Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?
Allright, I tried this again now, with revision 13042 - it's the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I'll try to help out as much as possible. Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 15 april 2009 23:21 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:49e651b332933023977319! ___ 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 many simultaneous calls support in freeswitch
Hello I've recently tried to test freeswitch with default configurations (just added more users and a regexp match in internal.xml SIP switching in dialplan) and it performed quite surprisingly slow.. I noticed a large disk swapping activity (CPU at registrations of 50 clients at 100% load!) and I think it's because of all the default settings there (like creating voicemail files for every call etc..). At least I hope that's it. I will try making it much simpler and see.. Else there must be some other issue for sure.. I don't know if freeswitch has support for generating originating calls, but there sure is support for outbound connections in means of connecting to third party providers etc.. Afaik, there are two modules for cdr provided, check http://wiki.freeswitch.org/wiki/Cdr. Martin On Wed, Apr 15, 2009 at 4:06 PM, Brian West br...@freeswitch.org wrote: You have to determine how far it will scale for YOUR needs nobody can answer this question. It all depends on what YOU are doing with it and how crazy wild you go with things in your implementation. ;) /b On Apr 15, 2009, at 8:59 AM, Parveen Kumar Jain wrote: Hi, I need to develop an IVR application which makes an outbound calls and then plays the some audio file for the user. For this I was trying to evaluate Freeswitch under following criteria: - Does freeswitch have outbound calls support(is there any conf file file avilabel where I just can put some series of no. and freeswitch just calls those no. sequentially)? - If yes, how many simultaneous calls are possible on a simple pentium-4 using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked the switch.conf file and it says that by default it can support upto 1000 calls , is it true for a small machine also)? In other terms is Freeswitch is scalable if I need to add more users to call from here? - Does freeswitch have the support of CDR(call data record) after succesfull calls ? It will be great help if any of you can comment on these question and it can save my several hours of testing before I can make my conclusion :) Best Regards, Parveen Jain Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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
[Freeswitch-users] How does FS compare with OpenSIPs?
Hello There's an excellent article on FS vs. Asterisk, but unless I missed it, there's no equivalent to OpenSIPs (www.opensips.org). At this point, apart from the fact that OpenSIPs is not available for Windows, how does FreeSwitch compare with OpenSIPs, what are the strengths and weaknesses of each project? I need to know before recommending one or the other. Thank you for any feedback. ___ 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 does FS compare with OpenSIPs?
FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy. Diego On Thu, Apr 16, 2009 at 5:45 AM, Gilles codecompl...@free.fr wrote: Hello There's an excellent article on FS vs. Asterisk, but unless I missed it, there's no equivalent to OpenSIPs (www.opensips.org). At this point, apart from the fact that OpenSIPs is not available for Windows, how does FreeSwitch compare with OpenSIPs, what are the strengths and weaknesses of each project? I need to know before recommending one or the other. Thank you for any feedback. ___ 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] How does FS compare with OpenSIPs?
Diego Viola wrote: FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy. Thanks Diego. Based on this list features list, what does FS offer that OpenSIPs doesn't? http://www.opensips.org/index.php?n=Resources.Features I don't know enough about VoIP etc. to be able to tell, but at first sight, it seems like OpenSIPs doesn't really handle PBX features, which would be a strong point in favor of FreeSwitch. -- View this message in context: http://www.nabble.com/How-does-FS-compare-with-OpenSIPs--tp23074733p23076256.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] Skypiax as a windows service
Hi Giovanni, We tried every available fake audio driver (i.e. virtual audio cable) but with no satisfying results. As for the Skype as a service - that's not a problem. It's working fine - but only on Session 0 - which makes it inaccessible for the SkypeAPI from Skypiax. The problem is not unique to RDP - but most notable when running the Skypiax via RDP session. You'll be able to replicate this problem whenever the Skypiax is not running in the same session / userID of the Skype. Decoupling the Skyiax from FS will solve the problem as I assume it'll use TCP/IP (winsock) to interface with FS - therefore, I can run it still on the same machine but two separate sessions. However, I think getting the Skypiax to work as a service will be more beneficial regardless if it's decoupled or not. Keep up the good work, Cheers, UV -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Wednesday, April 15, 2009 12:28 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Skypiax as a windows service Hi UV, seems a difficult one this one. I have no much experience in RDP/terminal server. If there is no way to have (or fake) audio driver on RDP/terminal server apps, probably the Skype clients will not works (as you experienced). I'm sure, I've read it (:-) ), that Skype clients can be run on a Windows machine as services, without any user logged in. That is what I would explore in the future, just adding the How To to the wiki page. What you are experiencing seems to be different, seems to be specific to the RDP/terminal server usage. I'm I understanding you correctly (that this is specific to RDP)? Can you send me more info/hints? In parallel, I'm slowly working on a way to farm out the Skype clients from the FS servers, so to have the Skype clients running on different machines on the same LAN. I've a proof of concept working on Linux for one channel. You think this would solve your problems (having the Skype clients running on separate machines other than the machines running FS)? I'm just back from Easter vacations, please allow a couple days for the accumulated backlog ;-) Thanks a lot for taking the time to explore Skypiax and report this, gm Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Apr 13, 2009 at 1:32 PM, UV u...@yuvalhertzog.com wrote: Great work on Skypiax, Giovanni. Weve tested it in our lab for sometime and it works very well. Unfortunately, when we tried deploying it on a production environment (running Win2K3 server farm), we ran into a barrier: FS is running as terminal server console application (to be easily maintained remotely by RDP) This is because Win2K does not allow RDP to access system console (session /userid 0) Skype does not work on terminal server due to a well known disappearing audio drivers problem, therefore it has to run either as a console or a service (both on session 0). FS can run well as a windows service Skypiax seem to load as service, but it cant find the skype client and exit with the following error: 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev 13006M[|37 ][ERRORA 990 ][skype_user ][-1, 0, 0] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypiax exiting This situation prevents me to run skypiax in production. I understand from the wiki page that windows service is not done yet so I presume this is a predicted outcome. Any idea when and if this is planned to be implemented? Keep up the good work! Cheers, UV ___ 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 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/Sofia not sending bye when FreeSWITCH ends the call?
