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Play Two files one for caller and other for callee
by Romeu Medeiros 21 Apr '14

21 Apr '14

Hello to alls, Im need to play two files, one for caller side and other for callee side, when the call is answered. But I need to play this files simultaneously. Does anyone have an idea how to do this? Thanks in advance R.Medeiros

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Cisco phone registration
by andpe 20 Apr '14

20 Apr '14

Hi I have a problem with the registration of the Cisco 79xx phones. The SIP Message Header (MH) which sends the phone, in the fields "TO" and "FROM", places the IP address instead of the domain name. The configuration of the phone is set to the domain name (xx.yy.com.) Is there any way to set the phone to send the domain name in the field instead of an IP address? Is there a firmware version that sends MH correctly? Andy

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perl missing
by Joli Martinez 19 Apr '14

19 Apr '14

Hello, I am having an issue compiling FreeSwitch with Perl. I am running CentOS: This is the error I get when I try to make freeswitch. I have installed Perl, Perl-devel, mod_perl, mod_perl-devel and I still get the error. What am I missing? making all mod_perl CC mod_perl_la-mod_perl.lo Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted.

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pickup endpoint break the attended transfer and blind transfer
by Lloyd Aloysius 19 Apr '14

19 Apr '14

Hi All After add the pickup in the user dial string. attended transfer and blind transfer stop working. dial-string - {sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}, *pickup/${dialed_user}@${dialed_domain}* *http://pastebin.freeswitch.org/22360 <http://pastebin.freeswitch.org/22360>* I try all the models - Polycom, Cisco, Snom, Aastra If I remove the pickup everything works perfect. Is this is a bug? Please let me know how to solve this issue Thank you Lloyd

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Extension does not ring after some time
by Mr.Gab 18 Apr '14

18 Apr '14

Hello everybody, I'll try to explain my strange problem: I have a Freeswitch in the cloud (set following the wiki for amazon ec2) and I cannot bridge calls to extensions (no ring) some seconds after the registration. When register the ip phone is all ok but after a while this is what i can see on the console if i try to call his extension: http://pastebin.com/YT4Pgi7K i cannot hear any ring from leg a and the phone on the leg b doens't ring and after some seconds the call is hanged up with cause "[CS_CONSUME_MEDIA] [NO_ANSWER]" obtaining this on the fs console. 2014-04-17 23:02:54.013897 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/900 [KILL] 2014-04-17 23:02:54.013897 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/900 [BREAK] 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/900) Running State Change CS_HANGUP 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/900) Callstate Change DOWN -> HANGUP 2014-04-17 23:02:54.013897 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/900) State HANGUP 2014-04-17 23:02:54.013897 [DEBUG] mod_sofia.c:506 Channel sofia/internal/900 hanging up, cause: NO_ANSWER 2014-04-17 23:02:54.013897 [INFO] mod_dptools.c:3201 Originate Failed. Cause: NO_ANSWER If i re-register the phone i can receive call for some seconds and than the problem returns. How can i manage that? Thanks in advice crazygabry

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session, channels, calls
by Anish Patel 18 Apr '14

18 Apr '14

Hello all, i've been reading the wiki, book, and using some google foo in trying to understand what freeswitch considers sessions, calls, and channels. So far i've gathered a session is just a connection between FreeSWITCH and a protocol, but i haven't been able to figure out what a channel is considered and what a call is considered. I know this is probably super simple, but i havent been able to make the connection yet as to what they are. Thanks in advance. Anish

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Re: [Freeswitch-users] OpenSIPS configuration for 2 or more FreeSWITCH installs
by Ali Pey 18 Apr '14

