Re: [Freeswitch-users] SIP Overlap support?
is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. 2009/11/3 Anthony Minessale : > The patch was it's ability to accept subsequent invites. > Your problem is that in sip each new attempt to send an invite is another > call. > > 484 is a final response so the call with too few digits is terminated. ___ 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] SIP Overlap support?
hi anthony, i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket. you wrote about a "1 line patch", which might not have been implemented - at least it seems so. is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest. kind regards, dennis 2009/10/15 Anthony Minessale : > right you can reply 484 in your dp at any time > > > then it should try again. > > The bit i can't remember is if we committed a certain 1 line patch that > makes sofia parse the next invite to the same call properly, the patch was > to the sofia lib itself so test it and see. I may need to dig up the answer > again from the sofia dev. ___ 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] SIP Overlap support?
ok, as written, i come back after some tests with fs and a thomson cirpack. it did not work - at least in our tests. we are using socket outbound and when a call comes in, it starts the socket of fs. the number may be 123456. fs sends the respond 484 and our carrier receives this information. but fs ends the call with hangup_cause = invalid_number_format. the carrier has one more digit for the phone number and sends 1234567 and the above mentioned behavior repeats. the behavior we want and expected is, that the call stays in the socket after response 484, so that the carrier can send the 1234567 into the same socket. the management, when fs should send response 484 and when fs should be answered would be programmed by us. it also important, that fs keeps the call in the socket, so we can tell fs, to answer the call after x seconds anyway. any ideas, what we could do? ___ 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] SIP Overlap support?
ok, we will try this with the cirpack of our carrier. this will take some days, till everything is set up. after the tests i will come back to report. 2009/10/15 Anthony Minessale : > right you can reply 484 in your dp at any time > > > then it should try again. > > The bit i can't remember is if we committed a certain 1 line patch that > makes sofia parse the next invite to the same call properly, the patch was > to the sofia lib itself so test it and see. I may need to dig up the answer > again from the sofia dev. ___ 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] SIP Overlap support?
> once you have 123456 won't you still be unsure if he will type the next 1 or > not and be forced to refuse it and wait anyway? basically you are right. BUT, we know, that a basic phone number has 6 digits - so, we do not have to check anything before. as soon as we have 6 digits, we look in our database for this number. a flag in the database tells us, if this number is allowed to use ddi (further digits) or not. if the number is not allowed, we can answer the call directly without any further waiting. other numbers are maked to have 2 ddi's - here we know, that we have to wait for 2 more digits, before we answer. if a number can have 6 digits or 8 digits, you are right, we have to wait anyway. but without having overlap, there is ALWAYS a waiting time. we would like to make the answer-times as short as possible. if no waiting is needed, we do not want to wait. we would like to tell fs to send a 484 response incomplete to the cirpack of our carrier if we need/want more digits. perhaps this works with fs, but not with "socket outbound", which we are using? we know, that this is more work for our servers, but they are powerful enough and have enough resources, that they should be able to handle that. of course we do not want to become to experimental, because fs works so extremly smooth and reliable, that it is nearly unbelievable. ___ 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] SIP Overlap support?
the thing we want to make working nicer is the following: we want the main/basic phonenumber (123456) to be reachable, so that the telephone rings. but we also want it to be expandable with ddi-digits. example: dial the 123456 to reach the company, dial the 123456 1 to reach the support. in the moment our carrier waits x (milli-)seconds to see, if the 123456 was dialed or if there were more digits attached. after this time, the carrier sends us the whole number as a block (may it be 123456 or 1234561) and we answer. we want to avoid the waiting time the carrier needs to wait for (possible) more digits. we want to receive every single digit from our carrier and we want to tell fs, when the number is complete and fs can answer. ___ 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] SIP Overlap support?
how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not. any hint? 2009/10/13 Anthony Minessale : > have you tried it? > I *think* either we did support it or we would with a small patch to sofia > lib that I cannot recall if we ever got committed. > > > On Tue, Oct 13, 2009 at 8:51 AM, Dennis wrote: >> >> hi there, >> >> i would like to ask, if fs has support for something like "SIP Overlap"? >> >> instead of receiving the phonenumber from our carrier in a block, we >> want to receive the phonenumber digit-by-digit and we want to tell fs >> when the number is complete. our carrier could send us the phonenumber >> digit-by-digit, but what about the fs-side? >> >> >> thanks and kind regards >> dennis >> >> ___ >> 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@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 > > ___ 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] SIP Overlap support?
hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side? thanks and kind regards dennis ___ 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 exits after 120 seconds with hangup cause
> If you look at your trace the call is sending a re-invite over and over and > over again with no reply > > you need to examine your network topology and find out why the packets FS is > sending to your phone > never make it. > > also try disabling session-timers on the snom are you talking about our problem, that we get a hangup after 120 seconds? i just ask to avoid, that we are talking about different things. if we connect without stun and call the 5900, we can hear the music, so i assume, that we are receiving packets from fs!? but after about 120 seconds, we receive the hangup. before everything was fine and as expected. we also tried disabling session-timers in the snom - same problem. thant's the problem, we tried soo many things, but simply nothing changes the problem or makes something different (like sending the hangup later or earlier). ___ 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 exits after 120 seconds with hangup cause
> I think the problem is the session-timeout is too long and your nat mapping > is being deleted. we changed the param session-timeout (in internal) to 20 and 40, without success. we changed the minimum-session-expires to 20, although we knew, the allowed minimum is 90 and 90 was shown in the console - no success. > try setting it to a smaller value like 20 or 40 sec > or try setting those phones to register more frequently (every 10 or 20 sec) we changed all possible options to 10 and 20, again without success :( ___ 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 exits after 120 seconds with hangup cause
the pastebin number is 10129 ___ 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 exits after 120 seconds with hangup cause
this is the line (without stun - so we only have one leg) and we called the 5900 to moh: 2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup sofia/internal/1...@212.18.215.102 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] we called the 5900 and waited 2 minutes... or did you mean something different? ___ 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] Question about presence
> are you setting presence_id=u...@domain variable on the outbound leg? > This is done for you in the DP via the user/ channel in the defaults but if > you are not using this > you have to set it manually. in directory default we have the following: everthing works fine with the led lights - the only problem is the described above. ___ 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 exits after 120 seconds with hangup cause
> Are the phones behind the same nat as FS? no, both phones are behind the same nat, fs is behind another nat (and the internet is inbetween). both phones are in our office and we are sitting behind a nat. we connect through our nat over the internet into the other nat, where fs is behind. ___ 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 exits after 120 seconds with hangup cause
> Is the snom firmware up to the latest? yes, the firmware is the latest. > I believe session timers should work properly with snom? i don't know, but i think so. we tested with session timers "off" and with setting the session timer to 0. both does not change anything. we played with using a stun-server and had a small success. if we are connected with a stund-server, the hangup will not come after 120 seconds. we can also call the 5900 and listen to the moh. this works fine. here we have a totally different problem. if another voip phone is connected to the server using stun (for example the 1000), both sides can not talk to each other. one won't hear, if the other side is talking. very strange (for us), but we have no idea, what we can do about that. ___ 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] Question about presence
sorry, but i do not know i which category i have to set this problem. could you help me with that? ___ 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 exits after 120 seconds with hangup cause
> if you are talking to a device that pretends to do session timers but really > does not, the re-invite can cause a hangup. > We have seen this with sonus many times, do you have sonus involved in your > call per chance? hi anthony, this is what we found on the snom website about the session timer settings: "If SIP Session Timer Support is enabled, this option specifies the SIP session timer in seconds. For instance, a Re-INVITE will be sent after 50% of its value has elapsed." our setting is 3600. is session timer and re-invite helpful or can we disable it? behaps disabling works? is it possible, that snom does not support a REAL session timer? sonus is not involved. kind regards dennis ___ 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] Question about presence
> ahh, I understand the issue now. Please open a jira on jira.freeswitch.org > for this issue. ok, we could not imagine, that this behavior is meant to be. we will try to open a jira with this issue (never opened a jira before). kind regards dennis ___ 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] Question about presence
> Do about what? Your description sounds fine, what is the problem 2 people are talking and only ONE led is on? for me it sounds wrong. the led of p1 AND p2 should be on! the led should indicate, who in the company/group is busy/talking and who is available - i think that's the joke about the leds. kind regards dennis ___ 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] Question about presence
hi, we have set manage-presence = true to see, who is talking on the phone and who is free. everyone here has his own snom-voip phone with 12 led-lights. incoming calls are handled in groups. this means, if someone is ringing, all voip-phones of one group are ringing and one can see all led-lights flashing. if someone answers the phonecall, the led of this person (p1) is on on everybodys phone, so everybody can see, that the person is talking. now, if another phonecall comes in, all led-lights are flashing - including the led of the person who is talking. if another person (p2) answers the second phonecall, the led of p2 is on, but the led of p1 is off, although p1 is still talking. is there something we can do about this? thanks & kind regards dennis ___ 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 exits after 120 seconds with hangup cause
> You have a NAT issue. as i wrote, we are quite sure, that this is a nat problem. but we have no idea how we could fix this. there must be a reason, why the hangup always comes after 120 seconds. > > > we tested that, but this does not change anything (nothing becomes worse or better). > param name="session-timeout" value="120" this line is commented in our profile and we believe, that this has nothing to do with our problem. to tell a little bit more about our problem: under http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2 you can find the scenario, which is exactly like ours. we played with all the mentioned settings, but it does not work. to mention one thing: if we connect with xlite (softphone), we do not have any problems. but we have snom-voip-phones and with them we have the problems. the big question is: wo or what might cause/trigger the hangup? is it freeswitch or something else? we have a firewall (IPCop) - might there be a setting, which needs to be set, to avoid theses problems? kind regards dennis ___ 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 exits after 120 seconds with hangup cause
hi, we are using fs for different services, but we never used it, to connect sip-phones directly with fs. now we want to do so, but we encounter big problems. everything works fine, but after 120 seconds fs hangs up with the hangup cause RECOVERY_ON_TIMER_EXPIRE. it seems that this has something to do with nat problems (i read about it in nabble), but we can not figure out, how to fix the problem. we are not using stun, because all ip-addresses are static. we opened our firewall for our ip-addresses and opened the following ports: 5060 - 5091 TCP / UDP 1 - 32767 UDP we are using a quite actual version of fs - the revision ist 13783. we set up a profile in the internal.xml and played a lot with different settings. we can talk to each other, make outbound calls and receive calls - but we can not figure out, how to avoid the hangup after 120 seconds. thanks a lot for your help dennis ___ 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] mux with uuid_displace: soundfile to loud
thanks for your answers. i did not know, that muxing is cpu intensive. i just thought, that it would not matter, if one is muxing 50/50 or 30/70. for playing back a soundfile, while one can hear the other end it seems, that muxing is required. so the level of muxing should not make a difference!? anyway, if there is no other/better way, we have to do it with sox. no, we are not using stereo-files. kind regards dennis ___ 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] mux with uuid_displace: soundfile to loud
hi, we want to use uuid_displace with mux to playback a soundfile to a bridged uuid, so that this uuid can hear the other side talk AND hear the soundfile (whispering). is there an option we can set, for defining the loudness level of the soundfile? in our tests the soundfile was way to loud, so that it was nearly impossible to hear the other side talk, while the soundfile was playing. we tried "uuid_displace uuid start /path/to/soundfile/soundfile.wav 0 mux 0.3", so that the loudness of soundfile only would be 30% - but this does not work. thanks & kind regards dennis ___ 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 to work with the debug logfile?
