said in its contact for us to contact them via 192.168.1.192:5060 so you'll need to enable stun on the phone or rport. If you were to enable rport on the aastra it would just work correctly. If Aastra doesn't support RFC3581 or STUN then they are worthless phones just like Polycom. Its not the registrars problem to fix your nat issues. The phone should support RFC3581 (rport) or STUN and it would just work like the Snom's do. Try adding this param to your sofia profile. It will break cisco phones or any other phone that follows the sip spec. This explicitly breaks RFC to accommodate broken phones. in your sofia profile. /b U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060 REGISTER sip:atl.teliax.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab. Max-Forwards: 70. Content-Length: 0. To: Aastra Test . From: Aastra Test ;tag=cbf2963ddfab0d6. Call-ID: [EMAIL PROTECTED] CSeq: 21481 REGISTER. Contact: Aastra Test ;expires=300. Allow-Events: talk,hold,conference. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO. Expires: 300. User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45. . U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34. From: Aastra Test ;tag=cbf2963ddfab0d6. To: Aastra Test ;tag=Sarvm1DjmU3Zg. Call-ID: [EMAIL PROTECTED] CSeq: 21481 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="atl.teliax.net", nonce="d3538e86-9d86-dd11-82bf-001143e64915", algorithm=MD5, qop="auth". Content-Length: 0. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Wrong port on response
Hello - Got an issue with Freeswitch not responding on the port that the initial request was made on. I'm not beyond believing that it is a NAT or router issue except that I can register a Cisco phone from another location or a softphone from the same location without any problem. This Aastra just won't work for some reason. We have connectile-dysfuntion turned on. Otherwise we are using the default profile settings. Auth is on (as you can see from the below). Basically, the Reg request comes from port 41450, but freeswitch responds on port 5060. Again, other UA's work fine, just one Cisco and one Aastra from this site do not. Meanwhile a soft phone from this site, and the same model cisco from another site do not work. SIP dump and external profile are below. Thank you for any help. David U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060 REGISTER sip:atl.teliax.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab. Max-Forwards: 70. Content-Length: 0. To: Aastra Test . From: Aastra Test ;tag=cbf2963ddfab0d6. Call-ID: [EMAIL PROTECTED] CSeq: 21481 REGISTER. Contact: Aastra Test >;expires=300. Allow-Events: talk,hold,conference. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO. Expires: 300. User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45. . U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34. From: Aastra Test ;tag=cbf2963ddfab0d6. To: Aastra Test ;tag=Sarvm1DjmU3Zg. Call-ID: [EMAIL PROTECTED] CSeq: 21481 REGISTER. User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: 100rel, timer, precondition, path, replaces. WWW-Authenticate: Digest realm="atl.teliax.net", nonce="d3538e86-9d86- dd11-82bf-001143e64915", algorithm=MD5, qop="auth". Content-Length: 0. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Let me show you some examples that I'm doing right now that work fine.
I do this at the CLI and it works:
originate {origination_caller_id_number=9184238080}sofia/gateway/
asterlink.com/1918XXX
And this is in my dialplan works:
/b
On Sep 19, 2008, at 4:44 PM, Jon Bruel wrote:
> Yes I can, but somehow I start to repeat myself. An INVITE is sent
> allright every time, but one of the headers differs:
> Works:
> Remote-Party-ID:
> ;screen=yes;privacy=off
> Doesn't work:
> Remote-Party-ID: "Extension 1000"
> ;screen=yes;privacy=off
> I have tried with different variable names in the bridge data:
> data="[effective_caller_id_number=45161061]sofia/gateway/
> 45161061/$1"/>
> as well as: origination_caller_id_number
> data="[origination_caller_id_number=45161061]sofia/gateway/
> 45161061/$1"/
>>
> the latter combined with: value="true"/>.
> But the only thing which works is setting the
> effective_caller_id_number
> in the dialplan before the bridge. That would be fine, unless I want
> to
> bridge to several destinations. Every time the call is rejected, the
> Remote-Party-ID-header hasn't the right value.
> I have tried to execute another dialplan at bridge, testing with other
> variables, and the info application doesn't reveal the variable I
> try to
> include in square brackets in the bridge data string. So maybe its not
> passed over to the B-leg.
> If you need more data, let me know, and I'll package it during the
> weekend.
> Jon
>
>
> ___
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> Freeswitch-users@lists.freeswitch.org
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> http://www.freeswitch.org
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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Yes I can, but somehow I start to repeat myself. An INVITE is sent allright every time, but one of the headers differs: Works: Remote-Party-ID: ;screen=yes;privacy=off Doesn't work: Remote-Party-ID: "Extension 1000" ;screen=yes;privacy=off I have tried with different variable names in the bridge data: as well as: origination_caller_id_number data="[origination_caller_id_number=45161061]sofia/gateway/45161061/$1"/ > the latter combined with: . But the only thing which works is setting the effective_caller_id_number in the dialplan before the bridge. That would be fine, unless I want to bridge to several destinations. Every time the call is rejected, the Remote-Party-ID-header hasn't the right value. I have tried to execute another dialplan at bridge, testing with other variables, and the info application doesn't reveal the variable I try to include in square brackets in the bridge data string. So maybe its not passed over to the B-leg. If you need more data, let me know, and I'll package it during the weekend. Jon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] billing platform
On Sat, Sep 20, 2008 at 2:07 AM, xbipin <[EMAIL PROTECTED]> wrote: > > whats the amount ur looking for this bounty, mayb guys like me who wanna > use dwiebe can let you know more about that it as a proper softswitch can pitch in a little and can also help in testing > and coding. what is sorely missing is thorough testing and documentation and requests for features for example, one cool thing that I couldn't do with asterisk and now can with fs and astpp is charge for codec conversion :) (not coded yet, but simple to add) svn co https://astpp.svn.sourceforge.net/svnroot/astpp/trunk[path/to/where/you/want/it] look in the freeswitch subdir and start firing away perhaps we can track this on the astpp website on the forum there OR on the astpp tracker mailing list so that the rest of the freeswitch people here aren't disturbed, looking forward to seeing you there as Mike said www.astpp.org is the place for this - wasim > > Darren Wiebe wrote: > > > > As the author of ASTPP I'd be very interested in negotiating a bounty to > > speed things up. User authentication is working with Freeswitch as well > > as lcr and call rating. It's largely a matter of documenting and > testing. > > > > Darren Wiebe > > [EMAIL PROTECTED] > > > > xbipin wrote: > >> are u interested in