yes open a jira http://jira.freeswitch.org *attach* the following (do not paste it inline into the comments and give all trace files a .txt extension) repeat the trace you did earlier with more debugging enabled. type these 3 cli commands before you call sofia profile internal siptrace on sofia loglevel all 9 console loglevel debug On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson peter.ols...@visionutveckling.se wrote: Allright, I tried this again now, with revision 13042 – it’s the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I’ll try to help out as much as possible. Peter *Från:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *För *Brian West *Skickat:* den 15 april 2009 23:21 *Till:* freeswitch-users@lists.freeswitch.org *Ämne:* Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:49e651b332933023977319! ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] how many simultaneous calls support in freeswitch
It's not surprising to us. 90% of people who try to load test manage to do it in unnatural conditions on on inadequate hardware and end up saying something similar, then they always come back in 2 weeks with 2000 calls up. On Thu, Apr 16, 2009 at 3:36 AM, Martin Fiala fial...@gmail.com wrote: Hello I've recently tried to test freeswitch with default configurations (just added more users and a regexp match in internal.xml SIP switching in dialplan) and it performed quite surprisingly slow.. I noticed a large disk swapping activity (CPU at registrations of 50 clients at 100% load!) and I think it's because of all the default settings there (like creating voicemail files for every call etc..). At least I hope that's it. I will try making it much simpler and see.. Else there must be some other issue for sure.. I don't know if freeswitch has support for generating originating calls, but there sure is support for outbound connections in means of connecting to third party providers etc.. Afaik, there are two modules for cdr provided, check http://wiki.freeswitch.org/wiki/Cdr. Martin On Wed, Apr 15, 2009 at 4:06 PM, Brian West br...@freeswitch.org wrote: You have to determine how far it will scale for YOUR needs nobody can answer this question. It all depends on what YOU are doing with it and how crazy wild you go with things in your implementation. ;) /b On Apr 15, 2009, at 8:59 AM, Parveen Kumar Jain wrote: Hi, I need to develop an IVR application which makes an outbound calls and then plays the some audio file for the user. For this I was trying to evaluate Freeswitch under following criteria: - Does freeswitch have outbound calls support(is there any conf file file avilabel where I just can put some series of no. and freeswitch just calls those no. sequentially)? - If yes, how many simultaneous calls are possible on a simple pentium-4 using G711U as a codec(1 GB RAM, 2.2 GHz machine) machine?(I had checked the switch.conf file and it says that by default it can support upto 1000 calls , is it true for a small machine also)? In other terms is Freeswitch is scalable if I need to add more users to call from here? - Does freeswitch have the support of CDR(call data record) after succesfull calls ? It will be great help if any of you can comment on these question and it can save my several hours of testing before I can make my conclusion :) Best Regards, Parveen Jain Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] How does FS compare with OpenSIPs?
We try not to brag about ourselves or *sell* FreeSWITCH to people. A SIP proxy like openSIPS and a b2bua/media gateway like FreeSWITCH and meant to be used together. There is some overlap in SIP functionality but SIP is just one aspect of FreeSWITCH where SIP is the only thing OpenSIPS is for. That is why you find no comparison because it's like comparing a bus to a car they both have wheels they both can provide transportation but they serve different purposes. On Thu, Apr 16, 2009 at 7:16 AM, Karl Vesterling k...@ken-ton.com wrote: H.323 (mod_opal I think)Skype Jingle/Jabber (via mod_dingaling) Text to speach, speach recognition, and far too many to list. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 16, 2009, at 7:45 AM, Fred-145 wrote: Diego Viola wrote: FreeSWITCH is a B2BUA, OpenSIPS is a SIP proxy. Thanks Diego. Based on this list features list, what does FS offer that OpenSIPs doesn't? http://www.opensips.org/index.php?n=Resources.Features I don't know enough about VoIP etc. to be able to tell, but at first sight, it seems like OpenSIPs doesn't really handle PBX features, which would be a strong point in favor of FreeSwitch. -- View this message in context: http://www.nabble.com/How-does-FS-compare-with-OpenSIPs--tp23074733p23076256.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause
turn on the debug option in mod_cdr_csv and you will get something similar
to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
mik...@bjerkeland.com wrote:
El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland escribió:
Hi,
I have two scenarios I'm having trouble figuring out and I'd be happy
if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a forked
dial with the following:
action application=bridge
data=user/mikael-no...@voip.domain.com
,[leg_delay_start=10]openzap/1/a/99355151/
However the second leg is called at exactly the same time as the first
one. I am away from my testing environment right now, so I'm sorry for
not posting my logs. It appears to me that leg_delay_start is broken
on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge, but not on
specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the
call I'd like to continue in the dialplan. Currently I have the
following:
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/mikael-no...@voip.domain.com/
!-- I will only get here if the first bridge is rejected or
TODO: I get a MEDIA_TIMEOUT on it --
action application=bridge data=openzap/1/a/99355151/
Any ideas on how to accomplish this?