hi, we encounter some small problems withing the past 2 days and we are trying to find out more about the problems. for this we downloaded the debug logfiles written by fs, but we do not manage to filter all log-entries for one single special call. we are using socket outbound and would like to see all entries (inbound/outbound) of one call. is this possible? kind regards dennis ___ 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] Fax through FS to Callweaver. How?
hi, sorry for the late reply and thanks for the replies. it working quite fine now. we still habe some problems, because we have 2 nic's (internal ip/external ip). we have to find a way through different ip-adresses, ip-areas, firewall and switch :-) or is there a way to make fs listen to two ip-adresses? kind regards dennis 2009/5/13 Antonio Gallo : > Dennis ha scritto: >> does someone know callweaver and can tell me, if there are some >> important settings to be set for making it work with fs in the middle? >> > Look at this, i needed to apply it using a Patton gateway too: > http://www.callweaver.org/ticket/487 > > > ___ > 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] Fax through FS to Callweaver. How?
hi, we are doing a lot with freeswitch and everything works absolutely perfect. the only thing we are missing, is a good support for fax over ip - may it be for offering a fax-2-mail service or just passing incoming fax-calls from our fs-servers to a "real" fax-machine. now we have another server only for callweaver, which has a great fax-support. if we send faxes from our fax-machine over our sip-provider (who supports t.38) directly to the cw-server, all faxes are beeing received and saved by cw. but if we send faxes over the same sip-provider first to fs and then do a bridge to the cw-server, all faxes fail. we set in internal.xml and play with bypass_media=true and/or proxy_media=true in the dialplan. it simply does not work. i think and hope that there are some other settings i can change, so that fs is nothing more than just a "tunnel" and let both sides communicate with eachother? does someone know callweaver and can tell me, if there are some important settings to be set for making it work with fs in the middle? thanks & kind regards dennis ___ 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] Socket-Event on "originate call"?
the problem is fixed in the latest version of fs - at least it is working as before without any errors. but there is still the question, if the changes where made because of our problem with the not starting socket!? we can see in the cli, that the var is set, but it does not change anything regarding our problem. 2009/2/11 Brian West : > Please collect the backtrace and report it on Jira. > > /b > > On Feb 11, 2009, at 2:11 PM, Dennis wrote: > >> this does not help. we are using socket outbound and everything worked >> before the changes yesterday. >> >> we have the same error with other dialplans. ___ 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] Socket-Event on "originate call"?
this does not help. we are using socket outbound and everything worked
before the changes yesterday.
we have the same error with other dialplans.
> 2009/2/11 Brian West :
> Try answer or pre_answer before park.
>
> /b
>
> On Feb 11, 2009, at 12:37 PM, Dennis wrote:
>
>> anthony, did you make the changes with "add {instant_ringback=true} to
>> make ringback not wait for indication to generate ringback" for the
>> described problem?
>>
>> we read something like this out of it, but we can not test it, because
>> we get errors with the latest fs version (switch_ivr.c:674
>> switch_ivr_park() Cannot park channels that have no read codec.).
___
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] Socket-Event on "originate call"?
anthony, did you make the changes with "add {instant_ringback=true} to
make ringback not wait for indication to generate ringback" for the
described problem?
we read something like this out of it, but we can not test it, because
we get errors with the latest fs version (switch_ivr.c:674
switch_ivr_park() Cannot park channels that have no read codec.).
2009/2/10 Dennis :
> yes, you are right. we are receiving the reply.
>
> but, we are using socket outbound and manage all calls over this
> socket. we also measure the durations (like variable_duration and
> variable_billsec) and count all outgoing calls over the socket.
> but, if the originate (without an inbound call) will not start the
> socket, we can not count up, how many calls failed because of "user
> busy" or how long the platform was in use.
>
> a possible workarround: is it possible to trigger a dialplan over the
> cli (like our default dialplan, which starts the socket), so that the
> dialplan starts the originates?
>
> the basic problem for us, that, if we just want to make dialouts, we
> are missing the inbound call to start the socket.
>
>
> kind regards
> dennis
>
>
>
> 2009/2/9 Anthony Minessale :
>> when an originate is unsuccessful the failure and the cause code is returned
>> as the reply to the originate request.
>>
>>
>> On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote:
>>>
>>> hi,
>>>
>>> i am using socket outbound with fs.
>>>
>>> if i do an originate over the console, for starting an outbound call
>>> without having an inbound call, and send the originate directly to the
>>> socket, the socket is first started, if the call is in answer or
>>> ringing state.
>>> before this, i will not receive any event, because the socket was not
>>> started. therefore i will not know, if the target is "busy" (hangup,
>>> hangup cause: user busy).
>>>
>>> it would be very helpful, if the socket would start immediately on an
>>> event like "channel originate".
>>>
>>>
>>> thanks for the help
>>> dennis
>
___
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] DTMF: Mute sound for the other side?
ok, i will try this, but how can it be possible, that inband tones are audible in conference, when we do not even have start_dtmf activated? i just don't understand, why it must be dtmf inband, if the tones are audible and how they can be audible, if start_dtmf is not set. is it, because the carrier just sends them as normal sound, which is played as a tone, without beeing used for dtmf? 2009/2/11 Brian West : > turn on the start_dtmf app and dial digits from the outside.. if you > get duplicate digits then they are sending both. > > /b > > On Feb 11, 2009, at 11:14 AM, Dennis wrote: > >> i can't tell, if they are sending both, but it seems so. we get 2833 >> for sure. they were kind enough to give it to us, because inband seems >> to be quite unreliable over sip. >> >> how can in find out, if both are coming and is there a way to "block" >> inband to test? >> >> perhaps we need both: if we bridge an inbound with another ivr on the >> outbound side, which is not sip and does not understand 2833, we need >> to pass inband through or something like this. or am i wrong with >> 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
Re: [Freeswitch-users] DTMF: Mute sound for the other side?
i can't tell, if they are sending both, but it seems so. we get 2833 for sure. they were kind enough to give it to us, because inband seems to be quite unreliable over sip. how can in find out, if both are coming and is there a way to "block" inband to test? perhaps we need both: if we bridge an inbound with another ivr on the outbound side, which is not sip and does not understand 2833, we need to pass inband through or something like this. or am i wrong with this? 2009/2/11 Brian West : > Well if they are sending both they are broken. I would call and yell > at them and beat them with a cluebat. > > /b > > On Feb 11, 2009, at 10:42 AM, Dennis wrote: > >> that is interesting. we are receiving the dtmf digits over 2833. might >> it be possible, that we receive 2833 AND inband (we asked our carrier >> for 2833, because we had problems with inband and fs - and we got it)? >> >> is there something we can setup in fs or is it a problem wich only our >> carrier can solve? > > > ___ > 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] DTMF: Mute sound for the other side?
that is interesting. we are receiving the dtmf digits over 2833. might it be possible, that we receive 2833 AND inband (we asked our carrier for 2833, because we had problems with inband and fs - and we got it)? is there something we can setup in fs or is it a problem wich only our carrier can solve? 2009/2/10 Michael Jerris : > If your in a conference and your hearing other people hitting dtmf > digits that IS inband, it means that the place upstream that is doing > inband to 2833 conversion is not properly clipping the dtmf, this > probably needs to be fixed on that device. ___ 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] Socket-Event on "originate call"?
yes, you are right. we are receiving the reply. but, we are using socket outbound and manage all calls over this socket. we also measure the durations (like variable_duration and variable_billsec) and count all outgoing calls over the socket. but, if the originate (without an inbound call) will not start the socket, we can not count up, how many calls failed because of "user busy" or how long the platform was in use. a possible workarround: is it possible to trigger a dialplan over the cli (like our default dialplan, which starts the socket), so that the dialplan starts the originates? the basic problem for us, that, if we just want to make dialouts, we are missing the inbound call to start the socket. kind regards dennis 2009/2/9 Anthony Minessale : > when an originate is unsuccessful the failure and the cause code is returned > as the reply to the originate request. > > > On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: >> >> hi, >> >> i am using socket outbound with fs. >> >> if i do an originate over the console, for starting an outbound call >> without having an inbound call, and send the originate directly to the >> socket, the socket is first started, if the call is in answer or >> ringing state. >> before this, i will not receive any event, because the socket was not >> started. therefore i will not know, if the target is "busy" (hangup, >> hangup cause: user busy). >> >> it would be very helpful, if the socket would start immediately on an >> event like "channel originate". >> >> >> thanks for the help >> dennis ___ 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] DTMF: Mute sound for the other side?