a paid implementation as im willing to post a bounty > >> for > >> it if there r others like us willing to use it with freeswitch. wikipbx > >> is > >> also good for a start but one thing i dont understand is if freeswitch > is > >> made for cross platform then y r other projects related to it limiting > it > >> to > >> just linux? > >> > >> > >> > >> Jair Santos wrote: > >> > >>> I've been trying to install. The problem is the lack of documentation > >>> on > >>> how to integrate with FS. > >>> > >>> Jair Santos > >>> > >>> > >>> > >>> > >>> > >>> > >>> > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of xbipin > Sent: Thursday, September 18, 2008 3:25 AM > To: freeswitch-users@lists.freeswitch.org > Subject: [Freeswitch-users] billing platform > > > > has any1 been able to use ASTPP billing with freeswitch and > if so then my next question, is it possible to use it on a > win2k3 platform and not linux as i have freeswitch running on > a win2k3 server since a few days and works fine, i just need > a prepaid type of script that basically stops a call once its > credit is over. i will add the credit etc to the account manually. > > im ready to post a bounty if any1 willing to do it. the > script just needs to end the call once it computes that > remaining time is 0 based on available time which is based on > the cost per minute and available credit and avoid calling > again which needs to be done as soon as call is placed so the > call is never connected if account has no credit. > -- > View this message in context: > http://www.nabble.com/billing-platform-tp19549891p19549891.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > http://www.freeswitch.org > > > >>> ___ > >>> Freeswitch-users mailing list > >>> Freeswitch-users@lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >>> > >> > >> > > > > > > -- > > Darren Wiebe > > [EMAIL PROTECTED] > > > > Aleph Communications > > www.aleph-com.net > > > > > > > > ___ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/billing-platform-tp19549891p19578736.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as you scope creep, so shall we reap ... ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswi
Re: [Freeswitch-users] billing platform
whats the amount ur looking for this bounty, mayb guys like me who wanna use it as a proper softswitch can pitch in a little and can also help in testing and coding. Darren Wiebe wrote: > > As the author of ASTPP I'd be very interested in negotiating a bounty to > speed things up. User authentication is working with Freeswitch as well > as lcr and call rating. It's largely a matter of documenting and testing. > > Darren Wiebe > [EMAIL PROTECTED] > > xbipin wrote: >> are u interested in a paid implementation as im willing to post a bounty >> for >> it if there r others like us willing to use it with freeswitch. wikipbx >> is >> also good for a start but one thing i dont understand is if freeswitch is >> made for cross platform then y r other projects related to it limiting it >> to >> just linux? >> >> >> >> Jair Santos wrote: >> >>> I've been trying to install. The problem is the lack of documentation >>> on >>> how to integrate with FS. >>> >>> Jair Santos >>> >>> >>> >>> >>> >>> >>> -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of xbipin Sent: Thursday, September 18, 2008 3:25 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] billing platform has any1 been able to use ASTPP billing with freeswitch and if so then my next question, is it possible to use it on a win2k3 platform and not linux as i have freeswitch running on a win2k3 server since a few days and works fine, i just need a prepaid type of script that basically stops a call once its credit is over. i will add the credit etc to the account manually. im ready to post a bounty if any1 willing to do it. the script just needs to end the call once it computes that remaining time is 0 based on available time which is based on the cost per minute and available credit and avoid calling again which needs to be done as soon as call is placed so the call is never connected if account has no credit. -- View this message in context: http://www.nabble.com/billing-platform-tp19549891p19549891.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw itch-users http://www.freeswitch.org >>> ___ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> >> > > > -- > Darren Wiebe > [EMAIL PROTECTED] > > Aleph Communications > www.aleph-com.net > > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/billing-platform-tp19549891p19578736.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] g729 sounds files - voicemail
I've pretty much standardized on using g729 as the codec of choice for the users connecting to my FS installation, primarily due to the users I have that are on very slow connections and all of my carriers support g729, so no need to transcode. The sounds files that comes with FS are raw PCM files. Since FS is unable to currently transcode to/from g729, is there any way to use sounds files that are in the g729 format, such that it is technically still "pass-through" on the FS side and no transcoding is required? Ideally, I'm looking to install pre-coded g729 files for voicemail and have FS playback those files when someone logs into voicemail. I haven't tried simply replacing the files in the sounds directory, what I would prefer is to have a separate directory structure with g729 sound files and have FS detect the codec in use and automatically playback the files for the appropriate codec? Thanks Much... Gabriel Kuri ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Can you show me a side by side comparison of working vs non-working? /b On Sep 19, 2008, at 12:37 PM, Jon Bruel wrote: > No the far side is BroadWorks, and as I said in the first mail, the > only > difference between an accepted and a rejected INVITE is in the > contents > of the Remote-Party-ID header. Generally in many European countries, > the > operators limit the A-number to the actual number of the line dialled > out on. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
No the far side is BroadWorks, and as I said in the first mail, the only difference between an accepted and a rejected INVITE is in the contents of the Remote-Party-ID header. Generally in many European countries, the operators limit the A-number to the actual number of the line dialled out on. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
I haven't encountered every event yet (I'm still in the process of writing my client) but perhaps some others can help? I'll post the ones I know about so far though.. - command/reply - CHANNEL_EXECUTE - CHANNEL_EXECUTE_COMPLETE - CHANNEL_ANSWER - CHANNEL_PARK The last two are uncertain because, in my view, they apply to channel state; something that the client can track independently without regard to whether it is related to a previously sent command. Note: I tend to use the word 'command' when referring to anything that the client is transmitting to mod_event_socket. I hope that doesn't confuse things - I'm not great on concise terminology. Killarny On Fri, Sep 19, 2008 at 10:45 AM, Michael Jerris <[EMAIL PROTECTED]> wrote: > > On Sep 19, 2008, at 12:19 PM, Luke Graybill wrote: > > > > On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill <[EMAIL PROTECTED]> wrote: > >> My suggested solution is to apply the job-id concept from bgapi to >> messages as well, and to go a step further; borrow the Asterisk idea of >> transmitting an identifier along with each command. Every response and event >> related to that command should then contain the very same identifier in the >> header. >> > > After speaking with bkw and MikeJ on irc, I'd like to further clarify my > suggestion to account for the fact that not all events apply directly to > specific commands, but for the ones which do (such as > CHANNEL_EXECUTE_COMPLETE, and for non-events, like command/reply) I believe > the suggestion stands :) > > Killarny > ___ > > Can you catalog the specific events this would effect?Mike > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] billing platform