I started testing this with the following dialplan:
extension name=mikael-nokia+fallback
condition field=destination_number expression=^503$
action application=set data=hangup_after_bridge=false/
action application=set data=continue_on_fail=true/
action application=bridge
data=user/mikael-no...@fs.voip.domain.com/
action application=info/
action application=set data=followme_extension=99355151/
action application=execute_extension
data=post_call_followme_check/
action application=hangup/
/condition
/extension
extension name=post_call_followme_check
condition field=destination_number
expression=^post_call_followme_check$/
condition field=${originate_disposition}
expression=^MEDIA_TIMEOUT|$${continue_on_fail_causes}$
break=on-true
action application=log data=1 Follow me transferring call
because of orig disposition: ${originate_disposition}/
action application=transfer data=${followme_extension}/
/condition
condition
action application=log data=1 Follow me call ended normally
with orig disposition: ${originate_disposition}./
action application=hangup/
/condition
/extension
${originate_disposition} never has the value of MEDIA_TIMEOUT since the
call was answered, which is absolutely correct, so what I am searching
for now is how to get the actual hangup cause. The info app doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread()
sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253[BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253
]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/mikael-ek...@fs.voip.domain.com [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:508
switch_core_session_run()
(sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253
)
State EXCHANGE_MEDIA going to sleep
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:397
switch_core_session_run()
(sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253
)
Running State Change CS_HANGUP
EXECUTE sofia/internal/mikael-ek...@fs.voip.domain.com info()
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:448
switch_core_session_run()
(sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253
)
State HANGUP
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:315 sofia_on_hangup() Channel
sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253hanging
up, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG] mod_sofia.c:370 sofia_on_hangup() Sending
BYE to
sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253
2009-04-16 10:02:34 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup()
sofia/internal/sip:mikael-no...@10.247.3.253sip%3amikael-no...@10.247.3.253Standard
HANGUP, cause:
MEDIA_TIMEOUT
2009-04-16 10:02:34 [DEBUG]
[Freeswitch-users] FreeSwitch Complex IVR System
Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this? ___ 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] Entire Wiki.FreeSwitch.org on Single PDF ?
It may be worth looking at http://www.mediawiki.org/wiki/Extension:Pdf_Export or http://www.mediawiki.org/wiki/Extension:Pdf_Book -- Regards, Will Boyce m...@willboyce.com tel: 07933 515 987 url: http://willboyce.com - Michael Collins m...@freeswitch.org wrote: | From: Michael Collins m...@freeswitch.org | To: freeswitch-users@lists.freeswitch.org | Sent: Wednesday, 15 April, 2009 18:12:26 GMT +00:00 GMT Britain, Ireland, Portugal | Subject: Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ? | | Unfortunately this isn't being maintained and Bret didn't give his script to any of us. If anyone out there is familiar with converting wiki pages to PDF and is willing to pick this up then by all means contact me off list and we'll discuss it. | | -MC | | | On Wed, Apr 15, 2009 at 12:22 AM, Mitul Limbani mi...@enterux.com wrote: | Hello there, | | In my previous encounter with FreeSwitch, I had found that Bret had posted on the Mailing List somewhere about availability of the entire FreeSwitch Wiki Documentation on a single PDF, this is useful coz at the offset apart from Wiki there is no other offline media to learn it. | | Is the same PDF available looking at the growth of Wiki pages and the updation. | | I look forward to hear from you guys, | | Thanks Regards, | Mitul Limbani, | Founder CEO, | Enterux Solutions, | The Enterprise Linux Company (TM), | www.enterux.com | +91-9820332422 | | Msg sent via Enterux Enterprise Email Server : http://www.enterux.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 ___ 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] Call bridge in free switch
Hello, I am trying to find a way to this through fs_cli 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108) ClientC (1...@192.168.1.108) 2) Bridge all the 3 legs together into one call Thanks Prabhu ___ 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 bridge in free switch
Prabhuram Mohan pisze: Hello, I am trying to find a way to this through fs_cli 1) call out to ClientA (1...@192.168.1.108 mailto:1...@192.168.1.108), ClientB (1...@192.168.1.108 mailto:1...@192.168.1.108) ClientC (1...@192.168.1.108 mailto:1...@192.168.1.108) 2) Bridge all the 3 legs together into one call Thanks Prabhu The only way I know to do this is to use conference module. Then for example: conference test dial clientA conference test dial clientB conference test dial clientC ___ 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] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause
Thanks. I just tested and got some more data but it didn't contain any
variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
variable_hangup_cause and variable_originate_disposition contain
NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
the real reason for the hangup of the bridge, which in this case is
MEDIA_TIMEOUT as you can see from the logs.