we are not using inband tones. we are using rfc2833. is it still neccessary, to do some extra programming? if yes: isn't there a way for fs to recognize, that there is a rfc2833 and simply does not play it back for the others? 2009/2/9 Anthony Minessale : > 1) don't use inband tones for dtmf. > 2) post a bounty to have FS clip the audio for x milliseconds when a tone is > detected. (you will still hear faint clicks between the start of the tone > and when the clipping activates) > > > > On Mon, Feb 9, 2009 at 8:59 AM, Dennis wrote: >> >> hi, >> >> i am having a small problem with the dtmf-sounds... >> >> if i press a dtmf digit while i am bridged with another leg, the other >> side will hear the dtmf sound. >> this is very annoying and even worse in a conference, when multiple >> people can press dtmf digits (for (un-)muting themselves or using >> other functions). >> >> is there a way, to NOT let the other side hear the dtmf sound (but of >> course still make fs listening to it)? >> >> >> thanks for the help >> dennis >> >> ___ >> 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 > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@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 > > ___ 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] Socket-Event on "originate call"?
hi, i am using socket outbound with fs. if i do an originate over the console, for starting an outbound call without having an inbound call, and send the originate directly to the socket, the socket is first started, if the call is in answer or ringing state. before this, i will not receive any event, because the socket was not started. therefore i will not know, if the target is "busy" (hangup, hangup cause: user busy). it would be very helpful, if the socket would start immediately on an event like "channel originate". thanks for the help dennis ___ 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] DTMF: Mute sound for the other side?
hi, i am having a small problem with the dtmf-sounds... if i press a dtmf digit while i am bridged with another leg, the other side will hear the dtmf sound. this is very annoying and even worse in a conference, when multiple people can press dtmf digits (for (un-)muting themselves or using other functions). is there a way, to NOT let the other side hear the dtmf sound (but of course still make fs listening to it)? thanks for the help dennis ___ 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] Conference and socket outbound
is it possible to define a profile and its params for a conference dynamically over socket outbound? in the moment, if we want to have multiple profiles for different clients, we (have to) setup a profile in the conference.conf - otherwise we get an error in fs. because we have multipple fs-servers and multiple clients using conference, this is not very compfortable to setup. therefor it would be great, if we could call a conference profile name and set params dynamically over the socket, without having to edit the conference.conf 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
Re: [Freeswitch-users] Performance testing: FS and own App?
ah, that sounds interesting. so the echo app is the 9996, right? how can i start/use the media reflector? is it something, i have to call sipp with? sorry for this question, but i am very new in this business. right now i call sipp with: sipp -sn uac xx.xx.xx.xx -s 123456 -r 50 -l 400 -d 4000 -rtp_echo 2008/12/23 Ken Rice : > The echo tester was refering to the echo app in freeswitch > > The media reflector is part of sipp that just echos media back to the > source... That's the proper way to test media handling capabilities > otherwise you are only seeing 1/2 the media stream > > >> From: Dennis >> Reply-To: >> Date: Tue, 23 Dec 2008 11:06:06 +0100 >> To: >> Subject: Re: [Freeswitch-users] Performance testing: FS and own App? >> >> sorry, i do not really understand what you mean with: "Try the echo >> tester but be sure you are using the media refector with sipp or you >> arent doing anything useful". >> >> what is the "echo tester" and what is "media refector" and how could i use >> it? >> >> i would like to find out, how many people can talk to each other over >> the fs server at the same time. a test setup, which simulates real >> calls would be very helpful for me. and then i would like to be able >> to compare the results with others, to see, if everything is working >> as it should. >> >> >> >> 2008/12/23 Ken Rice : >>> Oh! Well who knows how that will affect the performance... I have never >>> tested it with that... Try the echo tester but be sure you are using the >>> media refector with sipp or you arent doing anything useful >> >> ___ >> 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] Performance testing: FS and own App?
sorry, i do not really understand what you mean with: "Try the echo tester but be sure you are using the media refector with sipp or you arent doing anything useful". what is the "echo tester" and what is "media refector" and how could i use it? i would like to find out, how many people can talk to each other over the fs server at the same time. a test setup, which simulates real calls would be very helpful for me. and then i would like to be able to compare the results with others, to see, if everything is working as it should. 2008/12/23 Ken Rice : > Oh! Well who knows how that will affect the performance... I have never > tested it with that... Try the echo tester but be sure you are using the > media refector with sipp or you arent doing anything useful ___ 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] Performance testing: FS and own App?
the 9998 is an extension in the default.xml to test with media flowing through the line. 2008/12/23 Ken Rice : > Whats this 9998 to which you refer? > > >> From: Dennis >> Reply-To: >> Date: Tue, 23 Dec 2008 10:43:30 +0100 >> To: >> Subject: Re: [Freeswitch-users] Performance testing: FS and own App? >> >> because the latest result was with the 9998, it can't be out app (at >> the moment). >> >> so there are no other typical things or settings i could look for? >> >> >> 2008/12/23 Ken Rice : >>> There are a number of issues you can be running into... It really depends on >>> how your app works, what your actual configuration of freeswitch is, disk IO >>> subsystem, ulimits, etc etc >> >> ___ >> 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] Performance testing: FS and own App?
because the latest result was with the 9998, it can't be out app (at the moment). so there are no other typical things or settings i could look for? 2008/12/23 Ken Rice : > There are a number of issues you can be running into... It really depends on > how your app works, what your actual configuration of freeswitch is, disk IO > subsystem, ulimits, etc etc ___ 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] Performance testing: FS and own App?
if i do the same test with the 9998, it does not to seem much better: 20.0(4000 ms)/1.000s 5061 68.25 s 1174 xx.xx.xx.xx:5060(UDP) 0 new calls during 1.008 s period 9 ms scheduler resolution 710 calls (limit 1000) Peak was 782 calls, after 58 s 0 Running, 710 Paused, 0 Woken up 0 out-of-call msg (discarded) 1 open sockets 595580 Total echo RTP pckts 1st stream 2809.336 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg INVITE --> 1174 0 0 100 <-- 1174 0 0 180 <-- 0 0 0 183 <-- 0 0 0 200 <-- E-RTD1 480 0 0 ACK --> 480 0 Pause [ 0ms] 480 0 BYE --> 464 0 0 200 <-- 464 0 0 what settings could i review to get more out of the server? our setup is a new xeon quad core, 4 gb ram and ubuntu 64-bit. we also entered the ulimit lines and set "manage-presence" to false. thanks dennis 2008/12/23 Ken Rice : > Freeswitch can handle a large volume of call... I suggest you review your > configs to make sure you don't have any of the default or arbitrary other > limits in there... We routinely run > 1500 concurrent calls on dual quad > core hardware at call rates far above what you tested at > > Ken ___ 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] Memory question
thanks for the good explaination and for making me feel better :-) 2008/12/23 Ken Rice : > This is normal behavior... FS allocates memory into pools and re-uses that > same memory over and over... It is quite normal to see memory usage increase > as usage of FS increases to a point where it levels off for that load... As > the loa decreases memory is not released but used for later when loading > increases again > > >> From: Dennis >> Reply-To: >> Date: Tue, 23 Dec 2008 09:52:43 +0100 >> To: >> Subject: [Freeswitch-users] Memory question >> >> after doing some testing with fs, i can see in the console, when >> entering "top", that fs uses 9.9% of the memory. >> >> when i do some more calls, the used memory will raise - the memory >> will not beeing released, till i do a restart of fs. >> >> is this a normal behavior or do i have some problems? >> >> >> thanks >> dennis >> >> ___ >> 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] Memory question
after doing some testing with fs, i can see in the console, when entering "top", that fs uses 9.9% of the memory. when i do some more calls, the used memory will raise - the memory will not beeing released, till i do a restart of fs. is this a normal behavior or do i have some problems? thanks dennis ___ 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] Performance testing: FS and own App?
hi, i am quite new to freeswitch and now i finally have fs up and running as i want with my php scripts to handle the calls. now that i want to start the service in the near future, i would like to test the performance of the whole system and the reliability. what are your experiences and ideas, how i could do it the best way? and what do i have to look for? i startet to test with sipp, which seems to be a useful tool, although i am not sure, how to read the results. in the switch.conf i set "max-sessions" to 11000 and "sessions-per-second" to 2000. in the console i entered: sipp -sn uac xx.xx.xx.xx -s 123456 -r 20 -l 1000 -d 4000 -rtp_echo the results are as follow: 20.0(4000 ms)/1.000s 5061 51.18 s 1023 xx.xx.xx.xx:5060(UDP) 20 new calls during 1.001 s period 9 ms scheduler resolution 807 calls (limit 1000) Peak was 807 calls, after 51 s 0 Running, 807 Paused, 0 Woken up 0 out-of-call msg (discarded) 1 open sockets 75989 Total echo RTP pckts 1st stream 481.979 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg INVITE --> 1023 0 0 100 <-- 1023 0 0 180 <-- 0 0 0 183 <-- 249 0 0 200 <-- E-RTD1 249 0 0 ACK --> 249 0 Pause [ 0ms] 249 0 BYE --> 216 0 0 200 <-- 216 0 0 Counter Name | Periodic value| Cumulative value -+---+-- Elapsed Time | 00:00:00:072 | 00:08:30:106 Call Rate |0.000 cps |4.166 cps -+---+-- Incoming call created |0 |0 OutGoing call created |0 | 2125 Total Call created | | 2125 Current Call |0 | -+---+-- Successful call|9 | 2125 Failed call|0 |0 -+---+-- Response Time 1| 00:00:00:000 | 00:02:43:140 Call Length| 00:04:04:169 | 00:02:47:149 i do not get any errors. but i think that it is strange, that there are so little calls running at the same time. when i try to call the same number as sipp does, i have to wait a very long time, till the call is beeing answered. why are there so many paused calls waiting for fs to take care of? is this normal or are there some problems, which i have to take care of? i want to add, that i am using socket outbound with a number of php-scripts, which do a lot of call- and dialplan-handling. i am not using any xml. therefore i am sure, that the results are worse, that they would be with normal xml dialplans. thanks for your help and tipps. dannis ___ 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 Fax: Error, problems and questions...
