Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?
I didn't say i have a working FS on blackfin... i just said i've ported a lot of software to blackfin and it was always floating point, fork vs vfork ... main issues... but why do you think it cannot be done? T. On Mon, Sep 21, 2009 at 6:08 AM, Hadley Rich h...@nice.net.nz wrote: On Mon, 21 Sep 2009 15:58:33 Juan Backson wrote: Are you able to have freeswitch working on blackfin platform? This has been covered many times on the list now, currently the answer is no. hads -- https://nicegear.co.nz VoIP and Open Source Hardware ___ 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] Call Tracing
In that case you should turn on sip trace for profile where your callcentric
peer is configured. By default FS comes with two profiles namely internal
and external. If you haven't created any new profile and configured your
users and peers in these two profiles then you should try turning on sip
trace for external profile too (or just external profile alone).
*sofia profile external siptrace on*
Please check your peer configuration and turn on sip trace on appropriate
profile.
Thank you.
On Sun, Sep 20, 2009 at 5:49 PM, Klaus Teller klaus.tel...@gmx.net wrote:
Thanks. I tried that and what it shows me is the trace between my peer and
the SIP provider (i.e. les.net). The call is actually coming from
callcentric and i don't see that in the trace. Is it supposed to show this?
Klaus.
Original-Nachricht
Datum: Sun, 20 Sep 2009 17:11:50 +0600
Von: Muhammad Shahzad shaherya...@googlemail.com
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Call Tracing
there are a few variable that you can set in
/usr/local/freeswitch/conf/vars.xml.
* X-PRE-PROCESS cmd=set data=call_debug=false/
X-PRE-PROCESS cmd=set data=console_loglevel=info/
*
You can change it to something like (and then restart FS),
* X-PRE-PROCESS cmd=set data=call_debug=true/
X-PRE-PROCESS cmd=set data=console_loglevel=debug/
*
Usually it will give you enough information about call processing,
however
just in case you are looking for SIP trace of a call only then you can
enable it on per-profile basis at run-time,
for example,
*sofia profile internal siptrace on*
this will enable SIP trace for all calls to / from sofia internal profile
(which also includes directory users).
You can run following command on FS console to get information on what
profile etc. are available as well as their status.
*sofia status*
For more info consult Wiki page at,
http://wiki.freeswitch.org/wiki/Sofia
Thank you.
On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi T.,
I just tried that but i don't see anything different on the console. My
test call is going via callcentric and les.net, but besides the final
hop
which i normally see in the channel name, there is nothing else.
Any idea what i might be doing wrong here?
Thanks,
Klaus.
Original-Nachricht
Datum: Sun, 20 Sep 2009 10:33:01 +0200
Von: Tihomir Culjaga tculj...@gmail.com
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Call Tracing
switch.conf.xml (btw: in console you can enable/disable logging on
the
fly
-
F8/F7)
param name=loglevel value=debug/
your relevant sip profile:
param name=sip-trace value=yes/
T.
On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi,
Say i have an inbound VoIP/SIP call that hits my FS box. Is it
possible
to
to extract information about the intermediate hops that the call or
the
signaling went through? If so, what information can i get?
Thanks,
Gregoire.
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Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?
Or as a more affordable solution... is it possible to connect an entry-level GSM phone to a PC running Freeswitch and use this as a poor man's gateway? -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530241.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] Any FreeSWITCH training courses out there?