El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
turn on the debug option in mod_cdr_csv and you will get something
similar to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
mik...@bjerkeland.com wrote:
El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
escribió:
Hi,
I have two scenarios I'm having trouble figuring out and I'd
be happy
if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a
forked
dial with the following:
action application=bridge
data=user/mikael-no...@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151/
However the second leg is called at exactly the same time as
the first
one. I am away from my testing environment right now, so I'm
sorry for
not posting my logs. It appears to me that leg_delay_start
is broken
on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge,
but not on
specific hangup causes. If I get a MEDIA_TIMEOUT hangup
cause in the
call I'd like to continue in the dialplan. Currently I have
the
following:
action application=set
data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=true/
action application=bridge
data=user/mikael-no...@voip.domain.com/
!-- I will only get here if the first bridge is
rejected or
TODO: I get a MEDIA_TIMEOUT on it --
action application=bridge
data=openzap/1/a/99355151/
Any ideas on how to accomplish this?
I started testing this with the following dialplan:
extension name=mikael-nokia+fallback
condition field=destination_number expression=^503$
action application=set
data=hangup_after_bridge=false/
action application=set
data=continue_on_fail=true/
action application=bridge
data=user/mikael-no...@fs.voip.domain.com/
action application=info/
action application=set
data=followme_extension=99355151/
action application=execute_extension
data=post_call_followme_check/
action application=hangup/
/condition
/extension
extension name=post_call_followme_check
condition field=destination_number
expression=^post_call_followme_check$/
condition field=${originate_disposition}
expression=^MEDIA_TIMEOUT|$${continue_on_fail_causes}$
break=on-true
action application=log data=1 Follow me transferring
call
because of orig disposition: ${originate_disposition}/
action application=transfer
data=${followme_extension}/
/condition
condition
action application=log data=1 Follow me call ended
normally
with orig disposition: ${originate_disposition}./
action application=hangup/
/condition
/extension
${originate_disposition} never has the value of MEDIA_TIMEOUT
since the
call was answered, which is absolutely correct, so what I am
searching
for now is how to get the actual hangup cause. The info app
doesn't show
MEDIA_TIMEOUT anywhere, but my logs show this:
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:377
audio_bridge_thread()
sofia/internal/sip:mikael-no...@10.247.3.253
ending bridge by request from read function
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
sofia/internal/sip:mikael-no...@10.247.3.253 [BREAK]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:452
audio_bridge_thread() BRIDGE THREAD DONE
[sofia/internal/sip:mikael-no...@10.247.3.253]
2009-04-16 10:02:34 [DEBUG] switch_ivr_bridge.c:456
audio_bridge_thread() Send signal
Re: [Freeswitch-users] leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause
I think I know a bit more about the problem now. The MEDIA_TIMEOUT
hangup cause is probably coming from the B leg of the call and thus not
visible when I do info or debug on mod_cdr_csv.
I then tried the following after bridge to get it:
action application=set data=other_leg_hangup_cause=
${uuid_getvar(${bridge_uuid} hangup_cause)}/
However, since that bridge of the call is already hung up I got the
following in reply:
variable_other_leg_hangup_cause: [-ERR No Such Channel!
]
Is there a way to get it from the B leg of the call - assuming that's
where the hangup cause comes from?
Thanks!
El jue, 16-04-2009 a las 16:07 +0200, Mikael Aleksander Bjerkeland
escribió:
Thanks. I just tested and got some more data but it didn't contain any
variable containing MEDIA_TIMEOUT. Perhaps it's not really set anywhere?
variable_hangup_cause and variable_originate_disposition contain
NORMAL_CLEARING and SUCCESS respectively. I need a var which contains
the real reason for the hangup of the bridge, which in this case is
MEDIA_TIMEOUT as you can see from the logs.
El jue, 16-04-2009 a las 07:37 -0500, Anthony Minessale escribió:
turn on the debug option in mod_cdr_csv and you will get something
similar to the info app only at the end of the call
On Thu, Apr 16, 2009 at 3:19 AM, Mikael Aleksander Bjerkeland
mik...@bjerkeland.com wrote:
El mié, 15-04-2009 a las 17:43 +0200, Mikael Bjerkeland
escribió:
Hi,
I have two scenarios I'm having trouble figuring out and I'd
be happy
if someone could tell me what I'm doing wrong.
1. leg_delay_start=N not working
I am trying to delay the origination of the second leg in a
forked
dial with the following:
action application=bridge
data=user/mikael-no...@voip.domain.com,[leg_delay_start=10]openzap/1/a/99355151/
However the second leg is called at exactly the same time as
the first
one. I am away from my testing environment right now, so I'm
sorry for
not posting my logs. It appears to me that leg_delay_start
is broken
on at least rev 13013.
2. I'd like to stop processing the dialplan after a bridge,
but not on
specific hangup causes. If I get a MEDIA_TIMEOUT hangup
cause in the
call I'd like to continue in the dialplan. Currently I have
the
following:
action application=set
data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=true/
action application=bridge
data=user/mikael-no...@voip.domain.com/
!-- I will only get here if the first bridge is
rejected or
TODO: I get a MEDIA_TIMEOUT on it --
action application=bridge
data=openzap/1/a/99355151/
Any ideas on how to accomplish this?
