May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to retreive FreeSWITCH-IPv4 value ? (Julien Terrasson)
by julien terrasson 04 May '13

04 May '13

Hi Michael, I'm actually using a single IP address. But i'm trying to think of an architecture where multiple freeswitch would be in use, each of them acting as user-controled recorders. The reason i wanted to have the Freeswitch-IPv4 from the dialplan was to be able to include it when defining the record name : Ex : <FreeSWITCH-IPv4>-Record_<caller_id>_<sequence_nb>.wav That way i can get the information of where the recording took place. But in the end i used a workaround : i defined a server_id var in the vars.xml.. But anyway, thanks for the prompt feedback ! Have a nice week-end. Julien ------------------------------------------------------------------------------------------------------------------------------------ Julien, "FreeSWITCH-IPv4" is not a channel variable. Rather, it is an event header. Even so, I'm not sure that an individual event is necessarily the best place from which to be extracting this value. What problem are you trying to solve? Also, how many IP addresses do you have on your system? -MC

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Voicemail greeting length?
by Nathan Neulinger 03 May '13

03 May '13

Seeing a problem where recording a voicemail greeting is truncated on freeswitch. The strange thing is - FS acts like it's recording the entire duration I'm speaking (30-40 seconds), but when you play back the recording, it stops after 16-17 seconds fairly consistently. (Same thing if I tell it to save and then go and look at the actual wav file.) There's no indication of anything happening in the logs. I'm only seeing the truncated recording with calls to VM from skinny phones. It does not affect voicemail message recording at all, only greetings. This is with current trunk. Any suggestions on places to start looking? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Alarm, KISS OFF
by Miha 03 May '13

03 May '13

Hi, I have problem regarding alarm device. Alarm guys are saying that I am not detecting kiss off tone or something like that. Is is send on 1400Hz. I have been trying to detect this tone with mod spandsp but with no luck. has some manage to get this working or getting this to work on FS? Thanks! Miha

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 10
by Navnath Sonavne 03 May '13

03 May '13

Hi, As you said create new extensions for flex client and h323 endpoint(Ekiga softphone). I created two new extensions 1103.xml for flex client and 1104.xml for ekiga soft phone. I login using 1103 on flex client and make call to 1104. Now when I call from flex client(1103) to Ekiga softphone(1104),ekiga phone rings and when i answer the call ,call terminates immediately by showing local user rejected call. In another case where I call from Ekiga phone(1104) to flex client(1103),it says person at 1104 is not available and says record your voice mail.In this case flex client did not get any incoming call. Here is my default.xml <extension name="Local_Extension" continue="true"> <condition field="destination_number" expression="^(1104)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/>   <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> <condition field="destination_number" expression="^(1103)$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/>   <action application="set" data="bypass_media=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> Correct me if my dialstrings are wrong. Tell me if any other changes to made in other files also. How can I make both way calling ?Please help me its emergency , I have demo next week. -Navnath. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, 2 May 2013 8:59 PM Subject: FreeSWITCH-users Digest, Vol 83, Issue 10 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. make ekiga to flex call (Navnath Sonavne) 2. Re: make ekiga to flex call (Brian Foster) 3. Re: Check if UA is still there? (mehroz) 4. Re: Check if UA is still there? (Ken Rice) 5. Re: Check if UA is still there? (mehroz) 6. Re: Check if UA is still there? (Michael Collins) Hi All, I am using flex client(at 192.168.9.165) given in freeswitch source to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). I have two users registered on freeswitch with extension as 1100 and 1101 in default context. I loged in using one of extension in flex client,then I dail another extension. After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes in default.xml accordingly to forward calls to this ip. Here is My default.xml part : <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.27"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> Ekiga phone can answer the call and there is audio transmission from both end successfully. But this is one way calling i.e. from flex to ekiga. Now I want to call from Ekiga softphone to flex client. How to call flex client from ekiga softphone? Anybody please me on this issue. -Navnath. According to your dialplan you're forcing all calls that hit Local_Extension (any call with a destination number of 4 digits) to call your ekiga softphone. Your other problem lies in the fact that you should have different dialplans for different methods of calling unlike endpoints one for h232 one for flex. See below: -BDF On May 2, 2013 9:05 AM, "Navnath Sonavne" <navnath.sonavne(a)yahoo.com> wrote: > > Hi All, > > I am using flex client(at 192.168.9.165) given in freeswitch source > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). > I have two users registered on freeswitch with extension as 1100 and 1101 in default context. > I loged in using one of extension in flex client,then I dail another extension. > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes > in default.xml accordingly to forward calls to this ip. > Here is My default.xml part : > > <extension name="Local_Extension"> > <condition field="destination_number" expression="^(\d{4})$"> > <action application="export" data="dialed_extension=$1"/> > <action application="set" data="effective_caller_id_number=${dialed_extension}"/> > <action application="set" data="PEER_IP=192.168.8.27"/> <<Bad: you should never hardcode a specific endpoint IP if this dialplan should be dialing others as well > <action application="set" data="call_timeout=30"/> > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="proxy_media=true"/> > <action application="bridge" data="h323/$1@${PEER_IP}"/> <<Bad: you set your dialplan to bridge the call to your softphone > </condition> > </extension> > > Ekiga phone can answer the call and there is audio transmission from both end successfully. > > But this is one way calling i.e. from flex to ekiga. > Now I want to call from Ekiga softphone to flex client. > How to call flex client from ekiga softphone? > > Anybody please me on this issue. > > -Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > What if we want to even hungup the channels associated with those users? does freeswitch helps ? Like, i have a scenario in which , an active call disrupts , once BOTH users gets disconnected from network. Here, both users gets unregistered after option messages but the call sustains. Is there any thing i can handle this scenario? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp… Sent from the freeswitch-users mailing list archive at Nabble.com. options pings are really just for registered users. a user does not have to be registered to place a call. more useful in your scenario is rtp timers and session timers Ken Sent from my iPad On May 2, 2013, at 8:49, mehroz <mehroz.ashraf85(a)gmail.com> wrote: > What if we want to even hungup the channels associated with those users? does > freeswitch helps ? > > Like, i have a scenario in which , an active call disrupts , once BOTH users > gets disconnected from network. > Here, both users gets unregistered after option messages but the call > sustains. Is there any thing i can handle this scenario? > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org i cannot use session-timers, disabled due to an issue with video calling.... and rtp timers just does not seems to be working! i have But once network disconnects from both clients (works good, with a single client network outage), FS does not hangs up the call ..... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp… Sent from the freeswitch-users mailing list archive at Nabble.com. On Thu, May 2, 2013 at 8:13 AM, mehroz <mehroz.ashraf85(a)gmail.com> wrote: i cannot use session-timers, disabled due to an issue with video calling.... > >and rtp timers just does not seems to be working! > Can you elaborate on this? Do you have a full pcap and FS debug log of a call with RTP in both directions and then the incoming RTP stream stopping and yet the call not being disconnected? >i have > >But once network disconnects from both clients (works good, with a single >client network outage), FS does not hangs up the call ..... > Correct. No SIP signaling and no session timers means there's nothing left to tell FS that the call should be torn down, except for the RTP timers mentioned above. That's why you'll need to supply a complete debug w/ pcap in order to find out why RTP timers are not working. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Blind transfer with call waiting
by Tiago Sampaio 03 May '13