With help from Pakistan Software Export Board (PSEB), we formed Asterisk Pakistan community forum in early 2008. This forum is still active and we arranged many workshops during last 18 months in all major cities of Pakistan. It was a great success and we effectively introduced Asterisk in so many government and private sectors. FreeSWITCH is very new in Pakistan and a very few people have heard its name here right now. So, we (me and some of my friends from Pakistan Open Source Software Foundation) are trying to develop some skilled personals for FreeSWITCH, before we approach Ministry of Information Technology to launch a campaign similar to Asterisk Pakistan Forum for FreeSWITCH. So, that if our proposal gets approval we would have enough resources to execute workshops all over Pakistan for FS training. All people in this mailing list (especially Pakistanis) who are interested in this, may contact me off list for participation and coordination in these efforts. The goal is to secure greatest share for Pakistan in this newly emerging technology and its benefits. Thank you. On Mon, Sep 21, 2009 at 9:53 AM, Mitul Limbani mi...@enterux.com wrote: Gavin, Sorry for the earlier mail, I can see that you mentioned Asterisk to Freeswitch course, we have pretty much under gone the same cycle and have put that as the part of our training course, it's named: FreeSWITCH for AstMasters Please do get in touch off the list, also if anyone else is interested in this course do get in touch with me. Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 21-Sep-2009, at 1:17 AM, Gavin Henry gavin.he...@gmail.com wrote: Hi all, Is there anyone out there doing beginner courses or conversion courses from an Asterisk mindset? Cheers. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.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 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.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] Mod_perl or ESL
How far do you want things to scale? /b On Sep 19, 2009, at 4:34 AM, Nagalenoj wrote: Dear friends, I want to know which is the better way to do route calls and control calls. I've did a experiment which can be done in both ways, Mod_perl and ESL. I don't know which one is better to take. When I see some earlier posts, It is given like Mod_perl has some limitations and I don't know what kind of limitations they are., ___ 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] Not able to make call using external profile
If you refer to the latest internal.xml in the default config for sip profiles you'll see an example of how to use a single profile for phones inside and outside of NAT. So you no longer have to have two profiles thus cutting the confusion level to almost zero when you setup FreeSWITCH to talk inside and outside of nat. Key elements are local-network-acl, ext-sip-ip and ext-rtp-ip and you're all set. /b On Sep 19, 2009, at 2:04 AM, Tihomir Culjaga wrote: another issue you might have with RTP so check the wiki for NAT config as well. ___ 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] User Creation with DB in Freeswitch
You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: Yes use odbc in fs Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.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] How to set the IP of REGISTER message??
On Mon, Sep 21, Brian West wrote: No you no longer have to do this. Please refer to the internal.xml profile in the default config. If you set the local-network-acl and then set ext-sip-ip and ext-rtp-ip then the profile will figure out which IP to use based on the destination or source of the request/ response. With two interfaces? Isn't it required for both of the interfaces to be bound to sofia? If that is true then isn't it only possible to bind one address per profile? --FC ___ 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] Can I set From: field in originate command?
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Re: [Freeswitch-users] User Creation with DB in Freeswitch
Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong registrations (bad username or password) in less then 50 seconds (49496 ms to be exact) and it processed all of them and gave correct responses using XML CURL. I am willing to do this test again soon, with correct registration data this time, to see how many registration Sofia SIP module configured with XML CURL module can handle at a time. Thank you. On Mon, Sep 21, 2009 at 3:42 PM, Brian West br...@freeswitch.org wrote: You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: Yes use odbc in fs Thanks Regards,Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.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] User Creation with DB in Freeswitch