ahh, just a second. it seems that i did not realize a small missunderstanding in you answer. i do not want to SEND a fax, i just want to RECEIVE a fax. so the fax comes in at out carrier and the rest is sent over about 1m of cat6 to our fs server. is there a difference or does it not matter, if we want to receive or send a fax? 2008/12/19 Anthony Minessale : > You don't know where the audio goes after that switch in the same room until > it gets to the guy > with the fax machine. > > No it will not be improved by Christmas. Not a chance. > > Yes it will probably be much more reliable once it can do T38. > > Be happy with what you have for the holiday season. > > > > On Fri, Dec 19, 2008 at 10:44 AM, Dennis wrote: >> >> it's me again about mod fax... it is short before christmas and my >> whish is, to get mod fax working quite reliable. is this possible >> under optimal conditions? >> >> all our tests lead by far to more failed faxes than received faxes. i >> really like the fax feature and would like to see it beeing usable. >> >> is it just pure luck, if a fax was received or are there some >> conditions out there, which could help beeing mod fax reliable? >> second question: what about t38? will it come? is there chance, that >> it will come? where are the difficulties with mod fax? >> >> our fs servers are standing directly beside the sip switch of our >> carrier. from the carriers switch, there is a 50 cm long cat6 cable >> going into our cisco-switch. from the cisco switch there are 50 cm >> long cat6 cables going into our fs servers. >> i doubt, that there can be a signifant packet loss. >> are there some settings, we could try out or is the faxing stuff just >> unusable, till t38 is supported? >> >> ___ >> 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 > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@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 > > ___ 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 Fax: Error, problems and questions...
hi anthony, thanks a lot for the clear answer. that is something i can work with :-) i also want to thank you for the great support you gave us within the last months and the great freeswitch. our fs servers are up and running and everything works great (only fax is not working). have a nice christmas (till i contact you because of some consulting for final checks ;-) dennis 2008/12/19 Anthony Minessale : > You don't know where the audio goes after that switch in the same room until > it gets to the guy > with the fax machine. > > No it will not be improved by Christmas. Not a chance. > > Yes it will probably be much more reliable once it can do T38. > > Be happy with what you have for the holiday season. > > > > On Fri, Dec 19, 2008 at 10:44 AM, Dennis wrote: >> >> it's me again about mod fax... it is short before christmas and my >> whish is, to get mod fax working quite reliable. is this possible >> under optimal conditions? >> >> all our tests lead by far to more failed faxes than received faxes. i >> really like the fax feature and would like to see it beeing usable. >> >> is it just pure luck, if a fax was received or are there some >> conditions out there, which could help beeing mod fax reliable? >> second question: what about t38? will it come? is there chance, that >> it will come? where are the difficulties with mod fax? >> >> our fs servers are standing directly beside the sip switch of our >> carrier. from the carriers switch, there is a 50 cm long cat6 cable >> going into our cisco-switch. from the cisco switch there are 50 cm >> long cat6 cables going into our fs servers. >> i doubt, that there can be a signifant packet loss. >> are there some settings, we could try out or is the faxing stuff just >> unusable, till t38 is supported? >> >> ___ >> 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 > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@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 > > ___ 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 Fax: Error, problems and questions...
it's me again about mod fax... it is short before christmas and my whish is, to get mod fax working quite reliable. is this possible under optimal conditions? all our tests lead by far to more failed faxes than received faxes. i really like the fax feature and would like to see it beeing usable. is it just pure luck, if a fax was received or are there some conditions out there, which could help beeing mod fax reliable? second question: what about t38? will it come? is there chance, that it will come? where are the difficulties with mod fax? our fs servers are standing directly beside the sip switch of our carrier. from the carriers switch, there is a 50 cm long cat6 cable going into our cisco-switch. from the cisco switch there are 50 cm long cat6 cables going into our fs servers. i doubt, that there can be a signifant packet loss. are there some settings, we could try out or is the faxing stuff just unusable, till t38 is supported? ___ 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] Redirecting a call from one FS to another FS?
sendmsg redirect to an ip-adress of one of our fs server works great. thanks for your help. dannis ___ 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] Redirecting a call from one FS to another FS?
thanks for all your help! this sounds interesting. it seems, that these codes should be available by default with sip!? is this right? i will talk to the carrier tomorrow and ask, what is possible. as far as i can see, i am always dependant on the carrier? there is no way to pass a call from one fs to another? 2008/12/18 Raymond Chandler : > Gabriel Kuri wrote: >> I've tried to do the same and in my own experience, most carriers don't >> accept 302 redirects. What I've seen is they take the 302 as a failure >> and move on to the next switch, so worse case with 3 switches, it will >> take 2 retries before hitting the switch you want them to redirect to. >> >> > > could also just respond with a 503 in which case all carriers should > fail over to the next one... > > -Ray > > ___ > 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] Redirecting a call from one FS to another FS?
so at least they should react on a 302? this could help, although i do not really understand, what happens on a 302. if they support it, they would receive the target fs server ip where they should try next with deflect? if everything does not help and is not possible: what could i do else? it would be very helpful, if fs would support another way, if the provider does not offer specific features. 2008/12/18 Gabriel Kuri : > I've tried to do the same and in my own experience, most carriers don't > accept 302 redirects. What I've seen is they take the 302 as a failure > and move on to the next switch, so worse case with 3 switches, it will > take 2 retries before hitting the switch you want them to redirect to. > > Gabe > > Dennis wrote: >> i would like to know, what the best way is, to redirect an incoming >> call from one fs (fs1) to another fs (fs2). >> >> we use 3 freeswitch servers and the carrier passes calls to the three >> fs servers randomly. if on fs server is not offline, the carrier sends >> the call to the next fs. >> this is generally good, but for conferencing it not so good. >> >> i am using socket outbound and need to do this for conferencing. let's >> say, we have a conference going on on fs1. another person wants to >> enter this conference, but the call is passed to fs2. on fs2 we see, >> that the caller wants to enter the conference going on on fs1. >> >> now we have to redirect the call from fs2 to fs1. is this done with >> "redirect" and some according settings/params or are there other ways >> to do this? we would like to do this without our carrier doing >> something, to be a little more independant. >> >> thanks >> dennis >> >> ___ >> 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] Redirecting a call from one FS to another FS?
sorry, this is to difficult for me. what does that mean? they pass a call to one of our fs. then we see, that the call should be on another fs. we know, that the call is on the wrong fs, before we send an answer. so we could react accordingly. 2008/12/18 Brian West : > do they follow a 302 redirect? Because if the call isn't answered yet > then you can do a redirect ___ 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] Redirecting a call from one FS to another FS?
so if they do not suport it (which has to be seen), is there another way to redirect a call from one fs to another without the provider? like redirect from one fs to the other over the local lan? 2008/12/18 Brian West : > Well its a standard SIP Refer, They may not support it for good reason. > > /b > > On Dec 18, 2008, at 11:07 AM, Dennis wrote: > >> is deflect, what i understand? the provider has to support it? if yes, >> what could i tell and ask the provider, to find a solution to this >> problem? the provider is quite open for new ideas, although we do not >> want to be to dependant on the provider and his possibilities. > > > ___ > 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] Redirecting a call from one FS to another FS?
sorry, i do not know that. i could ask tomorrow. is deflect, what i understand? the provider has to support it? if yes, what could i tell and ask the provider, to find a solution to this problem? the provider is quite open for new ideas, although we do not want to be to dependant on the provider and his possibilities. 2008/12/18 Brian West : > What switch is your provider using? > > /b > > On Dec 18, 2008, at 10:52 AM, Dennis wrote: > >> i had a look at the deflect app, but as far as i understand it, the >> carrier has to support/understand it ans react on the signals. >> >> is that right or does this have nothing to do with our carrier? or >> does this work between fs servers in the same local network? >> >> >> another similar question is: how to reject calls, so that the carrier >> tries to route the call to another fs? if we want to make changes to >> fs and test them, we would like to block new incoming calls, till >> there are no running calls, to shut down the fs. > > > ___ > 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] Redirecting a call from one FS to another FS?
i had a look at the deflect app, but as far as i understand it, the carrier has to support/understand it ans react on the signals. is that right or does this have nothing to do with our carrier? or does this work between fs servers in the same local network? another similar question is: how to reject calls, so that the carrier tries to route the call to another fs? if we want to make changes to fs and test them, we would like to block new incoming calls, till there are no running calls, to shut down the fs. 2008/12/18 Brian West : > the deflect app. > > /b > > On Dec 18, 2008, at 10:36 AM, Dennis wrote: > >> i would like to know, what the best way is, to redirect an incoming >> call from one fs (fs1) to another fs (fs2). > > > ___ > 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] Redirecting a call from one FS to another FS?
