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Multi Tenant Installation / problem with call transfer (att_xfer)
by Thorsten Göllner 29 May '17

29 May '17

Hi, I am using Freeswitch 1.6.14 and set it up with a multi tenant installation. Some users are in company-a and some users are in company-b. Both companies have their own dialplan. Works fine. Now I want to make an attanded transfer (att_xfer). I make a call from device 1 to device 2. Then I want to initiate a tranfer. So I press *3 followed by the wanted target (say *312 to call 12). Yes, I bound the meta app key in the dialplan. But now I get the following message/error in the log: EXECUTE sofia/internal/12(a)my.domain.com att_xfer(user/14@my_public_ip) The transfer fails. The user 12 is registered in domain my.domain.com but the wanted user cannot be reached because att_xfer tries to reach the user in the "domain" set in $${domain} in vars.xml. In my vars.xml you can find: <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/> And in my features.xml you can find: <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="user/${digits}@$${domain}"/> </condition> </extension> So att_xfer tries to call the user in $${domain}. But it is set to my public ip. I followd the instruction on https://wiki.freeswitch.org/wiki/Multiple_Companies and in the last chapter wou will find the hint to set the domain to any other domain but not to one of the productive domains. So my IP should be fine. I also tried to set the domain variable before in the dialplan via <action application="set" data="domain_name=my.domain.com"/> <action application="set" data="domain=my.domain.com"/> but this does not take any effect. Do I missing something? Can you give me a hint, what the correct way is to make it? Thanks in advance, Thorsten

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Running conference from api and xml-dialplan
by Artur Mega 29 May '17

29 May '17

Hello community, I have a problem with conferences. When I start a conference from xml dialplan like this - everythiing works (plays audio, I'm asked to enter PIN): ``` <extension name="conf-test"> <condition field="destination_number" expression="^(\d+)$"> <action application="conference" data="arthur@default-en +1234+flags{endconf|moderator}"/> </condition> </extension> ``` But when I use fs_cli to call to my mobile number, then joining to conference - I don't hear audio (woman's voice "You are only one person in the conference"), and what is more important - I'm not asked to enter PIN code. Whyy???? ``` originate {origination_caller_id_number=8080000000}sofia/gateway/mygateway/my-mobile-number &conference(test@default-en+1234+flags{endconf|moderator}) ``` I need to use API to join a mobile phone number to conference, and ask him to enter a PIN number (becouse sometimes phone number can be offline, but owner can turn on a voicemail service, and voicemail will record conversation untill it became full (up to 1 hour)). I don't want to call to voicemail for 1 hour becouse of money. Thanks. -- С уважением, Артур Regards, Arthur

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Maintenance at the FreeSWITCH DataCenter
by Ken Rice 27 May '17

27 May '17

Hey FreeSWITCHers, over the extended holiday weekend, we'll be doing some maint at the DC you may see some services down as we finish realigning things to fully utilize the new hardware we installed last weekend. You can can help us with the data center expenses by hitting the donate button on the homepage @ https://freeswitch.org Thanks Ken Sent from my iPhone

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Freeswitch Packet Fragmentation issue
by Charles Bujold 27 May '17

27 May '17

Upgraded to the latest Freeswitch and now experiencing an issue with packet sizes being bigger than the 1500 allowed. The sip INVITE is 1577 bytes. Tried compress-header but it is not removing enough bytes. At the end of the Invoice there is the X-FS-Support and Remote-Party-ID which I tried to remove but the best I can do is shrink it. Also, the Contact section seems to duplicate the same information, is there a way to cut that down? How can I reduce the fragmentation?

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return to shell only when call from originate is complete?
by Bipin Patel 26 May '17

26 May '17

hi, i was wondering if its possible to to release handle from an originate command only once the call is completed, i mean i send a originate command from shell using fs_cli but only return back to prompt once the call is completed rather than as soon as a bridge is created or a UUID generated -- Regards, Bipin ------------------------------------------------------------------------

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Proper syntax of sched_hangup from webapi
by Rahul MathuR 26 May '17

26 May '17

Hi guys, I am trying to implement a callback functionality using mod_xml_rpc, my URL looks like - http://192.168.2.54:8080/webapi/originate?{origination_caller_id_name=88888… But FS sends out INVITE to 3333333333, receives 183 and then even before it could send out an INVITE to 4444444444, it feels it has reached to the last statement of execution and hence sends CANCEL and drops the call. When I put sched_hangup at the last, it just doesn't play any part in call duration control. I feel that somewhere I am making some mistake but I need another set of eyes to locate it. Could you kindly help me resolve this issue. -- Warm Regds. MathuR

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Error Deleting from sip_dialogs
by Ira Tessler 25 May '17

25 May '17

Hi, I saw this error in my freeswitch.log. Anyone know why this would occur or how to fix it? 2017-05-23 13:03:47.553338 [ERR] switch_odbc.c:522 ERR: [delete from sip_dialogs where uuid='3fc23dbe-4762-4d58-9a49-156d99001fef'] [STATE: HY000 CODE 1213 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.23-log]Deadlock found when trying to get lock; try restarting transaction ] 2017-05-23 13:03:47.553338 [ERR] switch_core_sqldb.c:587 ODBC SQL ERR [STATE: HY000 CODE 1213 ERROR: [MySQL][ODBC 5.1 Driver][mysqld-5.6.23-log]Deadlock found when trying to get lock; try restarting transaction ] delete from sip_dialogs where uuid='3fc23dbe-4762-4d58-9a49-156d99001fef' Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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configure playback outbound http proxy
by Anonim Stefan 25 May '17

25 May '17

Hi, Does Freeswitch provide a parameter to configure an outbound http proxy, when playback a http stream? What I am trying to avoid here is to add specific routes for each IP I am playing back from. Thank you, Stefan

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Freeswitch PRI support
by Deepika Yadav 25 May '17

25 May '17

Hi, I am using Freeswitch in an application that initiates conference calls amongst 20 users in cellular network (mobile phones in GSM network). Currently, for the VOIP-GSM gateway, we are using the service from a company called Doorvaani. But, since, the cost of call estimates to be high and we cannot debug the call drops; we are thinking to buy our own PRI card. I am seeking recommendation on following points: 1. Which card should I buy that is most easily configurable with the Freeswitch i.e. company and type. 2. Reference on how should I start to make Freeswitch configure with the PRI card and start sending and receiving calls. For the current gateway service in use, I simply put the authentication credentials for the corresponding VOIP line offered by the company Doorvaani in the external SIP Profile. In case of PRI, what all changes are needed? Regards, Deepika -- Regards Deepika https://www.iiitd.edu.in/~deepikay/

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timer issue
by Sergey Safarov 24 May '17

24 May '17

I found strange timestamp in FreeSwitch logs https://pastebin.freeswitch.org/view/2933090b message received at send 513 bytes to udp/[31.24.110.61]:5060 at 11:34: 20.608499: system time but call processing have time stamp 12:14:13.279724 Please help me understand why this take place. How it can be fixed? Used kazoo config <https://github.com/2600hz/kazoo-configs/tree/master/freeswitch>

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