I searched my sent emails and found the results, copying it below (after removing some sensitive info), 1,000 Calls == Total 1000 REGISTER calls sent in 890 ms at rate of 1123/sec Total 1000 responses receieved in 4516 ms at rate of 221/sec: Detailed responses received: - 403 responses: 1000 (Forbidden) -- TOTAL responses: 1000 (rate=221/sec) Maximum outstanding job: 894 Peak memory size: 15MB 5,000 Calls == Total 5000 REGISTER calls sent in 28539 ms at rate of 175/sec Total 5000 responses receieved in 36398 ms at rate of 137/sec: Detailed responses received: - 403 responses: 5000 (Forbidden) -- TOTAL responses: 5000 (rate=137/sec) Maximum outstanding job: 1001 Peak memory size: 63MB 10,000 Calls == Total 1 REGISTER calls sent in 60741 ms at rate of 164/sec Total 9289 responses receieved in 62740 ms at rate of 148/sec: Detailed responses received: - 403 responses: 9289 (Forbidden) -- TOTAL responses: 9289 (rate=148/sec) Maximum outstanding job: 1047 Peak memory size: 78MB 12,000 Calls == Total 12000 REGISTER calls sent in 49496 ms at rate of 242/sec Total 12314 responses receieved in 60582 ms at rate of 203/sec: Detailed responses received: - 403 responses:12314 (Forbidden) -- TOTAL responses:12314 (rate=203/sec) Maximum outstanding job: 1018 Peak memory size: 143MB So, FS doesn't crash even on 12,000 bad registrations (600 regs per second). I did tweak its configurations a little however no change was made to source code to make this happen. :-) Thank you. On Mon, Sep 21, 2009 at 4:07 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong registrations (bad username or password) in less then 50 seconds (49496 ms to be exact) and it processed all of them and gave correct responses using XML CURL. I am willing to do this test again soon, with correct registration data this time, to see how many registration Sofia SIP module configured with XML CURL module can handle at a time. Thank you. On Mon, Sep 21, 2009 at 3:42 PM, Brian West br...@freeswitch.org wrote: You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: Yes use odbc in fs Thanks Regards,Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.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] Affordable GSM gateway for one cellphone?
its a waste of time ... i doubt it can be done. T. On Mon, Sep 21, 2009 at 10:56 AM, Fred-145 codecompl...@free.fr wrote: Or as a more affordable solution... is it possible to connect an entry-level GSM phone to a PC running Freeswitch and use this as a poor man's gateway? -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530241.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 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] Affordable GSM gateway for one cellphone?
Check out this range http://www.noblesolutions.co.uk/shop/index.php?main_page=indexmanufactu rers_id=16 You should be able to find a local supplier We've used them for a couple of years now. They just plug into your network. Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Fred-145 Sent: 19 September 2009 11:34 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? Hello I'm selling a basic solution for SOHO customers (FS is installed on their work computer running Windows or Macs) to handle an analog phone line. When they're on the road, in addition or instead of getting a notification by e-mail when someone calls their office, some users might want to have the Freeswitch server actually ring their cellphone so they can take calls. Besides taking a subscription with a VoIP provider that the Freeswitch server will use to ring their cellphone, I'd like to know what my options are when it comes to setting up a GSM gateway on the customer's premises, in case they don't want to depend on the Internet. Are there Freeswitch-compatible, affordable solutions to handle a single GSM subscription? I guess all it takes is having them take a second subscription with their GSM provider and inserting the SIM chip inside the gateway to have Freeswitch ring their cellphone, but I've never used those things. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp255204 04p25520404.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 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] User Creation with DB in Freeswitch
Whoah what a term - ONCE BOMBED FS... Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong registrations (bad username or password) in less then 50 seconds (49496 ms to be exact) and it processed all of them and gave correct responses using XML CURL. I am willing to do this test again soon, with correct registration data this time, to see how many registration Sofia SIP module configured with XML CURL module can handle at a time. Thank you. On Mon, Sep 21, 2009 at 3:42 PM, Brian West br...@freeswitch.org wrote: You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: Yes use odbc in fs Thanks Regards,Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.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 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] Can I set From: field in originate command?
well first off you would setup a gateway and set the param 'from- domain' to what you wish it to be. /b On Sep 21, 2009, at 5:44 AM, Mikhail Krivushin wrote: I see that call_id_number placed in From:, but with wrong realm. I need a way for change realm in From:. Is any ability to do that? (I need to make calls over some telco from different accounts.) ___ 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] How to set the IP of REGISTER message??
yes but you can lie about IP's in the via/to and from if you set the local-network-acl ... I'm not talking two physical interfaces on FreeSWITCH... because that is one of the harder scenarios to setup... I'm talking single interface on FS sitting behind a nat router which is the most common. /b On Sep 21, 2009, at 4:45 AM, Frank Carmickle wrote: With two interfaces? Isn't it required for both of the interfaces to be bound to sofia? If that is true then isn't it only possible to bind one address per profile? --FC ___ 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] Affordable GSM gateway for one cellphone?