There should be nothing major keeping it from working on windows. It's based on perl and mysql. There are ports of both perl and mysql for windows. Darren Wiebe [EMAIL PROTECTED] Michael Jerris wrote: > On Sep 18, 2008, at 12:00 PM, xbipin wrote: > > >> are u interested in a paid implementation as im willing to post a >> bounty for >> it if there r others like us willing to use it with freeswitch. >> wikipbx is >> also good for a start but one thing i dont understand is if >> freeswitch is >> made for cross platform then y r other projects related to it >> limiting it to >> just linux? >> >> > > I would try contacting the guys from astpp via their site. I know he > is still in process of making it work with FreeSWITCH at all but I > can't imagine there being anything major keeping it from working on > windows. > > Mike > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications www.aleph-com.net ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] billing platform
As the author of ASTPP I'd be very interested in negotiating a bounty to speed things up. User authentication is working with Freeswitch as well as lcr and call rating. It's largely a matter of documenting and testing. Darren Wiebe [EMAIL PROTECTED] xbipin wrote: > are u interested in a paid implementation as im willing to post a bounty for > it if there r others like us willing to use it with freeswitch. wikipbx is > also good for a start but one thing i dont understand is if freeswitch is > made for cross platform then y r other projects related to it limiting it to > just linux? > > > > Jair Santos wrote: > >> I've been trying to install. The problem is the lack of documentation on >> how to integrate with FS. >> >> Jair Santos >> >> >> >> >> >> >> >>> -Original Message- >>> From: [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] On >>> Behalf Of xbipin >>> Sent: Thursday, September 18, 2008 3:25 AM >>> To: freeswitch-users@lists.freeswitch.org >>> Subject: [Freeswitch-users] billing platform >>> >>> >>> >>> has any1 been able to use ASTPP billing with freeswitch and >>> if so then my next question, is it possible to use it on a >>> win2k3 platform and not linux as i have freeswitch running on >>> a win2k3 server since a few days and works fine, i just need >>> a prepaid type of script that basically stops a call once its >>> credit is over. i will add the credit etc to the account manually. >>> >>> im ready to post a bounty if any1 willing to do it. the >>> script just needs to end the call once it computes that >>> remaining time is 0 based on available time which is based on >>> the cost per minute and available credit and avoid calling >>> again which needs to be done as soon as call is placed so the >>> call is never connected if account has no credit. >>> -- >>> View this message in context: >>> http://www.nabble.com/billing-platform-tp19549891p19549891.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> ___ >>> Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>> http://www.freeswitch.org >>> >>> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications www.aleph-com.net ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] billing platform
Jair Santos is correct, it's being used but the Freeswitch integration is new and lacking documentation. That's being worked on slowly. Darren Wiebe [EMAIL PROTECTED] Jair Santos wrote: > I've been trying to install. The problem is the lack of documentation on > how to integrate with FS. > > Jair Santos > > > > > > > >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On >> Behalf Of xbipin >> Sent: Thursday, September 18, 2008 3:25 AM >> To: freeswitch-users@lists.freeswitch.org >> Subject: [Freeswitch-users] billing platform >> >> >> >> has any1 been able to use ASTPP billing with freeswitch and >> if so then my next question, is it possible to use it on a >> win2k3 platform and not linux as i have freeswitch running on >> a win2k3 server since a few days and works fine, i just need >> a prepaid type of script that basically stops a call once its >> credit is over. i will add the credit etc to the account manually. >> >> im ready to post a bounty if any1 willing to do it. the >> script just needs to end the call once it computes that >> remaining time is 0 based on available time which is based on >> the cost per minute and available credit and avoid calling >> again which needs to be done as soon as call is placed so the >> call is never connected if account has no credit. >> -- >> View this message in context: >> http://www.nabble.com/billing-platform-tp19549891p19549891.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> ___ >> Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> http://www.freeswitch.org >> >> > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications www.aleph-com.net ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Using the Command API
In this case you wouldn't use XML RPC you would use mod_event_socket.
http://wiki.freeswitch.org/wiki/Mod_event_socket#SendMsg
telnet localhost 8021
auth ClueCon
sendmsg 32a5e11a-8649-11dd-bb78-fd02030a93ef
call-command: execute
execute-app-name: playback
execute-app-arg: /var/lib/freeswitch/sounds/hello.wav
/b
On Sep 19, 2008, at 11:51 AM, Klaus Teller wrote:
>
>> From my understanding the code for playing the audio should be:
>
> client.execute("freeswitch.api",
> new Object[] { "playback",
> "32a5e11a-8649-11dd-bb78-fd02030a93ef /var/lib/freeswitch/sounds/
> hello.wav" });
>
> Unfortunately, the latter doesn't work. I suspect it's because the
> call has been parked. What is the proper way to address this?
>
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[Freeswitch-users] Using the Command API
Hi,
I'm trying to figure out how to use the command API via XML-RPC. I want to do
the two following things one after the other: 1) place a call, and 2) play an
audio on the newly created call.
The Java code for placing a call (copied from wiki) works fine and is following:
XmlRpcClientConfigImpl config = new XmlRpcClientConfigImpl();
XmlRpcClient client = new XmlRpcClient();
config.setServerURL(new URL("http://192.168.50.70:8080/RPC2";));
config.setBasicUserName("freeswitch");
config.setBasicPassword("works");
client.setConfig(config);
String response= client.execute("freeswitch.api",
new Object[] { "originate",
"sofia/internal/1003 & park()" }) .toString();
>From my understanding the code for playing the audio should be:
client.execute("freeswitch.api",
new Object[] { "playback",
"32a5e11a-8649-11dd-bb78-fd02030a93ef /var/lib/freeswitch/sounds/hello.wav" });
Unfortunately, the latter doesn't work. I suspect it's because the call has
been parked. What is the proper way to address this?
