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call fails when cli starts with 00 and is replaced with a +
by Panagiotis Skoulikaritis 06 Jul '16

06 Jul '16

Dear all I'm trying to add a "+" sign in front of the cli before it is send out. If the cli contains already a "+" sign or if the cli starts with 1-9 everything works, but if the cli starts with 00 the call fails with 480. Any idea what might cause this ? Bellow is the dialplan <extension name="test_cli"> <condition field="network_addr" expression="^(^xx\.xx\.xx\.xx|^xx\.xx\.xx\.xx)$"/> <condition field="destination_number" expression="^1234#(\d+)$"> <condition field="caller_id_number" expression="^(00)(\d+)$" break="never"> <action inline="true" application="set" data="effective_caller_id_number=+$2"/> <action inline="true" application="set" data="effective_caller_id_name=+$2"/> </condition> <condition field="caller_id_number" expression="^([1-9]\d+)$" break="never"> <action inline="true" application="set" data="effective_caller_id_number=+$1"/> <action inline="true" application="set" data="effective_caller_id_name=+$1"/> </condition> <action application="set" data="call_direction=outgoing"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="sip_cid_in_1xx=false"/> <action application="set" data="inherit_codec=true"/> <action application="set" data="accountcode=${sip_h_X-ACCOUNTCODE}" inline="true"/> <action application="export" data="nolocal:accountcode=${sip_h_X-ACCOUNTCODE}" inline="true"/> <action application="unset" data="sip_h_X-ACCOUNTCODE" inline="true"/> <action application="set" data="execute_on_answer=sched_hangup +21600 alloted_timeout"/> <action application="bridge" data="sofia/external/+$1@yy.yy.yy.yy:5060"/> <action application="bridge" data="sofia/external/+$1@zz.zz.zz.zz:5060"/> </condition> </extension> the logs I'm getting when the call fails Dialplan: sofia/external/00123456789(a)192.168.157.14 parsing [public->test_cli] continue=false Dialplan: sofia/external/00123456789(a)192.168.157.14 Regex (PASS) [test_cli] network_addr(192.168.160.10) =~ /^(^192\.168\.160\.10|^192\.168\.170\.10)$/ break=on-false |--- Dialplan: Processing recursive conditions level:1 [test_cli_recur_1] require-nested=TRUE |--- Dialplan: sofia/external/00123456789(a)192.168.157.14 Regex (PASS) [test_cli_recur_1] caller_id_number(00123456789) =~ /^(00)(\d+)$/ break=never |--- Dialplan: sofia/external/00123456789(a)192.168.157.14 Action set(effective_caller_id_number=+123456789) INLINE EXECUTE sofia/external/00123456789(a)192.168.157.14 set(effective_caller_id_number=+123456789) 2016-07-05 04:12:06.258273 [DEBUG] mod_dptools.c:1469 sofia/external/00123456789(a)192.168.157.14 SET [effective_caller_id_number]=[+123456789] |--- Dialplan: sofia/external/00123456789(a)192.168.157.14 Action set(effective_caller_id_name=+123456789) INLINE EXECUTE sofia/external/00123456789(a)192.168.157.14 set(effective_caller_id_name=+123456789) 2016-07-05 04:12:06.258273 [DEBUG] mod_dptools.c:1469 sofia/external/00123456789(a)192.168.157.14 SET [effective_caller_id_name]=[+123456789] |--- Dialplan: sofia/external/00123456789(a)192.168.157.14 Regex (FAIL) [test_cli_recur_1] caller_id_number(00123456789) =~ /^([1-9]\d+)$/ break=never 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:216 (sofia/external/00123456789(a)192.168.157.14) State Change CS_ROUTING -> CS_EXECUTE 2016-07-05 04:12:06.258273 [DEBUG] switch_core_session.c:1396 Send signal sofia/external/00123456789(a)192.168.157.14 [BREAK] 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:528 (sofia/external/00123456789(a)192.168.157.14) State ROUTING going to sleep 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:472 (sofia/external/00123456789(a)192.168.157.14) Running State Change CS_EXECUTE 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:535 (sofia/external/00123456789(a)192.168.157.14) State EXECUTE 2016-07-05 04:12:06.258273 [DEBUG] mod_sofia.c:178 sofia/external/00123456789(a)192.168.157.14 SOFIA EXECUTE 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:258 sofia/external/00123456789(a)192.168.157.14 Standard EXECUTE EXECUTE sofia/external/00123456789(a)192.168.157.14 set(outside_call=true) 