Fred-145 wrote: Hello I'm selling a basic solution for SOHO customers (FS is installed on their work computer running Windows or Macs) to handle an analog phone line. When they're on the road, in addition or instead of getting a notification by e-mail when someone calls their office, some users might want to have the Freeswitch server actually ring their cellphone so they can take calls. Besides taking a subscription with a VoIP provider that the Freeswitch server will use to ring their cellphone, I'd like to know what my options are when it comes to setting up a GSM gateway on the customer's premises, in case they don't want to depend on the Internet. Are there Freeswitch-compatible, affordable solutions to handle a single GSM subscription? I guess all it takes is having them take a second subscription with their GSM provider and inserting the SIM chip inside the gateway to have Freeswitch ring their cellphone, but I've never used those things. Thank you. I have been using http://www.portech.com.tw/p3-product1_1.asp?Pid=13 for years with Asterisk and Freeswitch. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530400.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] skill-based ACD
On Sun, Sep 20, 2009 at 08:42:40PM +0200, Remko Kloosterman wrote: This actually sounds very good Andrew. You even have an agent interface. Do you have plans for a outbound campaign dialer? I know of a commercial dialer that is good in it's predictive algotithm, but very bad when it comes to campaign management. I don't have plans for an 'autodialer' in the traditional sense but I do have plans for some sort of campaign dialer - the idea is to use an API to load numbers to be called into a queue and the agents will just pop those stub calls off the queue and then the system will originate the call to the indicated number. This does mean that you'll be wasting agent time on voicemail/ringouts/whatever but hopefully you'll piss less people off. In addition, then you can farm out the system that decides the numbers to be called and in which order to an external system. An autodialer would certainly be possible under the current system, I just don't really care to implement one. Patches accepted, although really an autodialer might be better off remaining a binary-only module add-on (to prevent the doing of evil becoming too cheap :) ). And yes, to my knowledge it will remain under an open-source license for the forseeable future. Andrew ___ 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] Compile error
I' ve get the same error with a fresh tree Thanks in advance De: Brian West br...@freeswitch.org Para: freeswitch-users@lists.freeswitch.org Enviado: jueves 17 de septiembre de 2009, 10:12:36 Asunto: Re: [Freeswitch-users] Compile error NO you must not. The issue has been fixed in svn already please start with a fresh tree. /b PS: end users should NEVER have to reswig. On Sep 17, 2009, at 12:42 AM, Frank Carmickle wrote: On Thu, Sep 17, Luis M. Zuccolo wrote: Hi: Since svn version 13523 to current I get this error: make[5]: swig: Command not found You must install swig. If your on debian apt-get install swig. If your not see http://www.swig.org/ HTH --FC Yahoo! Cocina Encontra las mejores recetas con Yahoo! Cocina. http://ar.mujer.yahoo.com/cocina/___ 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] Displaying matched extension during a call
Hi All, I am new to Freeswitch. So please bear with me if I ask any silly questions. * Can anyone of you please tell me how to display the extension name which has matched an incoming/outgoing call. * And can you please elaborate what does 'action application=info/' mean. * Suppose we have set a variable in the extension of the dialplan XML. Is there anyway we can display this variable on CLI for our debugging purposes. Regards, -- Anil Kumar S. R. http://sranil.googlepages.com/ The best way to succeed in this world is to act on the advice you give to others. ___ 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] Displaying matched extension during a call
Hello Anil On Mon, Sep 21, Anil Kumar S. R. wrote: * Can anyone of you please tell me how to display the extension name which has matched an incoming/outgoing call. In the log you will find something like this 2009-09-21 14:36:15.574827 [INFO] mod_dialplan_xml.c:315 Processing fs-03977304 in context default * And can you please elaborate what does 'action application=info/' mean. Please see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info HTH --FC ___ 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] No ring tone while recording incoming call. Please help.