Thanks,
Klaus.
--
GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen!
Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED]
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Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
We don't see much samples for PRI outbound dialing . India the line is Euro ISDN. Has anyone tested sangoma A101 cards with Openzap ? I am tring to build a front end web application , to dial using JS in FS which will dial a outbound no and bridge the call to the extension. Thank you Imthiyaz Original Message: - From: Martin Joseph [EMAIL PROTECTED] Date: Fri, 19 Sep 2008 09:20:45 -0700 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch On Sep 19, 2008, at 6:23 AM, Gopal krishnan wrote: > Hi, > > Basically I just want to test outbound alone with freeswitch, so I > can use extensions.conf in the conf directory rite? > -- I would forget about the asterisk dialplan then. It's very simple to configure an outbound SIP provider in the XML config for FS. Look here for setting up your outbound provider: http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing Look here for a simple example (look for dialplan): http://wiki.freeswitch.org/wiki/Home_PBX_Example I set up a simple outbound SIP tester from these two pages in very little time. Good luck, hope this helps, Marty ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org mail2web.com - Microsoft® Exchange solutions from a leading provider - http://link.mail2web.com/Business/Exchange ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
On Sep 19, 2008, at 12:19 PM, Luke Graybill wrote: On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill <[EMAIL PROTECTED]> wrote: My suggested solution is to apply the job-id concept from bgapi to messages as well, and to go a step further; borrow the Asterisk idea of transmitting an identifier along with each command. Every response and event related to that command should then contain the very same identifier in the header. After speaking with bkw and MikeJ on irc, I'd like to further clarify my suggestion to account for the fact that not all events apply directly to specific commands, but for the ones which do (such as CHANNEL_EXECUTE_COMPLETE, and for non-events, like command/reply) I believe the suggestion stands :) Killarny ___ Can you catalog the specific events this would effect? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
Oops, I forgot to mention anthm as well - he provided great feedback on irc! On Fri, Sep 19, 2008 at 10:19 AM, Luke Graybill <[EMAIL PROTECTED]> wrote: > > > On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill <[EMAIL PROTECTED]> wrote: > >> My suggested solution is to apply the job-id concept from bgapi to >> messages as well, and to go a step further; borrow the Asterisk idea of >> transmitting an identifier along with each command. Every response and event >> related to that command should then contain the very same identifier in the >> header. >> > > After speaking with bkw and MikeJ on irc, I'd like to further clarify my > suggestion to account for the fact that not all events apply directly to > specific commands, but for the ones which do (such as > CHANNEL_EXECUTE_COMPLETE, and for non-events, like command/reply) I believe > the suggestion stands :) > > Killarny > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
On Sep 19, 2008, at 6:23 AM, Gopal krishnan wrote: > Hi, > > Basically I just want to test outbound alone with freeswitch, so I > can use extensions.conf in the conf directory rite? > -- I would forget about the asterisk dialplan then. It's very simple to configure an outbound SIP provider in the XML config for FS. Look here for setting up your outbound provider: http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing Look here for a simple example (look for dialplan): http://wiki.freeswitch.org/wiki/Home_PBX_Example I set up a simple outbound SIP tester from these two pages in very little time. Good luck, hope this helps, Marty ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill <[EMAIL PROTECTED]> wrote: > My suggested solution is to apply the job-id concept from bgapi to messages > as well, and to go a step further; borrow the Asterisk idea of transmitting > an identifier along with each command. Every response and event related to > that command should then contain the very same identifier in the header. > After speaking with bkw and MikeJ on irc, I'd like to further clarify my suggestion to account for the fact that not all events apply directly to specific commands, but for the ones which do (such as CHANNEL_EXECUTE_COMPLETE, and for non-events, like command/reply) I believe the suggestion stands :) Killarny ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Is the far side Asterisk? If so then I suspect they aren't trusting the RPID you're sending. They'll have to trustrpid. /b On Sep 19, 2008, at 10:56 AM, Jon Bruel wrote: > None of the suggestions worked, and I still can't control the A-number > individually when bridging to multiple destinations. > I have tried to change the dialplan using > [origination_caller_id_number=1234] > As a part of the string in the bridge data, but an info after bridge > did > not show the variable. I also tried with an unknown variable such as > new_var=1234, but it did not show up after the bridge using the info > application. Do we have a bug, or has the format been changed.. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
None of the suggestions worked, and I still can't control the A-number individually when bridging to multiple destinations. I have tried to change the dialplan using [origination_caller_id_number=1234] As a part of the string in the bridge data, but an info after bridge did not show the variable. I also tried with an unknown variable such as new_var=1234, but it did not show up after the bridge using the info application. Do we have a bug, or has the format been changed.. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
Large pastes on the mailing list are ok... if they aren't over 100k :P /b On Sep 19, 2008, at 10:24 AM, Christian Jensen wrote: > Use the pastebin please. > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
Use the pastebin please.
On Sep 19, 2008, at 7:00 AM, "Richard Open Source" <[EMAIL PROTECTED]
> wrote:
> oops...sorry for the mess...