i would like to know, what the best way is, to redirect an incoming call from one fs (fs1) to another fs (fs2). we use 3 freeswitch servers and the carrier passes calls to the three fs servers randomly. if on fs server is not offline, the carrier sends the call to the next fs. this is generally good, but for conferencing it not so good. i am using socket outbound and need to do this for conferencing. let's say, we have a conference going on on fs1. another person wants to enter this conference, but the call is passed to fs2. on fs2 we see, that the caller wants to enter the conference going on on fs1. now we have to redirect the call from fs2 to fs1. is this done with "redirect" and some according settings/params or are there other ways to do this? we would like to do this without our carrier doing something, to be a little more independant. thanks dennis ___ 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] Two major flaws: Could they be fixed?
thanks, now it works as we expected. and thanks for the hint, how we should send the hangup with sendmsg. we will do it your way :-) 2008/12/8 Anthony Minessale <[EMAIL PROTECTED]>: > channels in originate were not checking for private events. > now they should but if send them commands to do crazy stuff like play a file > while they are > in the middle of originating there could be ill side effects (e.g. play file > before media was established etc which could cause the call to abort) > > btw you can send > > call-command: hangup > hangup-cause: normal_clearing > > in place of > call-command: execute > execute-app-name: hangup > execute-app-arg: normal_clearing > > > On Mon, Dec 8, 2008 at 10:56 AM, Dennis <[EMAIL PROTECTED]> wrote: >> >> > you would get a hangup event in either case. >> >> yes, you are right. we just tested and saw that. the reason for >> sendmsg hangup, was the sometimes useful event-lock. >> >> it works with api uuid_kill as we wanted. but with sendmsg hangup it >> still does not work. shouldn't sendmsg hangup work like uuid_kill >> here? how useful could it be, to let it ring, when the hangup was >> already sent and is immediately executed when the anser is sent? >> >> >> #2 now works perfectly. thanks for the great support! >> >> ___ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED] > pstn:213-799-1400 > > ___ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ 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] Two major flaws: Could they be fixed?
> you would get a hangup event in either case. yes, you are right. we just tested and saw that. the reason for sendmsg hangup, was the sometimes useful event-lock. it works with api uuid_kill as we wanted. but with sendmsg hangup it still does not work. shouldn't sendmsg hangup work like uuid_kill here? how useful could it be, to let it ring, when the hangup was already sent and is immediately executed when the anser is sent? #2 now works perfectly. thanks for the great support! ___ 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] Two major flaws: Could they be fixed?
i have to shift places. will be back in a few minutes and test. no, we are using the simple sendmsg uuid hangup. as far as we remember, we do not use api uuid_kill, because we do not get a hangup event with this. 2008/12/8 Anthony Minessale <[EMAIL PROTECTED]>: > try the sendmsg issue again > > are you doing the hangup with > > api uuid_kill ___ 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] Two nice to have features in fs
great, that works! thanks a lot!
just tested the changes according an error, when a file is missing.
thanks again!
2008/12/8 Anthony Minessale <[EMAIL PROTECTED]>:
> done
>
> On Mon, Dec 8, 2008 at 9:18 AM, Dennis <[EMAIL PROTECTED]> wrote:
>>
>> > Huh?
>>
>> src/switch_core_session.c vom line 899 to 901:
>>
>>if (seconds < 10) {
>>seconds = 60;
>>}
>>
>> ___
>> Freeswitch-users mailing list
>> Freeswitch-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:[EMAIL PROTECTED]
> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[EMAIL PROTECTED]
> iax:[EMAIL PROTECTED]/888
> googletalk:[EMAIL PROTECTED]
> pstn:213-799-1400
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
___
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] Two major flaws: Could they be fixed?
> #2 was because when you sendmsg with no uuid on an outbound socket it > defaults to the session who called you. > I changed to code to make a distinction between not supplying a uuid and > supplying an invalid uuid. anthony, thanks for the quick reaction! we just tested you changes and it works the opposite way it should. this means: when we do not send an uuid, we get an an error (Reply-Text => -ERR invalid session id []). if we send a wrong/not existing uuid, the command will be executed on the inbound uuid. > #1 seems hard to believe. Please provide a console trace of the channel > *ignoring* the hangup command. i know it is hard to believe, we didn't believe it either ;-) have a look at http://pastebin.freeswitch.org/6367 what we simply do here: the inbound is coming in, then we do an originate and hang up the inbound. then we directly send a hangup for the outbound. the outbound will go on ringing. then, when the ringing outbound is answered, we directly get the hangup. fs gets the hangup and remembers it, but seems to wait till the answer to execute this command. ___ 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] Two nice to have features in fs
> Huh?
src/switch_core_session.c vom line 899 to 901:
if (seconds < 10) {
seconds = 60;
}
___
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] Two nice to have features in fs
hi, we are using socket outbound and found out, that there are two (perhaps) small things, we would like to see in fs. 1.) if we try to playback a soundfile, which does not exist, we do not get an error or something. in the cli of fs we can see the error, but we do not get anything about it over the socket. we get an execute event for the playback and directly after it an execute-complete. wouldn't it be nicer, if one could get a notice about an error, like: execute-complete-error or a variable, which is set with an error, if an error occours? 2.) the session heartbeat event is an absolutely great feature in fs. but sometimes we would like to get the session heartbeat every 5 seconds. at the moment the allowed minimum is every 10 seconds. a smaller setting, like every 5 seconds, will result in a session heartbeat of every 60 seconds. to help us out, we edit the switch_core_session.c at line 899 and change the "seconds < 10" to "seconds < 5". because one has to set "enable_heartbeat_events=5" manually, i do not think, that there is a risk, that others, who do not want the heartbeat to come that often, will be negatively affected by this change. might it be possible, to do the same changes to the default code? thanks dennis ___ 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] Two major flaws: Could they be fixed?
hi, we are fighting with two flaws in fs and would be happy, if they could be fixed. we are using socket outbound. 1.) hangup a call in ringing state: this worked in one of the last fs versions, but suddenly does not work anymore. let's say, we have an inbound call and do 3 originates to different targets. all 3 targets are in ringing state. the target, which answers first, will be bridged with the inbound call, the other two (still ringing) targets should be hung up. we do not want fs to hang up the other two originates automatically. we want to hang up the other two originates by sending the hangups. we set "hangup_after_bridge=false" and "park_after_bridge=true". we do sendmsg hangup uuid. but the originates are first hung up, when they are answered. when they are in ringing state, the hangup will do nothing (anymore). as i said, it worked before, so i assume, that something has changed in the latest trunks. 2.) sendmsg uuid *whatever* can cause to excute the command on the wrong uuid: let's say, we have an inbound call and an outbound call - at least we thing we have it ;-). now we do for example sendmsg outbound_uuid hangup to hangup the outbound call. but, if the uuid of the outbound call does not exist (because there was a problem or something), the inbound will be hung up instead. the same happens with all sent messages to an uuid, which does not exist. if we want to do a playback for the same outbound, the inbound will hear it, if the outbound_uuid does not exist. perhaps this is a feature, but i think that it would be nicer and more reliable, if the sendmsg is only executed on the given uuid. if the given uuid does not exist, nothing should happen or even nicer, an event with an error should be sent to the socket. thanks dennis ___ 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] Playing a file again and again
although i do everything with socket outbound and php and not with the xml-dialplans, i feel, that you are missing the basics. first you should find out how to make a basic dialplan and how fs is working. then you can start with playing soundfiles and then, how to do a looped playback and other nice built in features. there are lots of dialplan samples delivered with fs and the wiki will help you to start with the rest. dennis 2008/12/6 Faisal Maqsoodi <[EMAIL PROTECTED]>: > I need some more help. I used send msg this way. Is there anything missing > bcoz its not working. Plz let me know what else should i do. When i dial > 2003, file is not played. > > sendmsg 2003 > call-command: execute > execute-app-name: playback > execute-app-arg: /usr/local/freeswitch/sounds/enter_plistnum.wav > loops: -1 > > > --- On Sat, 12/6/08, Dennis <[EMAIL PROTECTED]> wrote: > > From: Dennis <[EMAIL PROTECTED]> > Subject: Re: [Freeswitch-users] Playing a file again and again > To: [EMAIL PROTECTED] > Date: Saturday, December 6, 2008, 2:13 AM > > with sendmsg playback send: loops: -1 > > 2008/12/6 Faisal Maqsoodi > <[EMAIL PROTECTED]>: >> Hi, >> Is there any built-in function, like playback, which plays a file > again >> and again unless interrupted. I want to use a simple function not FIFO. >> >> Faisal >> >> >> ___ >> 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] Playing a file again and again
with sendmsg playback send: loops: -1 2008/12/6 Faisal Maqsoodi <[EMAIL PROTECTED]>: > Hi, > Is there any built-in function, like playback, which plays a file again > and again unless interrupted. I want to use a simple function not FIFO. > > Faisal > > > ___ > 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] Mod Fax: Error, problems and questions...