Hi,
I have trouble recording incoming calls with FreeSwitch.
I have followed the instruction from Misc. Dialplan Tools record session
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session)
It works well for outgoing calls, but I have the problem with incoming
calls.
The person who is calling does not hear ring tone, he hears just the silence
until
I pick up the phone. Everything else is working, we can talk, conversation
is recorded.
Here is a copy of my dialplan for incoming calls
/usr/local/freeswitch/conf/dialplan/public/voipms.xml
include
extension name=voipms !-- your provider or any name you'd like to
call it --
condition field=destination_number expression=XX !--
your DID for this gateway--
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 2009/
action application=set data=RECORD_SOFTWARE=FreeSwitch/
action application=set data=RECORD_ARTIST=FreeSwitch/
action application=set data=RECORD_COMMENT=FreeSwitch/
action application=set data=RECORD_DATE=${strftime(%Y-%m-%d
%H:%M)}/
action application=set data=RECORD_STEREO=true/
action application=set data=RECORD_ANSWER_REQ=true/
action application=set data=ringback=${us-ring}/
action application=record_session
data=$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=bridge data=user/us...@${domain_name}/
/condition
/include
for outcoming calls I have a similar code added to the
/usr/local/freeswitch/conf/dialplan/default/user1.xml and it works well.
I have tried to move the line
action application=set data=ringback=${us-ring}/
between the lines
action application=record_session
and
action application=bridge
but it did not solve my problem.
Any ideas what am I doing wrong and how to fix it?
Igor
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Re: [Freeswitch-users] Displaying matched extension during a call
On Mon, Sep 21, 2009 at 11:27 AM, Anil Kumar S. R. sra...@gmail.com wrote:
Hi All,
I am new to Freeswitch. So please bear with me if I ask any silly
questions.
* Can anyone of you please tell me how to display the extension name which
has matched an incoming/outgoing call.
* And can you please elaborate what does 'action application=info/'
mean.
* Suppose we have set a variable in the extension of the dialplan XML. Is
there anyway we can display this variable on CLI for our debugging purposes.
Anil,
Here are few links to get started:
Handy tutorial: http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
Chan vars: http://wiki.freeswitch.org/wiki/Channel_Variables
Log app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log
Note: the info app dumps all sorts of information to the console and is a
great way to learn about many of the channel variables that FS has. The log
app will make it easy for you to pinpoint just a single channel variable:
action application=log data=INFO Dialed extension is ${dialed_ext}/
Have fun!
-MC
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Re: [Freeswitch-users] No ring tone while recording incoming call. Please help.
set ringback before record_session and also set transfer_ringback
because record_session causes an pre-answer.
/b
On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:
Hi,
I have trouble recording incoming calls with FreeSwitch.
I have followed the instruction from Misc. Dialplan Tools record
session
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session)
It works well for outgoing calls, but I have the problem with
incoming calls.
The person who is calling does not hear ring tone, he hears just the
silence until
I pick up the phone. Everything else is working, we can talk,
conversation is recorded.