>
>
> import org.apache.mina.common.IoSession
> import java.util.concurrent.BlockingQueue
> import java.util.concurrent.Executors
> import java.util.concurrent.ExecutorService
> import java.util.concurrent.Future
> import java.util.concurrent.Callable
> import java.util.concurrent.ExecutionException
> import java.util.concurrent.TimeoutException
> import java.util.concurrent.TimeUnit
>
>
>
> class Session {
> private IoSession session
>private FSEventHandler handler
>private BlockingQueue msgQ
>private final ExecutorService executor =
> Executors.newSingleThreadExecutor()
> private final static long DEFAULT_TIMEOUT = 5000
>
>def data
>
>def Session(IoSession s, BlockingQueue q) {
>session = s
>msgQ = q
>}
>
>private def executeAndWait(Closure task, long timeout=0) {
>Future f = executor.submit(task as Callable)
>def result
>def boolean success = false
>try {
>if (timeout != 0) {
>result = f.get(timeout,
> TimeUnit.MILLISECONDS)
>} else {
>result = f.get()
>}
>if (result.code == CommandResult.OK) data =
> result.data
>} catch (ExecutionException e) {
>// Should log here
>} catch (TimeoutException e) {
>f.cancel(true)
>}
>
> return result
>}
>
>def answer() {
>def task = {
>def done = false
>def r = new CommandResult()
>sendMessage("answer")
> while (! done) {
> def m = msgQ.take()
>if ((m?.event?.Name ==
> "CHANNEL_EXECUTE_COMPLETE") && (m?.Application == "answer")) {
>done = true
>r.code = CommandResult.OK
>r.data = m
>}
>}
>return r
>}
>executeAndWait(task, DEFAULT_TIMEOUT)
>}
>
>def unset(var) {
>def task = {
>def done = false
>def r = new CommandResult()
>sendMessage("unset", var)
>while (! done) {
>def m = msgQ.take()
>if ((m?.event?.Name ==
> "CHANNEL_EXECUTE_COMPLETE")
>&&
> (m?.Application == "unset")
>&&
> (m?.ApplicationData == var)) {
>done = true
>r.code =
> CommandResult.OK
>r.data = m
>}
>}
>return r
>}
>executeAndWait(task, DEFAULT_TIMEOUT)
>}
>
>def queueDtmf(dtmfs) {
>def task = {
>def done = false
>def r = new CommandResult()
>sendMessage("queue_dtmf", dtmfs)
>while (! done) {
>def m = msgQ.take()
>if ((m?.event?.Name ==
> "CHANNEL_EXECUTE_COMPLETE")
>&&
> (m?.Application == "queue_dtmf")
>&&
> (m?.ApplicationData == dtmfs)) {
>done = true
>r.code =
> CommandResult.OK
>r.data = m
>}
>}
>return r
>}
>executeAndWait(task, DEFAULT_TIMEOUT)
>}
>
> /*
>def hangup() {
>def task = {
>def done = false
>def r = new CommandResult()
>sendMessage("hangup")
>while (! done) {
>def m = msgQ.take()
>if ((m?.event?.Name ==
> "CHANNEL_EXECUTE_COMPLETE")
>
Re: [Freeswitch-users] plz compile latest snapshot for windows along with msi
On Sep 19, 2008, at 10:46 AM, xbipin wrote: > > the FS site says FS supports TLS etc so wouldnt it be good if the > windows > binary were compiled with TLS as by default they r not so guys like > me who > actually download the msi and install and run it can atleast have TLS > support by default on windows platform rather than them trying to > make it > work and then later posting in mailing list as if its a bug etc, > thatway it > can save a lot of posts that r done repeatidely and also it would be > better > if ppl added to wiki as soon as they got some of their issues solved. > If i could just get the msi with TLS support then atleast i would > try some > advanced features and then later post it as wiki for all those who r > beginners and willing to use FS on windows. I have already created > the wiki > offline, just waiting to try out that tLS and then ill post it as a > wiki > under beginners guide for windows platform. > > As has been said before, we would welcome someone adding this. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialpaln
Thanks for the reply, I tried dialing a number and the pastebin link as follows, http://pastebin.freeswitch.org/5611 and also I found that after dialed I saw the oz dump 1 2 and I found that the state is dialing and after few seconds automatically it seems to hangup. oz dump 1 3 API CALL [oz(dump 1 3)] output: span_id: 1 chan_id: 3 physical_span_id: 1 physical_chan_id: 3 type: B state: HANGUP last_state: DIALING cid_date: cid_name: Extension 1002 cid_num: 1002 ani: 9841799874 aniII: dnis: rdnis: cause: NO_ANSWER the dialect i am using is euro any help would be appreciated. -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] plz compile latest snapshot for windows along with msi
the FS site says FS supports TLS etc so wouldnt it be good if the windows binary were compiled with TLS as by default they r not so guys like me who actually download the msi and install and run it can atleast have TLS support by default on windows platform rather than them trying to make it work and then later posting in mailing list as if its a bug etc, thatway it can save a lot of posts that r done repeatidely and also it would be better if ppl added to wiki as soon as they got some of their issues solved. If i could just get the msi with TLS support then atleast i would try some advanced features and then later post it as wiki for all those who r beginners and willing to use FS on windows. I have already created the wiki offline, just waiting to try out that tLS and then ill post it as a wiki under beginners guide for windows platform. Carlos Talbot wrote: > > I'm uploading the msi to the site I have write access to. It should be > sync'd up on the files.freeswitch.org tonight. Remember to backup your > config folder before uninstalling! > > Carlos > > > On Mon, Sep 15, 2008 at 10:40 AM, Michael Jerris <[EMAIL PROTECTED]> wrote: > >> >> On Sep 15, 2008, at 9:25 AM, xbipin wrote: >> >> > >> > hi, >> > >> > this is for the freeswitch developers, can u plz compile the latest >> > snapshots of freeswitch and provide the msi installer for the latest >> > snapshot coz i have been using the installer which was last compiled >> > on 15 >> > aug and its has been quiet some time that an updated installer is >> > available. >> > >> > I was using the last installer on my win2k3 server, works fine >> > except that >> > it crashes on shutdown and the sip client hears a very fast saying >> > IVR when >> > dialed the 5000 number and the IVR is broken up, meaning it can be >> > heard but >> > with some bits missing or lets say with jitter, any way to make rtp >> > packets >> > much smoother or audio much smoother as i was running openSBC on the >> > same >> > server and it routes packets pretty fine so its got nothing to do >> > with the >> > server or the route from the client to the server. >> > -- >> > View this message in context: >> http://www.nabble.com/plz-compile-latest-snapshot-for-windows-along-with-msi-tp19492923p19492923.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> I spoke to the guy who makes the msi files yesterday and he said he >> would get a newer copy up there. That being said, if you download >> msvc express edition and advanced installer, both free, you can make >> your own msi any time you like. >> >> >> Mike >> >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/plz-compile-latest-snapshot-for-windows-along-with-msi-tp19492923p19573834.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialpaln
no it wouldn't When the channel is not in use it's down. it's absent of any activity and not running in the state machine. This terminology is for the sake of the coder not the guy using the code. On Fri, Sep 19, 2008 at 8:40 AM, Brian West <[EMAIL PROTECTED]> wrote: > IDLE would make more sense. > > /b > > On Sep 19, 2008, at 8:22 AM, Michael Jerris wrote: > > > in openzap, state down is like on-hook... we should change that name > > maybe. > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
On Fri, Sep 19, 2008 at 9:02 AM, Jon Bruel <[EMAIL PROTECTED]> wrote:
>
> Mike, you answered: effective_caller_id_number is meant to be set on the
> a leg, not the b-leg.