03 May '13

Is there something like blind transfer with call waiting when the target is busy? I have a situation here, EX: When Jhon (user 1001) answer a call and and customer ask to talk to Robert (user 1002), Jhon do a blind tansfer an hangup. Currently im wasting that call when Robert is busy. I need an way to park the call and when Robert become available ring his extension (and maybe put an timeout and call back to Jhon). Im currently writing an script with perl to do this, something like that (i do this from memory): use strict; use POSIX qw(strftime); my ($target_ext,$source_ext,$tryes); my $max_time = 30; our $session; $target_ext = $session->getVariable('target_ext'); $source_ext = $session->getVariable('source_ext'); while($tryes <= $max_time){ my $uuid = $session->execute("db","select/in_a_call/$target_ext"); if(!$uuid) $session->execute("trasfer","$target_ext XML transfer_context"); sleep(1); $tryes++; } $session->execute("transfer","$source_ext XML transfer_context"); Before call an extension i do db insert/in_a_call/<EXTENSION>/${uuid} and after db delete/in_a_call/<EXTENSION> But in thinking will be too hard to maintain this code and must be an easier way to do this... Any help?

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How to retreive FreeSWITCH-IPv4 value ?
by julien terrasson 03 May '13

03 May '13

Hello, I'm trying to dump the Freeswitch-IPv4 value in one of the session. But for some reason the getvar primitive returns an "_undef_" value Should i use a different primitive to retreive that value ? Example : freeswitch@debian> uuid_dump *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* Event-Name: CHANNEL_DATA Core-UUID: 73c3dc70-b3d4-11e2-b885-f9b7dfbfcfcb FreeSWITCH-Hostname: debian FreeSWITCH-Switchname: debian *FreeSWITCH-IPv4: 192.168.1.2* FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2013-05-03%2012%3A28%3A45 .... freeswitch@debian> uuid_getvar *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* * FreeSWITCH-IPv4* *_undef_* Regards, Julien

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Friday Free For All
by Jeff Leung 03 May '13

03 May '13

It's Friday and you know what we do today ;)

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Check if UA is still there?
by Peter Steinbach 03 May '13

03 May '13

Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Compatibilitie between SIP V1.3.0.23 and SIP V2 Sofia Stack
by Deon Vermeulen 03 May '13

03 May '13

Hi All Due to time constraints I'm posting here for the hope for a quick response. I'm currently trying to setup a Trunk to our upstream Carrier. The issues we are facing is that inbound calls from them are working 100%. Outbound calls to them are returned with a 500 error. The only immediate differences we could pickup was the SIP versions. They have a Huawei MSOFTX3000 running SIP version 1.3.0.23. I'm running freeSWITCH 1.5.1b+git~20130414T025052Z~969eae39d9 Is there any possibility someone could help me figure out what compatibility issues between the SIP versions could cause, especially for out bound calls to the Carrier? Thank you very much Kind Regards Deon Vermeulen

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multiple same header in INVITE
by Mickael Hubert 03 May '13

03 May '13

Hi list, I have my freeswitch in SBC "mode" and topology hiding. I'd like to copy some header between A leg and B leg (not to copy all headers). My code is OK, but when I have multiple same headers in my INVITE, freeswitch only copy the last header. Ex: A Leg: Diversion: <sip:0326123456@hexanet.fr>; reason=user-busy; privacy=off Diversion: <sip:0326654321@hexanet.fr>; reason=user-busy; privacy=full ;counter=3 B leg: Diversion: <sip:0326645321@hexanet.fr >;reason=user-busy;privacy=full;counter=3 Only last Diversion header is copied. how can I copy all same header please ? my code: *<condition field="${sip_h_Diversion}" expression="(.*)" break="never">* * <action application="log" data="LOG Modifying Diversion" />* * <action application="export" data="sip_h_Diversion=$1" />* *</condition>* thanks in advance -- Cordialement Hubert Mickaël Ingénieur VOIP - Hexanet --

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