Here is a copy of my dialplan for incoming calls
/usr/local/freeswitch/conf/dialplan/public/voipms.xml
include
extension name=voipms !-- your provider or any name you'd
like to call it --
condition field=destination_number
expression=XX !-- your DID for this gateway--
action application=set data=RECORD_TITLE=Recording $
{destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:
%M)}/
action application=set data=RECORD_COPYRIGHT=(c)
2009/
action application=set
data=RECORD_SOFTWARE=FreeSwitch/
action application=set
data=RECORD_ARTIST=FreeSwitch/
action application=set
data=RECORD_COMMENT=FreeSwitch/
action application=set data=RECORD_DATE=${strftime
(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_STEREO=true/
action application=set data=RECORD_ANSWER_REQ=true/
action application=set data=ringback=${us-ring}/
action application=record_session data=$${base_dir}/
recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$
{destination_number}_${caller_id_number}.wav/
action application=bridge data=user/us...@$
{domain_name}/
/condition
/include
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[Freeswitch-users] CudaTel Communications Server Version 1.0 Released
At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the FreeSWITCH team have been working together to create a new PBX appliance. Dubbed the CudaTel Communications Server, this new communications platform is both feature-rich and easy-to-use. We are pleased to announce that version 1.0 of the CudaTel Communcations Server has been released! The feature list for this affordable system is impressive: Automatic phone provisioning Multi-party conferencing Group calling SIP phone and provider support Automated attendant Voicemail TMD hardware option High definition codec support (G.722, G.722.1, G.722.1c) Call recording Active Directory and LDAP integration Encrypted VoIP support Many more features are included, all of which are controlled by an intuitive Web-based interface. We invite you to visit the CudaTel http://www.cudatel.com/ website or call 989-720-4000 for more information or to request evaluation units. -The FreeSWITCH Team ___ 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] BLF docs/howto?
Have not found anything usable in the wiki/mail list archives. I'm trying to setup BLF (busy lamp field) for Grandstream GXP-2000 phone. It offers BLF/eventlist BLF modes. Does Freeswitch supports both including the latter (RFC4662)? How to setup BLF on Freeswitch side? Are there any examples? - Dmitry Bely ___ 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] recompile with gdb
Hi Guys,
I have an issue running FS... it crashes apparently without leaving any log
... not even a core dump is left.
The machine is dual AMD opteron quad core with 8 GB RAM and i'm running 75
simultaneous calls (with media) with a rate of 5 calls per second.
As i was not able to reproduce the issue on a real traffic so i went back to
sipp and started generating some... sipp scenario files are ok.
after a while (few minutes)... on sipp i start getting retransmissions and
when i check FS i see two situations:
1. freeswitch has died
2. freeswitch process is running but it doesn't respond to any call... as
nothing has been sent ... and after a while it dies too.
I'm using sip profile external (moved to port 5060) with some semi-complex
dialplan... attached.
well .. the point is that i cannot even tell where it crashes as there is no
log.
I have:
param name=loglevel value=debug/
X-PRE-PROCESS cmd=set data=call_debug=true/
X-PRE-PROCESS cmd=set data=console_loglevel=debug/
fs is dumping the log to the log directory ... but nothing special can't bee
seen there...
I tried to recompile with gdb
export CFLAGS=-g -ggdb
export MOD_CFLAGS=-g -ggdb
./configure
but without luck...
ode1:/opt/freeswitch-trunk#
node1:/opt/freeswitch-trunk# sudo make
make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre'
make all-am
make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre'
g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF
.deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc
g++: Internal error: Segmentation fault (program cc1plus)
Please submit a full bug report.
See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions.
make[2]: *** [pcrecpp_unittest.o] Error 1
make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre'
make: *** [libs/pcre/libpcre.la] Error 2
node1:/opt/freeswitch-trunk#
node1:/opt/freeswitch-trunk#
Of course I'm using the latest trunk...
Can anyone help?