2016-07-05 04:12:06.258273 [DEBUG] mod_dptools.c:1469 sofia/external/00123456789(a)192.168.157.14 SET [outside_call]=[true] EXECUTE sofia/external/00123456789(a)192.168.157.14 export(RFC2822_DATE=Tue, 05 Jul 2016 04:12:06 +0000) 2016-07-05 04:12:06.258273 [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [RFC2822_DATE]=[Tue, 05 Jul 2016 04:12:06 +0000] 2016-07-05 04:12:06.258273 [NOTICE] switch_core_state_machine.c:315 sofia/external/00123456789(a)192.168.157.14 has executed the last dialplan instruction, hanging up. 2016-07-05 04:12:06.258273 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/00123456789(a)192.168.157.14 [CS_EXECUTE] [NORMAL_CLEARING] 2016-07-05 04:12:06.258273 [DEBUG] switch_channel.c:3242 Send signal sofia/external/00123456789(a)192.168.157.14 [KILL] 2016-07-05 04:12:06.258273 [DEBUG] switch_core_session.c:1396 Send signal sofia/external/00123456789(a)192.168.157.14 [BREAK] 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:535 (sofia/external/00123456789(a)192.168.157.14) State EXECUTE going to sleep 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:472 (sofia/external/00123456789(a)192.168.157.14) Running State Change CS_HANGUP 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:735 (sofia/external/00123456789(a)192.168.157.14) Callstate Change RINGING -> HANGUP 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:737 (sofia/external/00123456789(a)192.168.157.14) State HANGUP 2016-07-05 04:12:06.258273 [DEBUG] mod_sofia.c:413 Channel sofia/external/00123456789(a)192.168.157.14 hanging up, cause: NORMAL_CLEARING 2016-07-05 04:12:06.258273 [DEBUG] mod_sofia.c:549 Responding to INVITE with: 480 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:60 sofia/external/00123456789(a)192.168.157.14 Standard HANGUP, cause: NORMAL_CLEARING 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:737 (sofia/external/00123456789(a)192.168.157.14) State HANGUP going to sleep 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:504 (sofia/external/00123456789(a)192.168.157.14) State Change CS_HANGUP -> CS_REPORTING 2016-07-05 04:12:06.258273 [DEBUG] switch_core_session.c:1396 Send signal sofia/external/00123456789(a)192.168.157.14 [BREAK] 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:472 (sofia/external/00123456789(a)192.168.157.14) Running State Change CS_REPORTING 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:823 (sofia/external/00123456789(a)192.168.157.14) State REPORTING 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:104 sofia/external/00123456789(a)192.168.157.14 Standard REPORTING, cause: NORMAL_CLEARING 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:823 (sofia/external/00123456789(a)192.168.157.14) State REPORTING going to sleep 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:498 (sofia/external/00123456789(a)192.168.157.14) State Change CS_REPORTING -> CS_DESTROY 2016-07-05 04:12:06.258273 [DEBUG] switch_core_session.c:1396 Send signal sofia/external/00123456789(a)192.168.157.14 [BREAK] 2016-07-05 04:12:06.258273 [DEBUG] switch_core_session.c:1623 Session 40729659 (sofia/external/00123456789(a)192.168.157.14) Locked, Waiting on external entities 2016-07-05 04:12:06.258273 [NOTICE] switch_core_session.c:1641 Session 40729659 (sofia/external/00123456789(a)192.168.157.14) Ended 2016-07-05 04:12:06.258273 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/00123456789(a)192.168.157.14 [CS_DESTROY] 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:626 (sofia/external/00123456789(a)192.168.157.14) Running State Change CS_DESTROY 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:636 (sofia/external/00123456789(a)192.168.157.14) State DESTROY 2016-07-05 04:12:06.258273 [DEBUG] mod_sofia.c:323 sofia/external/00123456789(a)192.168.157.14 SOFIA DESTROY 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:111 sofia/external/00123456789(a)192.168.157.14 Standard DESTROY 2016-07-05 04:12:06.258273 [DEBUG] switch_core_state_machine.c:636 (sofia/external/00123456789(a)192.168.157.14) State DESTROY going to sleep 2016-07-05 04:13:01.618279 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/cdr-csv/Master.csv freeswitch@internal>

