April 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

1 0

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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call pickup vs application="intercept"
by Sean Devoy 18 Apr '14

18 Apr '14

Hi all, I have just installed a new site where the receptionist has a CISCO SPA phone and 32 key extended keypad. The buttons on the keypad are used like an operator console for various extensions. They provide Busy Line Feature (BLF or BLA) and Speed Dial (sd). I will update the CISCO SPA section of the WIKI to document how to do this as soon as I get an email with my username and password back! However, I think I need some clarification on "pickup" verses "intercept". To me these two dialplans are exactly equivalent, please correct me if I missed something. Note, I have restricted the pickup "group" to the dialed extension, eliminating the problem of multiple incoming calls when picking up in a "group". Scenario 1: Main dialplan routes call to extension "10" (simplified version): <extension name="2digit_internal" > <condition field="destination_number" expression="^(10)$" > [SNIP ...] <action application="db" data="insert/call_pickup_uuid/${destination_number}/${uuid}"/> <action application="bridge" data="{sip_invite_domain=[REDECTED]}user/${destination_number}@[REDECTED]" /> <action application="answer"/> <action application="voicemail" data="default voicemail_ESTA $1"/> <action application="hangup"/> </condition> </extension> The application "db" line is the key to link to the events. Now when the receptionist sees extension 10 blinking and unanswered, she can press that key to "pickup" that ringing line. (The Cisco phone dials **10) That dialplan is: <extension name="directed-call-pickup"> <condition field="destination_number" expression="^\*\*(\d\d)$"> <action application="answer"/> <action application="intercept" data="${db(select/call_pickup_uuid/$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> The receptionist "intercepts" the call by uuid stored in the above dialplan. Scenario 2: (same, but different dial plan) <extension name="2digit_internal" > <condition field="destination_number" expression="^(10)$" > [SNIP ...] <action application="bridge" data="{sip_invite_domain=[REDECTED]}user/${destination_number}@[REDECTED], pickup/$1" /> <action application="answer"/> <action application="voicemail" data="default voicemail_ESTA $1"/> <action application="hangup"/> </condition> </extension> No application "db" line, instead added ", pickup/10" to the bridge line. Now when the receptionist sees extension 10 blinking and unanswered, she can press that key to "pickup" that ringing line. (The Cisco phone dials **10) That dialplan is: <extension name="directed-call-pickup"> <condition field="destination_number" expression="^\*\*(\d\d)$"> <action application="answer"/> <action application="pickup" data="$1)}"/> <action application="sleep" data="2000"/> </condition> </extension> So, is this just two different ways to skin a cat or are there subtleties that make one a preferred implementation?

1 0

Couldnt open /etc/freeswitch/freeswitch.xml
by Joli Martinez 17 Apr '14

17 Apr '14

Hello, I have a production Freeswitch box and this morning I noticed FS was not working. I restarted it and got this message. As far as I know no one has touched the box, but not 100% sure. Any ideas why I am getting this. I researched and a lot of the hits I got was due to a bad install, but this box has been working correctly for over 6 months. [ERR] switch_xml.c:1385 Couldn't open /etc/freeswitch/freeswitch.xml (No such file or directory) Thanks,

3 3

The Sipml5 cannot register to freeswitch
by rockxsj 17 Apr '14

17 Apr '14

I found that sometimes the sipml5 client can not register to freeswitch.When I use tcpdump to capture the packet I can see the three-way handshake,but I execute show channels on the freeswitch can not see my channel.Then after several minutes,I re-try call,successful. Why?Could you help me?

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bleg_uuid
by mbo 17 Apr '14

17 Apr '14

I noticed that the variable bleg_uuid is only set when a bridged call gets connected, not if the call end with hangup cause NO_ANSWER or BUSY. Is this by intension? I need to linl the both call legs even when the call get’s bot connected, can I set the bleg_uuid by my self on bridge with the uuid of the a_leg or is this not a good idea? Thanks Markus

3 2

if user_busy then continue_on_fail=false else continue_on_fail=true
by Shahzad Bhatti 16 Apr '14

16 Apr '14

hi, i want to know how to impalement in XML_curl dialplan that *if user busy then **set continue_on_fail=false **else * *continue_on_fail=true* it means when freeswitch get response *USER_BUSY then it stop attempt otherwise continue.* here is my sample dialplan: <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="Bridge Call to Destination"> <condition field="destination_number" expression="^004917691444955$"> <action application="set" data="process_cdr=b_only"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="nibble_account=1"/> <action application="set" data="nibble_rate=0"/> <action application="export" data="origination_caller_id_number=004917691444955"/> <action application="export" data="nolocal:nibble_account=1"/> <action application="export" data="cust_ip=186.218.49.95"/> <action application="export" data="nolocal:nibble_rate=0.0229"/> <action application="export" data="nolocal:connection_charges=0"/> <action application="export" data="buy_rate=0.0199"/> <action application="export" data="sell_rate=0.0229"/> <action application="export" data="destination_id=470"/> <action application="export" data="destination_name=Germany Mobile O2"/> <action application="export" data="provider_id=9"/> <action application="export" data="domain_ip=194.152.251.110"/> <action application="export" data="nolocal:execute_on_answer=nibblebill adjust -0" /> <action application="export" data="absolute_codec_string=${ep_codec_string},PCMU,G729"/> <action application="phrase" data="sayrate,0:2:29"/> <action application="bridge" data="{absolute_codec_string=PCMA,G729}sofia/external/ 004917691444955(a)194.152.251.110"/> </condition> </extension> </context> </section> </document> Regards Shahzad Bhatti

2 1

Microsoft Lync Integration?
by Jerry Richards 16 Apr '14

16 Apr '14

Does Freeswitch support Microsoft LYNC integration? Or is anyone planning for that? I saw some mailing list references back in 2011/2012, but there are no references in the Freeswitch wiki. Regards, Jerry

2 1