I started testing this with the following dialplan:
extension name=mikael-nokia+fallback
condition field=destination_number expression=^503$
action application=set
data=hangup_after_bridge=false/
action application=set
data=continue_on_fail=true/
action application=bridge
data=user/mikael-no...@fs.voip.domain.com/
action application=info/
action application=set
data=followme_extension=99355151/
action application=execute_extension
data=post_call_followme_check/
action application=hangup/
/condition
/extension
extension name=post_call_followme_check
condition field=destination_number
expression=^post_call_followme_check$/
condition field=${originate_disposition}
expression=^MEDIA_TIMEOUT|$${continue_on_fail_causes}$
break=on-true
action application=log data=1 Follow me transferring
call
because of orig disposition: ${originate_disposition}/
action application=transfer
data=${followme_extension}/
/condition
condition
action application=log data=1 Follow me call ended
normally
with orig disposition: ${originate_disposition}./
action application=hangup/
/condition
/extension
${originate_disposition} never has the value of MEDIA_TIMEOUT
since the
call was answered, which is absolutely correct, so what I am
Re: [Freeswitch-users] [Remote SIP client] Couple of questions
mercutioviz wrote: Just following up... did you get these questions ironed out? Not yet, but I do need to have a clear understanding about how to set things up when NAT is involved, especially when remote SIP users are also behind a NAT router, and especially if they can't make any change to it (eg. staying in a hotel, or don't have the skills required to open up ports). Thank you. -- View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p23079426.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] FS/Sofia not sending bye when FreeSWITCH ends the call?
I've added this as jira case http://jira.freeswitch.org/browse/MODSOFIA-4 I wasn't sure if it should be under mod_sofia or sofia-sip. The report has a full debug log attached. Regards, Peter Olsson Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Anthony Minessale Skickat: den 16 april 2009 14:23 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? yes open a jira http://jira.freeswitch.org *attach* the following (do not paste it inline into the comments and give all trace files a .txt extension) repeat the trace you did earlier with more debugging enabled. type these 3 cli commands before you call sofia profile internal siptrace on sofia loglevel all 9 console loglevel debug On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson peter.ols...@visionutveckling.semailto:peter.ols...@visionutveckling.se wrote: Allright, I tried this again now, with revision 13042 - it's the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I'll try to help out as much as possible. Peter Från: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 15 april 2009 23:21 Till: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto: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:anthony_miness...@hotmail.commailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.commailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.nethttp://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgmailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orgmailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 !DSPAM:49e725b432939831339029! ___ 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] Skypiax as a windows service
EG: in the farm out scenario there will be FS talking via TCP to a farm client (on local machine or remote). The farm client talks with Skype client instances running on the same machine the farm client is running on. On Thu, Apr 16, 2009 at 1:47 PM, UV u...@yuvalhertzog.com wrote: Decoupling the Skyiax from FS will solve the problem as I assume it'll use TCP/IP (winsock) to interface with FS - therefore, I can run it still on the same machine but two separate sessions. yes, it uses TCP for this. So you would end up with FS (with Skypiax module) running on RDP while the Skype client instances are running as services, on the same machine (or in different machines). FS will talk to Skype client instances via TCP. Is this acceptable to you? Other question: why not running FS as a service too? If you run FS as a service and Skype clients as services, all things would works? Why you want to use RDP for? (sorry for the silly questions, I just want to understand better). However, I think getting the Skypiax to work as a service will be more beneficial regardless if it's decoupled or not. What do you mean? I believe that Skypiax (as an FS module) works when FS is run as service. Your problem seems to me that you cannot run Skype instances under RDP because they cannot access the sound device. Is this correct? gm ___ 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] FreeSwitch Complex IVR System
Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever platform you choose. The documentation starts here: http://wiki.freeswitch.org/wiki/Mod_event_socket Cheers -- Dave Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this? ___ 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] Re-2: FreeSwitch Complex IVR System
Hi Dave, thanks for the answer. I am playing around with FS and Event Socket Library for .NET. I get pretty much to run with this, but the reason why I came from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I get an Event as soon one dtmf digit is recognized. Unfortunately this isn't the case. If I use the default config files and map the keys with bind_meta_app the dtmf tones are recognized and the function behind the bound app is executed. Is this maybe a bug. I have read about mod_managed and that I should use it, but I haven't found anything about the usage of it. Any suggestions would help thanks...Guido Original Message Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 17:35) From:David Knell d...@3c.co.uk To: g...@exram.de Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever platform you choose. The documentation starts here: http://wiki.freeswitch.org/wiki/Mod_event_socket Cheers -- Dave Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this? ___ 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 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] Re-2: FreeSwitch Complex IVR System