2008/12/5 Steve Underwood <[EMAIL PROTECTED]>: >> 1.) there is one error, we get always - no matter, if the fax was sent >> successfully or not. >> in the pastebin under http://pastebin.freeswitch.org/6338 you can see >> the error in the last line. >> this is the full log of a fax in fs console loglevel debug. >> > That looks like something annoying but harmless. It looks like the comms > path is disabled slightly before the flow of packets is turned off. > That's probably just a silly slip in the code. yup, because the error always appear, i successful or not, this error can't be a big problem. i just do not like red lines in the log ;-) > Does Fritz FAX means the ISDN card stuff? If so, that should be > something well proven. However, the errors you are getting sound like > the FAX at the far end is buggy. I think a log of the audio from one or > two of these calls is needed for analysis. yes, fritz fax is the isdn stuff. normally it works very well. how can i get a log of the audio? when a fax is coming in, there happens quite little in the console at loglevel debug. i pasted all into the pastebin. if there are more possibilities to get mor information, please let me know. ___ 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 Fax: Error, problems and questions...
my server (including me) is your slave. anthony already feels home on my server, so you are greatly invited ;-) it is quite late in germany, so i feel, that we should meet in irc tomorrow, if this is ok for you. @brian we get everthing over sip. so we receive the faxes over sip. the faxes, which we send for testing (not over fs or the same machine) are sent over isdn. 2008/12/4 Michael Collins <[EMAIL PROTECTED]>: > Dennis, > > Thanks for your input on the fax stuff! We will check this out and report > back. > > Question: if one of the devs would like to SSH into your system to do > further testing, is that okay? ___ 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] Mod Fax: Error, problems and questions...
hi, after we managed to setup fs with mod_fax and our socket outbound script, we have some questioons about an error and problems, when sending a fax: 1.) there is one error, we get always - no matter, if the fax was sent successfully or not. in the pastebin under http://pastebin.freeswitch.org/6338 you can see the error in the last line. this is the full log of a fax in fs console loglevel debug. 2.) fax works quite good. we couls send long faxes over a normal fax machine without any problem. but for fast testing we are using a softfax (fritz fax). here we have some more problems. mostly a fax with one page will pass through, but more pages will mostly fail. because we are new to the fax thing, we do not really know, what the messages tell us about failed faxes. here are the top 3 messages (unordered and always one at a time - nerver at once): variable_fax_result_text => Received a DCN while waiting for a DIS fax_result_text => The HDLC carrier did not stop in a timely manner fax_result_text => Unexpected message received could someone please tell us, where the problem might be? thanks dennis ___ 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] Fax and Freeswitch: What is the status, what works?
hi, because we do not get tired of testing and playing a lot with the beloved fs, we now arrived at the fax feature :-) i am not sure if the docs are up to date or if there was a lot of development in the meantime. therefore i would like to ask, what is possible and what will come in the near future. we are using fs, socket outbound and php and would like to make something like fax to mail as an additional service. is t38 supported? can i pass incoming faxes over the same socket as calls? can i convert faxes into pdf? is fax over sip reliable (as far as i have heard, under asterisk fax is nothing one should use)? and so on, and so on i would be very happy to hear some user experiences with fs and fax. if it seems, that we can use fax with over socket outbound, we will do hardcore testing ;-) thanks, dennis ___ 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] Listen to a file, while recording?
sorry, problem solved :-) it works very good with icecast2. 2008/12/2 Brian West <[EMAIL PROTECTED]>: > And you have your shoutcast/icecast server set up and functional? > > /b > > On Dec 2, 2008, at 9:03 AM, Dennis wrote: > >> i am using the latest svn trunk from today. > > > ___ > 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] Listen to a file, while recording?
no, not yet. i am still fiddling arround with icecast2. we tried it with someone, who offers radiostreams. perhaps this just works with icecast(2) and shoutcast? 2008/12/2 Brian West <[EMAIL PROTECTED]>: > And you have your shoutcast/icecast server set up and functional? > > /b > > On Dec 2, 2008, at 9:03 AM, Dennis wrote: > >> i am using the latest svn trunk from today. > > > ___ > 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] Listen to a file, while recording?
i am using the latest svn trunk from today. 2008/12/2 Brian West <[EMAIL PROTECTED]>: > Are you on SVN trunk or what rev are you trying to use? > > /b > > On Dec 2, 2008, at 7:48 AM, Dennis wrote: > >> it seems, that fs has to stream to recording file to a streaming >> server (like icecast), right? but if we do "api uuid_record uuid start >> shout://user:[EMAIL PROTECTED]:12345/" (and other combinations), we get >> an error: >> 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid >> URL: x >> 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851 >> switch_ivr_record_session() Error opening shout:// > > > ___ > 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] Listen to a file, while recording?
we configured mod_shout and are able to record mp3. but if we start to playback the file, it will only be played back to that point, which was recorded, when we started the player. we do this with "api uuid_record uuid start /var/www/test.mp3". we are also able to playback a (radio-)stream to an uuid with shout://ip-adress:12345 but what do we have to do, to listen to the file/stream with a player? it seems, that fs has to stream to recording file to a streaming server (like icecast), right? but if we do "api uuid_record uuid start shout://user:[EMAIL PROTECTED]:12345/" (and other combinations), we get an error: 2008-12-02 14:28:38 [ERR] mod_shout.c:730 shout_file_open() Invalid URL: x 2008-12-02 14:28:38 [ERR] switch_ivr_async.c:851 switch_ivr_record_session() Error opening shout:// are we on the right track? is there something else we have to do to make it work? thanks for your help. 2008/12/1 Anthony Minessale <[EMAIL PROTECTED]>: > yes, > > mod_shout will broadcast calls as MP3 that you can listen to in > itunes/winamp live. ___ 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 A terminated by Leg B on a uuid_bridge
hi simon, i am not sure, if i understood your problem right, but if you do not want leg a to hang up after leg b (the originated call) hangs up, set "park_after_bridge=true" when you make the originate. as far as i know, "hangup_after_bridge=false" is only for the inbound and helps nothing with the outbound. if you want something different, please explain me a little more. dennis 2008/11/28 Simon Tang <[EMAIL PROTECTED]>: > Hello, > > > > I'm using event socket outbound, and have an issue where, after a bridge > ends and is terminated by Leg B, Leg A is also terminated. Here's the call > flow: > > > > 1. Call comes in (Leg A), session created, play welcome message. > > 2. From this session, originate and dial out using api originate > > 3. After the target answers (Leg B), bridge the 2 calls using api > uuid_bridge > > 4. Leg B hangs up. > > 5. Leg A will be terminated. > > > > After step 4, Leg A is terminated. I do not want Leg A to hang up. I've > tried setting "hangup_after_bridge=false" prior to the call, and that > doesn't work. > > > > Having said that, I tried a similar test which does not end Leg A's call > after Leg B hangs up, but I can't use this solution because, functionally, > does not accomplish what I want it to do (i.e., I want to perform some > actions on Leg B prior to the bridge, like send some DTMF tones, playback > some messages, etc). I did not need to set the "hangup_after_bridge" > variable (default should be false anyway). > > > > 1. Call comes in (Leg A), session created, play welcome message. > > 2. From this session, do a bridge by doing an execute bridge. > > 3. The target answers (Leg B) > > 4. Leg B hangs up. > > 5. Leg A will still be active. > > > > Any ideas would be appreciated. Thanks! > > > > Simon > > ___ > 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] Listen to a file, while recording?
so i would have to make a call with a phone to a specific dialplan? if so, this would not be, what i whished (although it is nice to have the option). isn't there something, which can stream the voice of a given uuid? so i could place a link in the html admin-area to spy an uuid and to hear everything over the speaker? this would be really sexy ;) 2008/11/18 Anthony Minessale <[EMAIL PROTECTED]>: > you can use the eavesdrop dialplan app from a new call to spy on an in > progress session > it takes the uuid of the channel you want to listen to as the arg. ___ 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] send_dtmf: Understanding question...
hi, i wonder, if or what i do not understand how to do send_dtmf in the right way. for example i want to send dtmf tones to my mobile mailbox the enter the menu and do some changes to the settings. but whatever i try, it does not work. i tried: sendmsg send_dtmf outbound_uuid 123# sendmsg playback outbound_uuid tone_stream://123# sendmsg gentones outbound_uuid 123# api uuid_send_dtmf outbound_uuid 123# [bleg] i am using socket outbound. if i test from one phone to another phone, i will not get any events about incoming dtmf-tones. so i myself could not even react on sent dtmf tone. if i press a digit on the phones, i get events about that. fs generates the tones: i can see them in the console and i can hear them. but the other side does not seem to receive them - at least not right. i always get the message, that the login was wrong. do i have to uncomment something in sofia conf? ___ 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] Big problems with disconnects
hi, i have big problems with disconnects, when bridging and unbridging calls. because i had random diconnects when testing fs from one softphone to the other, i set up a little dialplan (socket outbound), to do some hardcore testing. after the inbound is answered, i do an originate to an outbound. when the outbound answers, i bridge the inbound and outbound. then i do uuid_transfer, 5 seconds later a uuid_bride, 5 seconds later a uuid_transfer and so on and so on. latest after 5 minutes of switching, the connection seems to hang for about 29 seconds and then ends. this ALWAYS happens after the debug message: switch_core_state_machine.c:187 switch_core_standard_on_park() sofia/internal/98765 Standard PARK see http://pastebin.freeswitch.org/6281 at line 82. i repeated that for at least 30 times - always with the same result. what could be the problem? 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] Listen to a file, while recording?
hi, i would like to be able to listen to conversations, while they are ongoing. this should not happen over a phone. i would like to be able to have a link or something in my admin-area, where i can click, if i want to listen to a conversation. i thought about to start a record with socket inbound on a specific uuid and while the recording is done, i would like to play the file. but i have the feeling, that this is not possible, while the file is not finished. is there a way to stream the audio over the web? thanks for your help. ___ 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] Problems with DTMF and * on inbound leg
i am using socket outbound and if an inbound call comes in, i answer the call and play a soundfile for the caller. if the caller presses the dtmf key * the playback is stopped and i receive an channel_execute_complete playback for this file. this only happens with the * - the other keys do nothing (as i whish). i there somewhere a default setting for this, which i can turn off? i looked through all conf-files, but could not find enything. for my tests i use the default 1000 user as inbound. in the ivr.conf.xml i found something with menu-top and digits *. i commented this line out, but this didn't change anything. thanks for your help. ___ 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] Conference and socket outbound
hi, now, that everything is working, what i wanted to do with freeswitch, i want to look a little bit more left and right. one interesting thing to "play" arround with, is conferencing. is it possible to do conference calls, when using socket outbound and php? where could i start? ok, of course with "mod conference" in the wiki. but i wonder, how i can put a caller in a conference room and how to set the properties (what dtmf does what and which soundfiles are played when) dynamically. let's say i get a socket inbound call and i answer it. then i play a soundfile, which asks to enter the pin... how can i move the inbound call or originated calls into a conference room? it seems, as if one can do a lot with api in the conference room. but i wonder what i have to do in the beginning, to put people (specific uuid's) into a conference room. thanks, dennis ___ 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] Optimal server hardware and software setup (freeswitch, php & mysql)?