2 3

Setting variables with mod_hiredis
by Abdul Hakeem 05 Jul '16

05 Jul '16

Hello, Does anyone have list of useful commands for mod_hiredis. I am interested to know if it's possible to set channel variable as in the mod_memcache example below: <action application="set" data="ignore=${memcache(set key ${channel_var})}"/> Cheer, AH

3 4

Change language of voice conference announcements to different language
by Денис 04 Jul '16

04 Jul '16

Hello. We have a trouble with mod_conference. How to change lang? Even with <action application="set" data="default_language=de"/> in log i got [INFO] mod_conference.c:10370 using channel sound prefix: /usr/share/freeswitch/sounds/en/us/callie I see http://lists.freeswitch.org/pipermail/freeswitch-users/2014-October/108544.… and add ivr/sounds.xml from en... nothing changed. Need i clean conference.conf.xml from all profiles? Or change all sounds to phrase:...? Or create many profiles with only different sound_prefix?

1 0

default muted media on conference ?!
by Anonim Stefan 04 Jul '16

04 Jul '16

Hello, I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called. I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports. The problem I have is that I can hear no media. I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything). Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs". Any ideas what might be the problem? Thanks in advance, Stefan [1] https://postimg.org/image/4cwxn91bb/

3 9

FreeSWITCH gets restarted automatically
by Sagar Malam 04 Jul '16

04 Jul '16

Hello, I have been using FreeSWITCH for conference solution.Recently i have been facing freeSWITCH automatic restart issue randomly.Only relavant thing i can find from logs is following : 2016-07-01 16:49:28.172089 [INFO] switch_core.c:2581 Shutting down Number of calls were around 150.FreeSWITCH was consuming around 50 % CPU.So server was not overloaded.Can someone tell me in what cases FreeSWITCH will restart itself ? -- Thanks & Regards, Sagar Malam Sr. Software Engineer | Ecosmob Technologies Pvt. Ltd. (+91)9601533171 | www.ecosmob.com <http://www.google.com/url?q=http%3A%2F%2Fwww.hodusoft.com&sa=D&sntz=1&usg=A…> Skype: sagar.ecosmob

1 0

FreeSWITCH Salesforce integration
by Moshe BT 02 Jul '16

02 Jul '16

Hi It is probably not a freeswitch question, however as a place to start, does anyone have experience in setting up integration of salesforce and freeswitch, what documentation best to follow, advice or offline consultancy for it. Thanks

5 5

Freeswitch scaling.
by Mimiko 01 Jul '16

01 Jul '16

Hello dear community. Currently I run a FS server in production for a company to handle all necessary tasks. But it is nit fail-tolerant. It is only one server. So I would like your hints about scaling. In fact, FS handles about 100 users with its phone numbers. Two voip providers, internet access for remote users and direct sip calling. There are conferences, callcenter, voicemail, ivr. When calling outside destinations, there is a fail-tolerant connection between two providers in case one failed. Both providers have own connection needs for header configuration. For several extensions calls are recorded. Also each user can initiate a call record. The most important is callcenter, which have dynamic agent join and leave. Agents join and leave using there permitted extensions and using a password, so the FS can know who is login in and record this in DB. This is important to allow then accounting for agents the time they where in system for answer. Also each answer is recorded in file and in DB with the id of the agent who answered. Of course, clients hear its position while they wait for an answer. Now I want to extend FS to two physical servers. One DB, and shared storage with drdb. I would like to implement config load from DB. As I've read, the big problem in this case to handle the route to registered user: on which server it is registered. This can be done using DB for sharing these data. Others says that Opensips come in handle, to allow opensips for registration purpose and routing, and FS to server for voicemail, conference, callcenter. My questions is how in this case will be call record handled between two channels? If routing is in Opensips, how would I handle agents join and leave using password in the dialplan, like there is in FS? Do I have to double dialplan for FS and for Opensips? What will be better to use Opensips: as a registrar, or only as a load-balancer? Thank you. -- Mimiko desu.

1 0

Ring back
by Josh Richesin 01 Jul '16

01 Jul '16

hello everyone, I have a small issue here. Sorry for the duplicate post if you are on the fushionpbx IRC yesterday and today I am logged in via IRC app. Any ways, when users call in, they get prompted with a "please hold while I connect your call and then music plays instead of us-ring. I know how to change that on IVR but this is on all inbound routes except for the ring groups or ivrs...Any ideas? Josh

1 0

parameters in the Contact
by royj＠yandex.ru 30 Jun '16

30 Jun '16

Hi We have an UA that sends in REGISTER to FreeSWITCH in Contact header some parameter. And that UA expects that paramter in INVITE RURI FreeSWITCH sends to it. Is there ability to convey parameter in INVITE?

1 1

Verto + 180 Ringing
by Gregor Nanger 29 Jun '16

29 Jun '16

Hi! Can someone plese help me. I have strange behaviour on outbound calls. Provider that terminates calls sends only SIP 180 , not also SIP183. And it looks that verto doesn't forward 180 to client and there is no ringing tone in verto client neither I can capture this event in client. I tried to ring_ready, but verto still doesn't send anything to client (originator). It only sends event on 183. Is this by design? How can I set in dialplan and send client 183? Thank you, Gregor

3 6