> Well I need to control the "effective_caller_id_number" (or whatever it
> is called in the B-leg) individually for each location when I bridge to
> many destinations. In my case, I need to have different identities
> (A-numbers) for local calls and for external calls. I have followed the
> WIKI:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Calling_
> multiple_destinations, though. Anyhow how is this solved? Could it be
> possible to loopback to the dialplan as in Asterisk local channel or to
> go through some steps before dial attemps are made, but after the bridge
> application?
> Jon
>
>
>
vars inside {} in the very beginning of the string are global to each dial
string listed henceforth
vars inside [] prior to each channel in a , or | separated list are unique
to just the first subsequent dial string.
data="[origination_caller_id_number=1234]sofia/default/[EMAIL PROTECTED]
,[origination_caller_id_number=4321]sofia/default/[EMAIL PROTECTED]"
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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
The effective_caller_id_name/number variables can be set on the A leg but those will be used to override those values on the b-legs if they spawn off the a-leg /b On Sep 19, 2008, at 9:02 AM, Jon Bruel wrote: Mike, you answered: effective_caller_id_number is meant to be set on the a leg, not the b-leg. Well I need to control the "effective_caller_id_number" (or whatever it is called in the B-leg) individually for each location when I bridge to many destinations. In my case, I need to have different identities (A-numbers) for local calls and for external calls. I have followed the WIKI: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Calling_ multiple_destinations, though. Anyhow how is this solved? Could it be possible to loopback to the dialplan as in Asterisk local channel or to go through some steps before dial attemps are made, but after the bridge application? Jon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
Mike, you answered: effective_caller_id_number is meant to be set on the a leg, not the b-leg. Well I need to control the "effective_caller_id_number" (or whatever it is called in the B-leg) individually for each location when I bridge to many destinations. In my case, I need to have different identities (A-numbers) for local calls and for external calls. I have followed the WIKI: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Calling_ multiple_destinations, though. Anyhow how is this solved? Could it be possible to loopback to the dialplan as in Asterisk local channel or to go through some steps before dial attemps are made, but after the bridge application? Jon -- Message: 4 Date: Fri, 19 Sep 2008 09:22:51 -0400 From: Michael Jerris <[EMAIL PROTECTED]> Subject: Re: [Freeswitch-users] dialpaln To: freeswitch-users@lists.freeswitch.org Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote: > Hi, > > Since I am not able to make the outbound call, when I use this > command in the console, I used to get the all the 31 > channels , for a refrence I am posing one channel block here, > > span_id: 1 > chan_id: 31 > physical_span_id: 1 > physical_chan_id: 31 > type: B > state: DOWN > last_state: DOWN > cid_date: > cid_name: > cid_num: > ani: > aniII: > dnis: > rdnis: > cause: NONE > > In this I am seeing that the state seems to be DOWN, but when I try > wantouer status, it show as connected like , > in openzap, state down is like on-hook... we should change that name maybe. > Wanrouter Status: > > Device name | Protocol | Station | Status| > wanpipe1| AFT HDLC | N/A | Connected | > wanpipe2| AFT HDLC | N/A | Connecting| > > where I am using is Sangoma A102D card with hardware echo > cancellation. And when I try to dial from the softphone in the > console I get this as, > > [EMAIL PROTECTED]> 2008-09-19 18:38:58 [NOTICE] > switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] > [a75028c4-554a-473e-a8cc-8c52d7f72df4] > 2008-09-19 18:38:58 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() > Processing FreeSwitch->99841799874 in context default > 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 > switch_channel_set_name() New Channel OpenZAP/1:1/9841799874 > [bd977e0d-e078-4193-a005-d751bdb26db8] > 2008-09-19 18:39:28 [NOTICE] sofia.c:2705 sofia_handle_sip_i_state() > Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] > [ORIGINATOR_CANCEL] > 2008-09-19 18:39:28 [NOTICE] switch_ivr_originate.c:1321 > switch_ivr_originate() Hangup OpenZAP/1:1/9841799874 > [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > 2008-09-19 18:39:28 [INFO] mod_dptools.c:1814 > audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 > switch_core_session_thread() Session 1 (sofia/internal/[EMAIL PROTECTED] > ) Ended > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 > switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] > [CS_HANGUP] > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 > switch_core_session_thread() Session 2 (OpenZAP/1:1/9841799874) Ended > 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 > switch_core_session_thread() Close Channel OpenZAP/1:1/9841799874 > [CS_HANGUP] > > So where i am wrong, can you please correct me. > It looks like the hang-up is coming from the sip side of things, can you take a look at the sip trace and confirm there is really a cancel on sip. Mike -- next part -- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080 919/59c1da81/attachment-0001.html -- Message: 5 Date: Fri, 19 Sep 2008 18:53:27 +0530 From: "Gopal krishnan" <[EMAIL PROTECTED]> Subject: Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch To: freeswitch-users@lists.freeswitch.org Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi, Basically I just want to test outbound alone with freeswitch, so I can use extensions.conf in the conf directory rite? -- Thank you with regards, Gopal, -- next part -- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080 919/7951fd9a/attachment-0001.html -- Message: 6 Date: Fri, 19 Sep 2008 19:04:43 +0530 From: "Gopal krishnan" <[EMAIL PROTECTED]> Subject: Re: [Freeswitch-users] dialpaln To: freeswitch-users@lists.freeswitch.org Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi Mike, I changed the name in openzap.conf and also in default.xml, but the same thing persisting, this hangup I terminated from the softphone, its not like coming from the sip
Re: [Freeswitch-users] Asynchronous communication with FreeSWITCH's mod_event_socket
oops...sorry for the mess...