Hi Guido, The event socket interface will give you DTMF events for bridged calls - just tried it and it works fine. There's one mild snag, which is that outbound sockets (which are easier for inbound call handling) will only give you events relating to the specific call leg that's attached to that socket - i.e. you can use an outbound socket app to bridge that leg to an outbound call leg just fine, but you won't get events related to that outbound call. So what we do is use an outbound socket app for call control and scripting, and have a separate inbound socket app which listens for call state changes and DTMF on all call legs, and a database table which glues the two together. Cheers -- Dave Hi Dave, thanks for the answer. I am playing around with FS and Event Socket Library for .NET. I get pretty much to run with this, but the reason why I came from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought that I get an Event as soon one dtmf digit is recognized. Unfortunately this isn't the case. If I use the default config files and map the keys with bind_meta_app the dtmf tones are recognized and the function behind the bound app is executed. Is this maybe a bug. I have read about mod_managed and that I should use it, but I haven't found anything about the usage of it. Any suggestions would help thanks...Guido Original Message Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 17:35) From:David Knell d...@3c.co.uk To: g...@exram.de Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever platform you choose. The documentation starts here: http://wiki.freeswitch.org/wiki/Mod_event_socket Cheers -- Dave Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this? ___ 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 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] ekiga and freeswitch
i've posted the freeswitch svn trunk console debug and sip trace to the pastebin. On Sun, Apr 12, 2009 at 1:34 PM, Brian West br...@freeswitch.org wrote: Collect a full sip trace and FULL console debug. Put it on our pastebin... Chances are Ekiga is doing something stupid... it usually does silly things. Also are you on SVN trunk? /b On Apr 12, 2009, at 12:11 PM, e schmidbauer wrote: Not sure if this is a bug in the program or just in my setup. I've tried using the svn version of freeswitch (as of yesterday) and i got the exact same error. Any input would be appreciated. Thanks! ___ 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] RTP errors
Hi Guys, I'm getting a few of these errors below sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error! Are these caused by a fax machine? Or am I barking up the wrong tree? Regards, ___ 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] RTP errors
Well a little more detail would be great :P /b On Apr 16, 2009, at 2:46 PM, Nik Middleton wrote: Hi Guys, I’m getting a few of these errors below sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error! Are these caused by a fax machine? Or am I barking up the wrong tree? Regards, ___ 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 Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Issues detecting DTMF tones
Hey guys. Has anyone else experienced the inability to detect/receive DTMF tones? Just yesterday I had about 4-5 hours where One of my IVR scripts would not detect 1, 2 or 3, but detected the other digits perfectly. If I removed the sound file that was playing, and substituted silence it worked, add the sound file in, and it broke. I have a strong feeling that this is not an issue with FS, but with an upstream system. But wanted to know if anyone has seen this before, and how they went about identifying the culprit and/or fixing it. Some background: - Using FS trunk - Both legs of the call were via SIP gateway. - Setting loglevel to 9 (console and sofia) showed that the RTP packets were not received by FS for 1/2/3 but were received for other digits - Both legs of calls were to/from ATT cell phones - Was using session:setInputCallback() to receive tones, did not test with playAndGetDigits() Thanks for any help. -pete ___ 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] Optimum sound file format
Hi Guys, I'm looking for the optimum audio format when using streamfile in a lua script. I've found CPU load increases rapidly with the number of threads playing a .wav file. Can anyone tell me the optimum when using g711a? Right now the the .wav files are Audio format: PCM Sample rate : 8 kHz Mono Sample Size: 16 bit Bit rate :128kbps Will it help CPU load if I resample to a bit rate of 64kbps and sample size of 8 bit? I have read that the sample size needs to be 13-14bit +1 for alaw/ulaw though Regards, ___ 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] Optimum sound file format
Looking at your post, You are already using the best format. If you do not have a fast filesystem try making a ram disk and play the files from there instead. if you *really* want you can use sox to turn them all into raw alaw files and rename them with a .PCMA extension to avoid the g711 transconding but g711 to PCM is pretty trivial. it's more likely a file i/o distress you see. On Thu, Apr 16, 2009 at 5:04 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, I’m looking for the optimum audio format when using streamfile in a lua script. I’ve found CPU load increases rapidly with the number of threads playing a .wav file. Can anyone tell me the optimum when using g711a? Right now the the .wav files are Audio format: PCM Sample rate : 8 kHz Mono Sample Size: 16 bit Bit rate :128kbps Will it help CPU load if I resample to a bit rate of 64kbps and sample size of 8 bit? I have read that the sample size needs to be 13-14bit +1 for alaw/ulaw though Regards, ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] Skypiax as a windows service
are you planning on just signaling on TCP or both audio and signalling cos realtime audio over TCP kinda stinks. you may find that just running FS as the farm and calling to it with sip is more or less the same idea with no work ;) On Thu, Apr 16, 2009 at 10:09 AM, Giovanni Maruzzelli gmar...@celliax.orgwrote: EG: in the farm out scenario there will be FS talking via TCP to a farm client (on local machine or remote). The farm client talks with Skype client instances running on the same machine the farm client is running on. On Thu, Apr 16, 2009 at 1:47 PM, UV u...@yuvalhertzog.com wrote: Decoupling the Skyiax from FS will solve the problem as I assume it'll use TCP/IP (winsock) to interface with FS - therefore, I can run it still on the same machine but two separate sessions. yes, it uses TCP for this. So you would end up with FS (with Skypiax module) running on RDP while the Skype client instances are running as services, on the same machine (or in different machines). FS will talk to Skype client instances via TCP. Is this acceptable to you? Other question: why not running FS as a service too? If you run FS as a service and Skype clients as services, all things would works? Why you want to use RDP for? (sorry for the silly questions, I just want to understand better). However, I think getting the Skypiax to work as a service will be more beneficial regardless if it's decoupled or not. What do you mean? I believe that Skypiax (as an FS module) works when FS is run as service. Your problem seems to me that you cannot run Skype instances under RDP because they cannot access the sound device. Is this correct? gm ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] Skypiax as a windows service