hi, i am about to set up servers for freeswitch with php and mysql. i would like to ask, what the optimal setup would be. planned software: - freeswitch - php-script as socket server (socket outbound) - mysql (for call-logfiles and dialplan-selection) - ubuntu server 64bit planned hardware: => 2 servers for web, the php socket server script and mysql - each with 2 xenon quad cores, 16 gb ram => 3 servers for freeswitch - each with 1 xenon quad core, 4 or 8 gb ram one of the web-server could smoke off and the other server jumps in. two freeswitch server could smoke off and the last can handle all calls. or one of the freeswitch server could be taken out of the row to test new versions. for me it is important, that everything is very redundant, because i want to avoid trouble with clients. the big questions are: 1.) is this hardware setup ok? 2.) where are the weak spots, when using an app like freeswitch (or something similar)? we would prefer, if each of the three freeswitch servers could stand to have its own php socket server to be run and to have a copy of the mysql database to READ the needed information. this is because the web server or other servers could fail, while each freeswitch server can go on handling call independantly. but, would this setup cause any problems in case of stability or speech quality? i have no idea, what task i should keep away from the application servers. thanks, dennis ___ 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] Socket outbound: How to bridge two calls?
we read every single reply and we make socket_read with the legth returned by Content-Length. what we can do with fs, socket outbound and our php-script: 1.) we can answer the inbound, make a bridge to another phone and both lines are connected and can talk to each other. 2.) we can make an inbound, then make two bgapi originates and uuid_bridge both originated calls, so that they can talk to each other. 3.) we made some thausend testcalls with sipp over the socket to our script with 30cps and had not one failed call. the only thing we can not do: => make an inbound call, then make an originate and uuid_bridge the inbound and outbound. i just can't imagine where the problem could be. could we talk about some kind of consulting to take a little deeper look into this problem? 2008/11/10 Anthony Minessale <[EMAIL PROTECTED]>: > you must read the reply to the commands when you send them or you will block > the tcp socket. ___ 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] Socket outbound: How to bridge two calls?
you are right, the shown script is very simple. we shortened it a lot just to show the problem. in these tests we do not need any events, because we just answer and originate - for these actions we do not need any events, filters and so on. the rest we do in the cli. in the cli we do show channels and uuid_bridge. but the uuid_bridge is only done, when we kill the process of the socket script. there must be a problem, how we handle the inbound or something like this. every alse we want to do with any call and uuid works, so i can't imagine, that our php script does everything so wrong. what makes uuid_bridge or intercept so different to other api commands? isn't it possible to see something in the logfiles? perhaps, that the inbound is in a wrong state or stuck somewhere in the socket? 2008/11/10 Anthony Minessale <[EMAIL PROTECTED]>: > The way your script is parsing appears too simple. > The event socket has a specific protocol. > > Every command has a reply and you must read it and take into account the > content len etc. > > all replies, events etc have a content-type and content-len. ___ 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] Socket outbound: How to bridge two calls?
sorry for my late answer, i had to fly to a different place... we are quite sure, that we have a problem with our php script. we stripped down the code to the absolute minimum, to remove most possible error sources, but it still does not work. the code can be found under http://pastebin.freeswitch.org/6059. we have a very strange behavior: we get an inbound call over the socket to our script, we send an answer and then we make an originate - nothing more. the full fs debug can be found under http://pastebin.freeswitch.org/6060. then we change to the cli, make show channels and then uuid_bridge both uuid's. the result is "State RESET going to sleep". but, as soon as we kill the process of the script, the uuid_bridge is done. i have no idea where what the reason for the problem could be. as i said, everything else works, but to make an uuid_bridge, we have to kill the php process!? thanks dennis 2008/11/7 Anthony Minessale <[EMAIL PROTECTED]>: > Can you capture the whole log from the instant you get the inbound call > until you give up on the uuid bridge? > I don't see any of the log about your outbound call. > > are you doing > > api originate sofia/internal/[EMAIL PROTECTED] &park() > > For the outbound call like the cli example? > > The socket application *is* park. You do not need to tell it to park. > > as soon as you get the new inbound connection from calling socket app > all you need to do is answer it. > > In your code when you fork are you closing the socket in the parent side so > it's not > in use by both processes? ___ 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] Socket outbound: How to bridge two calls?
an addition: if we make a call to our socket (inbound), we get the following error in our log: 2008-11-07 16:58:57 [ERR] switch_ivr.c:498 switch_ivr_park() Cannot park channels that are under control already. if an inbound comes in, we send a connect, then a park and then an answer - nothing else. directly after we sendmsg park, we get the error above... ___ 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] Socket outbound: How to bridge two calls?
anthony, brian, you are right. it really works the way anthony described over the cli. under http://pastebin.freeswitch.org/6038 you can find the debug log of the working uuid_bridge and of another not working uuid_bridge over the socket and our script. in our second test we made the inbound over the socket and an originate from our script. then we changed into the cli, entered show channels and made an uuid_bridge with the two shown uuid's. this did not work. the call goes immediately into sleep. but the two shown uuid's must be correct, right? is there a way to see in the debug, where our problem might be? what do we have to look for? we start to get creazy... is it possible, that the inbound is in a wrong state, so that we have those problems? > maybe your event socket client code is not coded right? this might be the problem, but how could we check that? > Are you adding 2 to the end of every event? yes, we do that. > Are you reading in the appropriate bytes from the Content-Length header? till now we used 2048, but we also tested it with 4096 the reason, why we have problems to understand the problem is, that EVERYTHING else works. we can do with the inbound and the outbound, whatever we want. we can play different soundfiles to the different uuid's, we can hangup calls with a specific uuid and so on. thanks for you patience dennis ___ 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] Socket outbound: How to bridge two calls?
ok, now i understand to way intercept should work (i was always wonderung, why there only was one uuid). BUT, it still does not work. Could you please be so kind and have a look at http://pastebin.freeswitch.org/6033. there you can see the results of our latest tests. 1.) intercept with inbound-uuid and (originated) outbound-uuid (now the right way) => does not work for us. 2.) uuid_bridge with inbound-uuid and (originated) outbound-uuid => does not work for us. this does not work either. we do not get any events with channel_bridge or something. after we make a bridge, nothing happens till we hang up. might it be possible, that there is a reason that i does not work and that the reason is the same because intercept does not work? 3.) uuid_bridge with two originated calls => WORKS we have an inbound call and then we make two originates. then we uuid_bridge both originated calls. we can talk to each other and get all channel events like bridge and unbridge. it seems, that there is a problem, when we want to do the whole thing with the inbound. the only problem is, that we originate both calls with hangup_after_bridge=false, but as soon as one of the bridged originated calls hangs up, the other call is also beeing hangup. we just can't see, what we are doing wrong or what we could try or change to make it work. thanks, dennis 2008/11/6 Anthony Minessale <[EMAIL PROTECTED]>: > you are missing execute-app-arg > > sendmsg > call-command: execute > execute-app-name: intercept > execute-app-arg: http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Socket outbound: How to bridge two calls?
but what could be the cause, that there is no uuid in the intercept? we are sending the uuid of the outbound with the intercept. as you can see under http://pastebin.freeswitch.org/6019 in the "test intercept" part, we send the message with the uuid and get the answer below (with intercept, outbound and the uuid of the outbound -> in_uuid is the inbound uuid). so i can not understand, why the uuid of the intercept is empty. i was testing with two softphones, which are connected to the standard 1000 and 1001. this might be the reason, why there were 4 calls in the cli. i now tested over our carrier, so that there are only two uuid's. the result is the same with only two possible uuid's (see log in http://pastebin.freeswitch.org/6019). i really have no clue, what i am doing wrong. thanks a lot dennis 2008/11/6 Anthony Minessale <[EMAIL PROTECTED]>: > in your intercept example there is no uuid (note the empty () where the > uuid should be and the syntax error) > > 2008-11-06 16:17:26 [DEBUG] switch_ivr.c:391 switch_ivr_parse_event() > sofia/internal/[EMAIL PROTECTED] Command Execute intercept() > 2008-11-06 16:17:26 [ERR] mod_dptools.c:256 intercept_function() Usage: > [-bleg] > > in the cli example you specified one of the uuids from the wrong call it > should have been. > > b05f58aa-ac16-11dd-9f4b-f1b593b2c0b8 b0714c7c-ac16-11dd-9f4b-f1b593b2c0b8 > > you had an existing bridged call and you used one of the uuid from that call > instead of your parked calls. > > the one you specified was in a bridge already. ___ 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] Socket outbound: How to bridge two calls?
hi anthony, i just pasted the results of 3 unsuccessful tests into the pastebin: http://pastebin.freeswitch.org/6018 perhaps you could be so nice and have a look at it... in the first test i made an uuid-bridge over our socket script. in the second test i made an originate and then an intercept (which shows an error in the debug log). in the last test, i entered to uuid-bridge into the cli. some more information arround the whole thing: 1.) the uuid's must be correct, because we can do everything with the inbound and the outbound by sending messages with the uuid of the leg, we want to talk to. 2.) if we make two originates, so that we have 2 outbound legs, we are able to uuid-bride the two outbound legs with each other. it simply does not work with the inbound... thanks dennis 2008/11/6 Anthony Minessale <[EMAIL PROTECTED]>: > If that doesn't work one of your uuids are wrong or contains a superfluous > space or some other character > You have to understand that the inbound call that uses "socket" is the same > thing as &park() > They are both in a park state at that point and there is no functional > difference. > > once both channels are answered and in the park state, > you have 2 choices and they both work, so watch the console log on debug > level to find your mistake. > > api uuid_bridge > sendmsg to with call-command: execute to execute intercept on > > > to prove it, just set your script to accept and answer the inbound call then > make the outbound call and do nothing more. > Then at the CLI type: >>console loglevel debug >>show channels > > get the 2 uuid from the output > and type: >>uuid_bridge > > if this does not work, send me a complete copy of the console output ___ 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] Socket outbound: How to bridge two calls?
hi birgit,
thanks for your reply!
some days ago anthony told me about "intercept" in the irc, but i seem
not to be able to use it, at least not for connecting two channels
with each other.
we need to first originate, because we want to have complete control
over all channels and chose, what to do with them.
what we have, is an answered inbound channel and an originate at
&park(). let's say, we play a soundfile for the originated channel
(outbound) and then want connect both channels with each other.
when we sendmsg intercept (the uuid is the uuid of the
inbound), we get channel_execute intercept and
channel_execute_complete intercept for call-direction outbound. but we
can not talk to each other and won't get any more information about
the status.
do you have a clue, what the problem could be or what we could try else?
regards,
dennis
2008/11/5 Birgit Arkesteijn <[EMAIL PROTECTED]>:
> Hi Dennis,
>
> I had a similar problem, but we're using javascript, not sockets.