18 Apr '14

You need to have separate group of destinations and dispatch the calls based on where they are coming from: http://www.opensips.org/html/docs/modules/1.5.x/dispatcher.html#id293400 Regards, Ali Pey On Wed, Apr 16, 2014 at 10:39 AM, Marcin M. <m2m(a)poczta.fm> wrote: > Ali , what i want to reach is SIP proxy (5060,5080) with dispatcher and > freeswitch servers in multidomain mode with sip profiles (internal- > 5060,external-5080 ). Now have freeswitch part working , but have no idea > how i can modify opensips configuration to work on two ports,5060,5080 . > > scenario: > 1. From user proxy will recieve sip msg at port 5060 then this msg will > be dispatched to freeswith servers to the same port 5060(internal profile) > 2.From provider proxy will recieve sip msg at port 5080 and i need to > dispatch it to freeswitch servers to port 5080 (external profile) > > Do you know how i can modify opensips configuration from this link ??: > > http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSW…<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> > > Thanks for any help ALI > > W dniu 16.04.2014 15:48, Ali Pey pisze: > > Then you need two entries. Each are a separate identity and sip profile. > There is no such as thing as port range. > > Regards, > Ali Pey > > > On Wed, Apr 16, 2014 at 2:25 AM, Marcin M. <m2m(a)poczta.fm> wrote: > >> yes, but i want to use both ports : 5060 and 5080 >> >> W dniu 15.04.2014 23:23, Ali Pey pisze: >> >> In opensips, Add the port at the end of your destination ip. e.g. >> 10.11.12.13:5070 >> >> Regards, >> Ali Pey >> >> >> >> On Tue, Apr 15, 2014 at 5:23 PM, Marcin M. <m2m(a)poczta.fm> wrote: >> >>> Hi Ali, >>> >>> Iam using dispatcher to load balance . It is the same configuration as >>> in sample link. On freeswitch side iam using multidomain configuration. >>> How i can modify opensips conf to handle sip msg on 5080 port ? >>> >>> >>> W dniu 15.04.2014 19:40, Ali Pey pisze: >>> >>> Hello, >>> >>> I use opensips with other ports than 5060. You just need to configure >>> it properly in opensips. What modules do you use for load balancing? >>> >>> Regards, >>> Ali Pey >>> >>> >>> >>> >>> On Tue, Apr 8, 2014 at 8:36 AM, Marcin M. <m2m(a)poczta.fm> wrote: >>> >>>> Hi , >>>> iam using this configuration >>>> http://wiki.freeswitch.org/<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs#Download_and_compile_OpenSIPS_1.7.1 >>>> for load balancing FreeSWITCH servers, everything works fine if i use >>>> port 5060 ( user -> opensips-> freeswitch ). If i want to use other >>>> port for incoming calls from providers , let say 5080 as is in external >>>> profile in Freeswitch ( provider -> opensips:5080-> freeswitch:5080 ), >>>> how i can do that ?? maybe someone have same situation ?? or maybe i can >>>> configure gateways on internal profile on freeswitch , if that will >>>> work ?? >>>> >>>> Thanks for any info >>>> Marcin >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> freeswitchsolutions.com<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> http://wiki.freeswitch.org<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> http://www.cluecon.com<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists. <FreeSWITCH-users(a)lists.freeswitch.org> >>>> freeswitch.org<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> http://lists.freeswitch.org/<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> freeswitch.org/mailman/<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> options/freeswitch-users >>>> http://www.freeswitch.org<https://contactmonkey.com/api/tracker?cm_session=4d6a68b2-af2d-4bbf-bc8d-da…> >>>> >>> >>> >>> >> >> > >

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Freeswitch not terminating call after receiving "503 Service not available" response.
by Camila Troncoso 18 Apr '14

18 Apr '14

Hi, I have this scenario: Proxy - FS -- LCR I make a call from an user register in my proxy, passing trough freeswitch ( for media transcoding) , to my LCR server. The LCR looks for one route for my call, but I doesn't found any, so It replays "503 Service not available, no more gateways". When FS receives this response , *it sends a re invite* to the LCR server, falling again in the same error reply. In my case I don't what FS to send the re invite , I need it to pass the reply ("503 Service not available, no more gateways") to the proxy. How can I manage this? Regards, Camila *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

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Outbound ESL bridge sync mode
by Grant Bagdasarian 18 Apr '14

18 Apr '14

Hello, When specifying the outbound socket connection to be in sync mode, is it still possible to initiate a bridge command in async mode from the external application? I'm using .NET to create the external call control application, and the following page: https://wiki.freeswitch.org/wiki/Event_Socket_Outbound#Q:_Can_I_bridge_a_ca… tells me I can bridge a call using outbound. The example explains it should be done like below: sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808 event-lock: true How do I create this using the ManagedESL library? Like this? ESLevent ev = new ESLevent("sendmsg", ""); ev.AddHeader("call-command", "execute"); ev.AddHeader("execute-app-name", "bridge"); ev.AddHeader("execute-app-arg", "{ignore_early_media=true}sofia/gateway/myGW/177808}"); ev.AddHeader("event-lock", "true"); eslConnection.sendMSG(ev, "SOME_UUID") Thanks, Grant

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How to check the "fax_result_code" in the script? such as php script?
by sparklezou 18 Apr '14

18 Apr '14

Hi Buddies, Reviewed the wiki, https://wiki.freeswitch.org/wiki/Mod_spandsp How to check the "fax_result_code" in the php script? Also other result info, such as fax_result_text, then could provide more detail response info to the client. Thanks! 2014-03-03 sparklezou

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