Ok, I think I know where's the confusion here. Let me clarify: 1. FS run beautifully as a service - that's why I assumed it should work. 2. Skype client runs as a service very well too. 3. When running FS as a service with Skypiax (hence Skypiax as a service), Skypiax doesn't seem to find the SkypeAPI. In the Wiki page http://wiki.freeswitch.org/wiki/Skypiax#Running_Skypiax_on_Windows_as_a_Serv ice it's says that Running Skypiax on Windows as a Service is Not yet written therefore I assumed it's a known limitation. Are you saying it isn't? Anyway, the farming solution you suggested should solve the problem - I'd assume. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Friday, April 17, 2009 1:10 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Skypiax as a windows service EG: in the farm out scenario there will be FS talking via TCP to a farm client (on local machine or remote). The farm client talks with Skype client instances running on the same machine the farm client is running on. On Thu, Apr 16, 2009 at 1:47 PM, UV u...@yuvalhertzog.com wrote: Decoupling the Skyiax from FS will solve the problem as I assume it'll use TCP/IP (winsock) to interface with FS - therefore, I can run it still on the same machine but two separate sessions. yes, it uses TCP for this. So you would end up with FS (with Skypiax module) running on RDP while the Skype client instances are running as services, on the same machine (or in different machines). FS will talk to Skype client instances via TCP. Is this acceptable to you? Other question: why not running FS as a service too? If you run FS as a service and Skype clients as services, all things would works? Why you want to use RDP for? (sorry for the silly questions, I just want to understand better). However, I think getting the Skypiax to work as a service will be more beneficial regardless if it's decoupled or not. What do you mean? I believe that Skypiax (as an FS module) works when FS is run as service. Your problem seems to me that you cannot run Skype instances under RDP because they cannot access the sound device. Is this correct? gm ___ 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] Optimum sound file format
Thanks for this. One of the servers is using sata and the other scsii drives, so that may be the problem, I'll give it a go. Problem seems to escalate past 200 active calls. Below that all is well. That said, it could also be a db issue, so I've changed my log tables to innodb (I'm hoping that now I have row level locking as opposed to table level it will help) Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 16 April 2009 23:25 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Optimum sound file format Looking at your post, You are already using the best format. If you do not have a fast filesystem try making a ram disk and play the files from there instead. if you *really* want you can use sox to turn them all into raw alaw files and rename them with a .PCMA extension to avoid the g711 transconding but g711 to PCM is pretty trivial. it's more likely a file i/o distress you see. On Thu, Apr 16, 2009 at 5:04 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, I'm looking for the optimum audio format when using streamfile in a lua script. I've found CPU load increases rapidly with the number of threads playing a .wav file. Can anyone tell me the optimum when using g711a? Right now the the .wav files are Audio format: PCM Sample rate : 8 kHz Mono Sample Size: 16 bit Bit rate :128kbps Will it help CPU load if I resample to a bit rate of 64kbps and sample size of 8 bit? I have read that the sample size needs to be 13-14bit +1 for alaw/ulaw though Regards, ___ 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:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org 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] [Remote SIP client] Couple of questions
definitely stop by IRC and talk real-time with others who've dealt with this kind of thing. -MC On Thu, Apr 16, 2009 at 7:44 AM, Fred-145 codecompl...@free.fr wrote: mercutioviz wrote: Just following up... did you get these questions ironed out? Not yet, but I do need to have a clear understanding about how to set things up when NAT is involved, especially when remote SIP users are also behind a NAT router, and especially if they can't make any change to it (eg. staying in a hotel, or don't have the skills required to open up ports). Thank you. -- View this message in context: http://www.nabble.com/-Remote-SIP-client--Couple-of-questions-tp22698296p23079426.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] FreeSwitch Complex IVR System
To add to David's comments: Have a look at ESL, which is the event socket library that the fs devs created. It's an abstraction layer that makes it easier to use the event socket with the programming language of your choice. In fact, the program fs_cli.c is a great example of a program that uses the ESL to talk to a FreeSWITCH server. (fs_cli = FreeSWITCH Command Line Interface program, kinda like asterisk -r if that means anything to you...) Your project is most definitely possible with FreeSWITCH. A number of people in the FS community have done bits and pieces of what you've described. Your big challenge is that you're going to need a person or a team who understands multiple technology concepts and how to integrate them: telephony signaling, socket communications, database management, scripting and programming, etc. This is a bold project and I would love to see you use FS to be the telephony engine. Let us know what you decide. Also, if you need professional FS assistance you can request it at consult...@freeswitch.org. -MC On Thu, Apr 16, 2009 at 8:35 AM, David Knell d...@3c.co.uk wrote: Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever platform you choose. The documentation starts here: http://wiki.freeswitch.org/wiki/Mod_event_socket Cheers -- Dave Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this? ___ 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 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 bridge in free switch