>
> Is there any reason why your first original and then bridge instead of
> bridging your inbound call immediately to the (what will be your)
> outbound call?
>
> If that doesn't work, I had to take a channel out of a conference call
> and bridge that channel to a new person. I used 'intercept' in the end:
>
> var sSession = new
> Session("{ignore_early_media=true,originate_timeout=10}" +
> customer_service_url);
> sSession.setAutoHangup(false);
> sSession.execute("intercept", uuid);
>
> where sSession is the newly (latest) originated call
> and uuid the identifier of the channel in the conference call.
>
>
> As far as I know, 'unbridge' as command doesn't exist.
> I wouldn't know how to achieve that.
>
> Hope the above helps.
>
> Cheers, Birgit
___
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] Socket outbound: How to bridge two calls?
hi, i am using socket outbound and want to bridge two calls with each other. i do the following: a call comes in and it gets answered (inbound call). after i send the answer to the inbound, i do an originate to &park (outbound). after i answered the outbound, i want both sides to be able to talk to each other. but when i try uuid_bridge it does not work. i do not even get a channel_bridge or something. when i do 2 originates and send them to &park(), i can bridge them with each other with uuid_bridge. what am i doing wrong? the other question is, how to unbridge two calls, if they are bridged? i can't find anything like an unbridge command. thanks dennis93 ___ 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] Custom variable, which will be sent with all events?
hi, is there a way to set a custom variable, which will be sent with all events for a specific uuid? i am using socket outbound and would like to set a custom variable for an inbound call or all orginates, so i can filter this variable. but this variable need to be sent will ALL events, also with dtmf, heartbeat and so on. thanks dennis ___ 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] What happend to variable_* in socket_outbound?
they are still there, at least if you register to myevents or all events. i use the default settings from fs and get plenty of them. 2008/10/28 Andy Spitzer <[EMAIL PROTECTED]>: > Woof! > > I used to get lots of variable_* lines when using socket_outbound. They > have disappeared. Is there something I need to configure to get them back? > > --Woof! > > ___ > 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] Socket inbound or outbound with PHP?
Yes, that worked Great, thanks a lot! Now I am able to control the inbound call from the outbound call :-D As far as I understand, this has something to do with changes you made to the myevents mode, right? I think that the myevents mode is better for us, because with events all there are s many information which have to be parsed vom the PHP script. One more question about myevents: Is there a way to configure, which events I get with myevents? In the moment I am missing 2 very important events: 1.) "ringing": If I make an originate to 123, the target-phone itself is ringing, but I do not get to know, that it is ringing. I hoped for something like: Event-Name:CHANEL_EXECUTE, Application:ringing, Call-Direction:outbound (or Answer-State:ringing). For example this is important, if I want to let it ring for x seconds and then do something. 2.) "connected" or "call_accepted" (or something like this): If I make an originate, the target-phone is ringing, but I do not get to know, when the call is answered by pressing the green button. For example I want to wait till the originated caller lifts the phone (is ready to talk) and then play a soundfile or make an uuid_bridge. If I am registered to events all, I a lot of information (to much), but not with myevents. Any idea what I could do? Thanks again for your great help Dennis 2008/10/26 Anthony Minessale <[EMAIL PROTECTED]>: > try latest trunk, i think i can fix you issue by allowing sendmsg to work on > outside uuid even on myevents mode. ___ 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] Socket inbound or outbound with PHP?
I tried &park several times and it does not work for me. The problem
must be on my side and I will test it tomorrow morning again.
I know that I get back the uuid of the originated call and the
targeted phone is ringing, although it is not shown anywhere, that it
is ringing somewhere.
I get "+OK uuid". Then I remove the "+OK " and then try to work with it.
The problem is the same as always: playback does not with uuid -
neither in one nor in the other direction. It simply does not matter,
if I send sendmsg with or without the uuid - the uuid doesn't seem to
matter.
2008/10/26 Anthony Minessale <[EMAIL PROTECTED]>:
> one suggestion i was trying to make was.
> instead of originating the next call to 123 to put it in it's own script.
> try replacing 123 with &park()
>
> {origination_caller_id_number=1234567,inbound_uuid=".$event['Unique-ID']."}sofia/internal/[EMAIL
> PROTECTED]
> &park()
>
> This will put the new call into park where it will be waiting happily for
> commands you can send from the original script
> so you would not need 2, the reply from that originate will give you the
> uuid of the new call if it's successful and you can
> then send it messages just like the other one.
>
> you can have a one to many relationship with event socket to channels it can
> control with 1 socket.
>
>
> On Sun, Oct 26, 2008 at 1:37 PM, Dennis <[EMAIL PROTECTED]> wrote:
>>
>> This is what I have in my "/dialplan/default.xml":
>>
>>
>>
>>> data="hangup_after_bridge=false"/>
>>> data="enable_heartbeat_events=10"/>
>>
>>
>>
>>
>> At the "192.168.1.5:8022" the PHP script is listening. The scripts
>> forks every new call over the socket.
>> So if I have one caller and then call the command "api originate
>>
>> {origination_caller_id_number=1234567,inbound_uuid=".$event['Unique-ID']."}sofia/internal/[EMAIL
>> PROTECTED]
>> 123 " a new PHP process is startet.
>> But this should be no problem, if it would be possible to control the
>> inbound call in the first process from the outbound call in the second
>> process (when using event_socket outbound). "api uuid_bridge" works
>> for example very good, if i call it from the outbound call process.
>> But "api uuid_playfile" does not exist and "sendmsg playfile
>> /path/file.wav" does not work, because it
>> only seems to be available when using mod_event_socket in inbound
>> mode. For me it is not only "playfile" which I am missing, it is also
>> "hangup" and other commands, which seem to work in mod_socket_event in
>> inbound mode.
>>
>> You said, you wanted to lift these restrictions, but they do not work
>> for me (at least I do not know how to use it). Are there any settings
>> I have to configure to make it work the way I need it?
>>
>> Be sure I will be on IRC tomorrow. Perhaps this helps to avoid
>> missunderstandings ;-)
>>
>> Thanks
>> Dennis
>>
>>
>> 2008/10/26 Anthony Minessale <[EMAIL PROTECTED]>:
>> > question what args do you pass to the socket app when you call it in
>> > your dialplan.
>> >
>> > are you using the "full and async" keywords? the full one is necessary
>> > to
>> > have the power to control
>> > other calls. Could this be your prob?
>>
>> ___
>> Freeswitch-users mailing list
>> Freeswitch-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:[EMAIL PROTECTED]
> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[EMAIL PROTECTED]
> iax:[EMAIL PROTECTED]/888
> googletalk:[EMAIL PROTECTED]
> pstn:213-799-1400
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
___
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] Socket inbound or outbound with PHP?
This is what I have in my "/dialplan/default.xml":
At the "192.168.1.5:8022" the PHP script is listening. The scripts
forks every new call over the socket.
So if I have one caller and then call the command "api originate
{origination_caller_id_number=1234567,inbound_uuid=".$event['Unique-ID']."}sofia/internal/[EMAIL
PROTECTED]
123 " a new PHP process is startet.
But this should be no problem, if it would be possible to control the
inbound call in the first process from the outbound call in the second
process (when using event_socket outbound). "api uuid_bridge" works
for example very good, if i call it from the outbound call process.
But "api uuid_playfile" does not exist and "sendmsg playfile
/path/file.wav" does not work, because it
only seems to be available when using mod_event_socket in inbound
mode. For me it is not only "playfile" which I am missing, it is also
"hangup" and other commands, which seem to work in mod_socket_event in
inbound mode.
You said, you wanted to lift these restrictions, but they do not work
for me (at least I do not know how to use it). Are there any settings
I have to configure to make it work the way I need it?
Be sure I will be on IRC tomorrow. Perhaps this helps to avoid
missunderstandings ;-)
Thanks
Dennis
2008/10/26 Anthony Minessale <[EMAIL PROTECTED]>:
> question what args do you pass to the socket app when you call it in
> your dialplan.
>
> are you using the "full and async" keywords? the full one is necessary to
> have the power to control
> other calls. Could this be your prob?
___
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] Socket inbound or outbound with PHP?
Anybody else, who has an idea, what I could do (till I find someone in the IRC)? ___ 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] Socket inbound or outbound with PHP?
2008/10/23 Michael Collins <[EMAIL PROTECTED]>: > What is your IRC nick? Mine is mercutioviz. I'm interested in this issue > because I've been dialing in some somewhat similar scenarios and I might > be able to help, at least a little bit. My nick is Dennis93. Looking forward to it. ___ 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