include
extension name=VAS
condition field=destination_number expression=^0(\d+)$
action application=log data=INFO Entering VAS \n/
action application=execute_extension data=0$1_priceAdvice XML public/
action application=execute_extension data=0$1_serviceDiscriminator XML public/
action application=hangup data=NORMAL_CLEARING/
/condition
/extension
extension name=priceAdvice
condition field=destination_number expression=(\d+)_priceAdvice$
action application=log data=INFO Price Adviced \n/
!--action application=getServiceTypeID_db data=in $1, out service_type_id/--
action application=set data=service_type_id=1/
action application=pre_answer/
!--action application=getPricePrompt_db data=in $1, in ${caller_id_number} , out price_prompt/--
action application=set data=price_prompt=4.93kn_novo_upozorenje.wav/
action application=playback data=vas/${price_prompt}/
!--action application=sched_hangup data=+${cond(${regex($1|3856(\d)\d+|%1)} == 8 ? 120 : 3600)}/--
action application=sleep data=2000/
/condition
/extension
extension name=ServiceDiscriminator
condition field=destination_number expression=(\d+)_serviceDiscriminator$
action application=log data=INFO Service Discriminator \n/
!--action application=getServiceTypeID_db data=in $1, out service_type_id/--
action application=set data=dialed_number=$1/
action application=log data=INFO ### service_type_id = '${service_type_id}' ##/
action application=log data=INFO ### dialed_number = '${dialed_number}' ##/
/condition
condition field=${service_type_id} expression=^1$ break=on-true
action application=log data=INFO KVIZ \n/
action application=execute_extension data=${dialed_number}_getVars_Kviz XML public/
/condition
/extension
extension name=getVars_Kviz
condition field=destination_number expression=(\d+)_getVars_Kviz$
action application=log data=INFO GetVars Kviz /
action application=set data=bNum=$1/
!--action application=getQuizServiceStatus_ch data=in $1, in ${caller_id_number}, out service_status1, out number_2_connect, out next_number_2_connect, out next_number_2_display/
action application=getServiceOutOfWorkingHoursPrompt_db data=in $1, out not_working_prompt/
action application=getServiceWinPrompt_db data=in $1, out service_win_prompt/
action application=getServiceLoosePrompt_db data=in $1, out service_loose_prompt/--
!--action application=sched_hangup data=+${cond(${regex($1|3856(\d)\d+|%1)} == 8 ? 120 : 3600)}/--
action
Re: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released
We invite you to visit the CudaTel http://www.cudatel.com/ website or call 989-720-4000 for more information or to request evaluation units. ___ 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] CudaTel Communications Server Version 1.0 Released
URL??? On 21/09/2009, Michael Collins m...@freeswitch.org wrote: At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the FreeSWITCH team have been working together to create a new PBX appliance. Dubbed the CudaTel Communications Server, this new communications platform is both feature-rich and easy-to-use. We are pleased to announce that version 1.0 of the CudaTel Communcations Server has been released! The feature list for this affordable system is impressive: Automatic phone provisioning Multi-party conferencing Group calling SIP phone and provider support Automated attendant Voicemail TMD hardware option High definition codec support (G.722, G.722.1, G.722.1c) Call recording Active Directory and LDAP integration Encrypted VoIP support Many more features are included, all of which are controlled by an intuitive Web-based interface. We invite you to visit the CudaTel http://www.cudatel.com/ website or call 989-720-4000 for more information or to request evaluation units. -The FreeSWITCH Team -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.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] CudaTel Communications Server Version 1.0 Released
On Tue, 22 Sep 2009 11:13:13 Gavin Henry wrote: URL??? On 21/09/2009, Michael Collins m...@freeswitch.org wrote: We invite you to visit the CudaTel http://www.cudatel.com/ website or call 989-720-4000 for more information or to request evaluation units. -- https://nicegear.co.nz VoIP and Open Source Hardware ___ 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] CudaTel Communications Server Version 1.0 Released
On Mon, Sep 21, 2009 at 4:19 PM, William Suffill william.suff...@gmail.comwrote: We invite you to visit the CudaTel http://www.cudatel.com/ website or call 989-720-4000 for more information or to request evaluation units. Hehe, thanks for pointing that out. Also, I said TMD hardware option when I really meant TDM hardware option :) -MC ___ 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] ALLOTTED_TIMEOUT hangup cause?
Hi, Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the Wiki and Google, but I couldn't find a detailed explanation. Does anybody know what does it mean exactly? Thanks! Nicolas ___ 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