Do you need to monitor the possible failure of one of these calls? Just curious. You can call them individually and drop them into a conference right at the FS cmd line: originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); You can control the conference behavior with numerous options. See http://wiki.freeswitch.org/wiki/Mod_conference for lots of great information. -MC On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan mprabhu...@gmail.comwrote: Hello, I am trying to find a way to this through fs_cli 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108) ClientC (1...@192.168.1.108) 2) Bridge all the 3 legs together into one call Thanks Prabhu ___ 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 bridge in free switch
Hi Mike, I tried the following command per ur advice.. but getting the error CHAN_NOT_IMPLEMENTED originate (sofia/profile/1...@192.168.1.108) conference(3085-192.168.1.102); freeswi...@internal originate (sofia/profile/1...@192.168.1.102) conference(3085-192.168.1.102); -ERR CHAN_NOT_IMPLEMENTED freeswi...@internal 2009-04-16 20:28:30 [ERR] switch_core_session.c:303 switch_core_session_outgoing_channel() Could not locate channel type (sofia 2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [(sofia] cause: [CHAN_NOT_IMPLEMENTED] 2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] Thanks prabhu On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins m...@freeswitch.org wrote: Do you need to monitor the possible failure of one of these calls? Just curious. You can call them individually and drop them into a conference right at the FS cmd line: originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); You can control the conference behavior with numerous options. See http://wiki.freeswitch.org/wiki/Mod_conference for lots of great information. -MC On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan mprabhu...@gmail.comwrote: Hello, I am trying to find a way to this through fs_cli 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108) ClientC (1...@192.168.1.108) 2) Bridge all the 3 legs together into one call Thanks Prabhu ___ 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 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 bridge in free switch
First off remove the () around the sofia URI. /b On Apr 16, 2009, at 10:33 PM, Prabhuram Mohan wrote: Hi Mike, I tried the following command per ur advice.. but getting the error CHAN_NOT_IMPLEMENTED originate (sofia/profile/1...@192.168.1.108) conference(3085-192.168.1.102); Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Call bridge in free switch
Take out the brackets - originate sofia/profile/1001... (and you might want to replace profile with the name of the profile to use) There's documentation here which might help: http://wiki.freeswitch.org/wiki/Mod_commands#originate --Dave Hi Mike, I tried the following command per ur advice.. but getting the error CHAN_NOT_IMPLEMENTED originate (sofia/profile/1...@192.168.1.108) conference(3085-192.168.1.102); freeswi...@internal originate (sofia/profile/1...@192.168.1.102) conference(3085-192.168.1.102); -ERR CHAN_NOT_IMPLEMENTED freeswi...@internal 2009-04-16 20:28:30 [ERR] switch_core_session.c:303 switch_core_session_outgoing_channel() Could not locate channel type (sofia 2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [(sofia] cause: [CHAN_NOT_IMPLEMENTED] 2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] Thanks prabhu On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins m...@freeswitch.org wrote: Do you need to monitor the possible failure of one of these calls? Just curious. You can call them individually and drop them into a conference right at the FS cmd line: originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); You can control the conference behavior with numerous options. See http://wiki.freeswitch.org/wiki/Mod_conference for lots of great information. -MC On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan mprabhu...@gmail.com wrote: Hello, I am trying to find a way to this through fs_cli 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108) ClientC (1...@192.168.1.108) 2) Bridge all the 3 legs together into one call Thanks Prabhu ___ 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 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 bridge in free switch
Thanks Brain/ Dave, I ran the modified command as follows originate sofia/default/1...@192.168.1.102 conference(3085-192.168.1.102) this time fs is able to create channel but am getting a different error : sofia.c:3845 Cannot Blind Tranfer one legged call Prabhu On Thu, Apr 16, 2009 at 8:58 PM, David Knell d...@3c.co.uk wrote: Take out the brackets - originate sofia/profile/1001... (and you might want to replace profile with the name of the profile to use) There's documentation here which might help: http://wiki.freeswitch.org/wiki/Mod_commands#originate --Dave Hi Mike, I tried the following command per ur advice.. but getting the error CHAN_NOT_IMPLEMENTED originate (sofia/profile/1...@192.168.1.108) conference(3085-192.168.1.102); freeswi...@internal originate (sofia/profile/1...@192.168.1.102) conference(3085-192.168.1.102); -ERR CHAN_NOT_IMPLEMENTED freeswi...@internal 2009-04-16 20:28:30 [ERR] switch_core_session.c:303 switch_core_session_outgoing_channel() Could not locate channel type (sofia 2009-04-16 20:28:30 [ERR] switch_ivr_originate.c:1486 switch_ivr_originate() Cannot create outgoing channel of type [(sofia] cause: [CHAN_NOT_IMPLEMENTED] 2009-04-16 20:28:30 [DEBUG] switch_ivr_originate.c:2084 switch_ivr_originate() Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] Thanks prabhu On Thu, Apr 16, 2009 at 4:29 PM, Michael Collins m...@freeswitch.org wrote: Do you need to monitor the possible failure of one of these calls? Just curious. You can call them individually and drop them into a conference right at the FS cmd line: originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); originate (sofia/profile/1...@192.168.1.108) conference(myconfname); You can control the conference behavior with numerous options. See http://wiki.freeswitch.org/wiki/Mod_conference for lots of great information. -MC On Thu, Apr 16, 2009 at 1:09 AM, Prabhuram Mohan mprabhu...@gmail.com wrote: Hello, I am trying to find a way to this through fs_cli 1) call out to ClientA (1...@192.168.1.108), ClientB (1...@192.168.1.108) ClientC (1...@192.168.1.108) 2) Bridge all the 3 legs together into one call Thanks Prabhu ___ 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 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 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