import org.apache.mina.common.IoSession
import java.util.concurrent.BlockingQueue
import java.util.concurrent.Executors
import java.util.concurrent.ExecutorService
import java.util.concurrent.Future
import java.util.concurrent.Callable
import java.util.concurrent.ExecutionException
import java.util.concurrent.TimeoutException
import java.util.concurrent.TimeUnit
class Session {
private IoSession session
private FSEventHandler handler
private BlockingQueue msgQ
private final ExecutorService executor =
Executors.newSingleThreadExecutor()
private final static long DEFAULT_TIMEOUT = 5000
def data
def Session(IoSession s, BlockingQueue q) {
session = s
msgQ = q
}
private def executeAndWait(Closure task, long timeout=0) {
Future f = executor.submit(task as Callable)
def result
def boolean success = false
try {
if (timeout != 0) {
result = f.get(timeout, TimeUnit.MILLISECONDS)
} else {
result = f.get()
}
if (result.code == CommandResult.OK) data = result.data
} catch (ExecutionException e) {
// Should log here
} catch (TimeoutException e) {
f.cancel(true)
}
return result
}
def answer() {
def task = {
def done = false
def r = new CommandResult()
sendMessage("answer")
while (! done) {
def m = msgQ.take()
if ((m?.event?.Name ==
"CHANNEL_EXECUTE_COMPLETE") && (m?.Application == "answer")) {
done = true
r.code = CommandResult.OK
r.data = m
}
}
return r
}
executeAndWait(task, DEFAULT_TIMEOUT)
}
def unset(var) {
def task = {
def done = false
def r = new CommandResult()
sendMessage("unset", var)
while (! done) {
def m = msgQ.take()
if ((m?.event?.Name ==
"CHANNEL_EXECUTE_COMPLETE")
&&
(m?.Application == "unset")
&&
(m?.ApplicationData == var)) {
done = true
r.code = CommandResult.OK
r.data = m
}
}
return r
}
executeAndWait(task, DEFAULT_TIMEOUT)
}
def queueDtmf(dtmfs) {
def task = {
def done = false
def r = new CommandResult()
sendMessage("queue_dtmf", dtmfs)
while (! done) {
def m = msgQ.take()
if ((m?.event?.Name ==
"CHANNEL_EXECUTE_COMPLETE")
&&
(m?.Application == "queue_dtmf")
&&
(m?.ApplicationData == dtmfs)) {
done = true
r.code = CommandResult.OK
r.data = m
}
}
return r
}
executeAndWait(task, DEFAULT_TIMEOUT)
}
/*
def hangup() {
def task = {
def done = false
def r = new CommandResult()
sendMessage("hangup")
while (! done) {
def m = msgQ.take()
if ((m?.event?.Name ==
"CHANNEL_EXECUTE_COMPLETE")
&&
(m?.Application == "queue_dtmf")
&&
(m?.ApplicationData == dtmfs)) {
done = true
Re: [Freeswitch-users] dialpaln
IDLE would make more sense. /b On Sep 19, 2008, at 8:22 AM, Michael Jerris wrote: > in openzap, state down is like on-hook... we should change that name > maybe. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge destinations
Also if the far side is Asterisk and they don't have trustrpid=yes then you have to set the param on the gatway: /b On Sep 19, 2008, at 8:20 AM, Michael Jerris wrote: > effective_caller_id_number is meant to be set on the a leg, not the b > leg. > > Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialpaln
Hi Mike, I changed the name in openzap.conf and also in default.xml, but the same thing persisting, this hangup I terminated from the softphone, its not like coming from the sip phone automatically. so Is there anything else I need to check? -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
Hi, Basically I just want to test outbound alone with freeswitch, so I can use extensions.conf in the conf directory rite? -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialpaln
On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote: Hi, Since I am not able to make the outbound call, when I use this command in the console, I used to get the all the 31 channels , for a refrence I am posing one channel block here, span_id: 1 chan_id: 31 physical_span_id: 1 physical_chan_id: 31 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE In this I am seeing that the state seems to be DOWN, but when I try wantouer status, it show as connected like , in openzap, state down is like on-hook... we should change that name maybe. Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT HDLC | N/A | Connected | wanpipe2| AFT HDLC | N/A | Connecting| where I am using is Sangoma A102D card with hardware echo cancellation. And when I try to dial from the softphone in the console I get this as, [EMAIL PROTECTED]> 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [a75028c4-554a-473e-a8cc-8c52d7f72df4] 2008-09-19 18:38:58 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing FreeSwitch->99841799874 in context default 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel OpenZAP/1:1/9841799874 [bd977e0d-e078-4193-a005-d751bdb26db8] 2008-09-19 18:39:28 [NOTICE] sofia.c:2705 sofia_handle_sip_i_state() Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-09-19 18:39:28 [NOTICE] switch_ivr_originate.c:1321 switch_ivr_originate() Hangup OpenZAP/1:1/9841799874 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-09-19 18:39:28 [INFO] mod_dptools.c:1814 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 1 (sofia/internal/[EMAIL PROTECTED] ) Ended 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 2 (OpenZAP/1:1/9841799874) Ended 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel OpenZAP/1:1/9841799874 [CS_HANGUP] So where i am wrong, can you please correct me. It looks like the hang-up is coming from the sip side of things, can you take a look at the sip trace and confirm there is really a cancel on sip. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge destinations
On Sep 19, 2008, at 9:07 AM, Jon Bruel wrote: > I have tested the option of adding channel variables to the bridge > string, and it does not work. > This dialplan works: > > > > > > > > While this one does not work, it is rejected by the gateway: > > > data="[effective_caller_id_number=45161061]sofia/gateway/ > 45161061/$1"/> > > > The INVITE Remote-Party-ID header differs. In the firs case it is: > "Extension 1000" > ;screen=yes;privacy=off > And in the second case it is: > "Extension 1000" [EMAIL PROTECTED]>;screen=yes;privacy=off > > Is this an error on my behalf, or do we have a bug? effective_caller_id_number is meant to be set on the a leg, not the b leg. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialpaln
Hi, Since I am not able to make the outbound call, when I use this command in the console, I used to get the all the 31 channels , for a refrence I am posing one channel block here, * span_id: 1 chan_id: 31 physical_span_id: 1 physical_chan_id: 31 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE* In this I am seeing that the state seems to be DOWN, but when I try wantouer status, it show as connected like , *Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT HDLC | N/A | Connected | wanpipe2| AFT HDLC | N/A | Connecting|* where I am using is Sangoma A102D card with hardware echo cancellation. And when I try to dial from the softphone in the console I get this as, [EMAIL PROTECTED]> 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/ [EMAIL PROTECTED] [a75028c4-554a-473e-a8cc-8c52d7f72df4] 2008-09-19 18:38:58 [INFO] mod_dialplan_xml.c:232 dialplan_hunt() Processing FreeSwitch->99841799874 in context default 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel OpenZAP/1:1/9841799874 [bd977e0d-e078-4193-a005-d751bdb26db8] 2008-09-19 18:39:28 [NOTICE] sofia.c:2705 sofia_handle_sip_i_state() Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-09-19 18:39:28 [NOTICE] switch_ivr_originate.c:1321 switch_ivr_originate() Hangup OpenZAP/1:1/9841799874 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-09-19 18:39:28 [INFO] mod_dptools.c:1814 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 1 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 2 (OpenZAP/1:1/9841799874) Ended 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel OpenZAP/1:1/9841799874 [CS_HANGUP] * So where i am wrong, can you please correct me. Thanks -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Possible problem in adding channel variables to the bridge destinations
I have tested the option of adding channel variables to the bridge string, and it does not work. This dialplan works: While this one does not work, it is rejected by the gateway: The INVITE Remote-Party-ID header differs. In the firs case it is: "Extension 1000" ;screen=yes;privacy=off And in the second case it is: "Extension 1000" ;screen=yes;privacy=off Is this an error on my behalf, or do we have a bug? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fs only in lan
On Sep 19, 2008, at 5:48 AM, Rocco Lucente wrote: Hello, mod_sofia don't start correctly when there isn't a internet connection. We can use fs only in lan (without internet connection)? Is there a particolar configuration for this situation? Regards, The default configuration includes stun server lookups to find the external address for sip and the automatic ip guesser will not work on a non internet connected machine. This only means you will have to manually set the sip-ip and rtp-it (and omit the ext-*-ip) settings on the sip profile. If you are only on one segment, you can eliminate all but the one internal sip profile. Mike___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
On Sep 19, 2008, at 8:45 AM, Wasim Baig wrote: On Fri, Sep 19, 2008 at 6:41 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote: Hi, Is there any way that Asterisk dialplan can be used for freeswitch, since there is a flle extensions.conf in PREFIX/conf directory, is th possible with this file I can write a normal asterisk dialplan so that it will hit the freeswitch. If possible how can it be done any examples or any wiki is avaialble. http://wiki.freeswitch.org/wiki/Mod_dialplan_asterisk make sure you load the mod first in modules.conf Also, make sure to understand a big difference between asterisk and FreeSWITCH. in FreeSWITCH the dialplan is not parsed as-you-go as it is in asterisk, so conceptually most ivr like applications will not work with mod_dialplan_asterisk. If you understand the limitations (and some of the additional functionality) its a useful getting to know path. Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] fs only in lan
Hello, mod_sofia don't start correctly when there isn't a internet connection. We can use fs only in lan (without internet connection)? Is there a particolar configuration for this situation? Regards, -- Rocco Lucente Vi preghiamo di considerare l'ambiente prima di stampare questa e-mail Please consider the environment before printing this e-mail ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
On Fri, Sep 19, 2008 at 6:41 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote: > Hi, > > Is there any way that Asterisk dialplan can be used for freeswitch, since > there is a flle extensions.conf in PREFIX/conf directory, is th possible > with this file I can write a normal asterisk dialplan so that it will hit > the freeswitch. If possible how can it be done any examples or any wiki is > avaialble. > http://wiki.freeswitch.org/wiki/Mod_dialplan_asterisk make sure you load the mod first in modules.conf -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as you scope creep, so shall we reap ... ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Asiterisk Dialplan for Freeswitch
Hi, Is there any way that Asterisk dialplan can be used for freeswitch, since there is a flle extensions.conf in PREFIX/conf directory, is th possible with this file I can write a normal asterisk dialplan so that it will hit the freeswitch. If possible how can it be done any examples or any wiki is avaialble. -- Thank you with regards, Gopal, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SCCP aka Skinny
[EMAIL PROTECTED] wrote: > There is no SCCP module for FS. CM only uses SCCP to talk to phones, > it uses either MGCP or SIP to talk to gateways. So if you have a > version that has SIP support (I believe > 4.0), then you could > connect CM to FS. It would be possible in this way to assign to ring fs at my deskphone number for example? I think I need fs to acts as a phone to register himself to CM. I don't want to change anything in CM. Thanks, C Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SCCP aka Skinny
There is no SCCP module for FS. CM only uses SCCP to talk to phones, it uses either MGCP or SIP to talk to gateways. So if you have a version that has SIP support (I believe > 4.0), then you could connect CM to FS. Cavalera Claudio Luigi wrote: > Hello, > is there a way to interconnect fs to a Cisco Call Manager which is > configured to speak SCCP protocol (aka Skinny) and not SIP? > I did not found a mod_SCCP in the docs :-) > Thanks, > Claudio > > > Internet Email Confidentiality Footer > - > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i > destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees above, > you may take no action based on it, and you may not copy or show it to > anyone; please reply to this e-mail and point out the error which has > occurred. > - > > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] SCCP aka Skinny
Hello, is there a way to interconnect fs to a Cisco Call Manager which is configured to speak SCCP protocol (aka Skinny) and not SIP? I did not found a mod_SCCP in the docs :-) Thanks, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialpaln
Zaptel isn't required for sangoma. Just make sure when installing wanpipe the value you select is the TDM API with the words FreeSWITCH in the menu option. /b On Sep 19, 2008, at 12:57 AM, Gopal krishnan wrote: Hi Anthony, My conf as follows, openzap.conf [span wanpipe] trunk_type => e1 b-channel => 1:1-15 d-channel=> 1:16 b-channel => 1:17-31 openzap.conf.xml ztcfg -vv Still I have not installed zaptel, I have went thru that for openzap with sangoma no need of zaptel, so I never installed zaptel, and also I am not sure whether zaptel is required. Can I try installing zaptel? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
