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Event Socket not displaying all events
by Raimundo Pérez Nieves 15 Jan '18

15 Jan '18

Hi, I can’t receive all events, like ringtone, exchange media, answered, etc, from sofia/external/550075372035172(a)64.34.75.23 What I am missing? Thanks iam using this dial plan: <extension name="outbound event socket"> <condition field="destination_number" data="^(5004)$">\ <action application="socket" data="127.0.0.1:8040 async"/> </condition> </extension> And this is my perl script: require ESL; use IO::Socket::INET; my $ip = "127.0.0.1"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8040', Proto => 'tcp', Listen => 1, Reuse => 1 ); my $dial_string = 'sofia/external/550075372035172(a)64.34.75.23'; my $codec_string = 'G729'; die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { print "New child pid $pid created...\n"; close($new_sock); next; } my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); $con->sendRecv("event plain ALL"); $con->execute("export","nolocal:absolute_codec_string=$codec_string"); $con->execute("bridge",$dial_string); while($con->connected()) { my $e = $con->recvEvent(); if ($e) { my $name = $e->getHeader("event-name"); print "EVENT [$name]\n"; } } print "BYE\n"; close($new_sock); }

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False Answer Supervision Detection in Freeswitch
by Raimundo Pérez Nieves 15 Jan '18

15 Jan '18

Hi guys, Anybody knows how deals with FAS detection in Freeswitch? Thanks

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Frequent sqlite database corruption
by Jeremy Lainé 15 Jan '18

15 Jan '18

Hi all, Just as Geoff Mina recently reported, I too have encountered numerous instances of sqlite database corruptions, without any power cuts or disk failures. I have experienced this both on EC2 instances and on physical servers. I saw António Silva asked whether Geoff Mina was using wss endpoints, could you elaborate on why this might be the root cause? In the setups where I encountered the corruption, I am indeed using wss endpoints. How about you Geoff? As a workaround I am considering migrating to postgresql (core + sofia) but it's frustating not to understand the root cause of the problem. Cheers, Jeremy

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Lose Race instead of missed call
by Peter Steinbach 15 Jan '18

15 Jan '18

Hello, we have some Problems with the right signalling of missed calls when calling multiple phones in parallel Here's the scenario: Phone no 49 is calling a group with 2170 and 3275 with the following dialstring <action application="bridge" data="{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59eee5e0-c0bc-0135-7dc9-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/2170@flex.mydomain.de:_:{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59f07af0-c0bc-0135-7dca-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/3275@flex.mydomain.de" /> Destination dialstrings are seperated by ":_:" ("Enterprise Origination"). We use curly brackets instead of "<" as we sometimes have to insert asserted identy tags into the dialstring. We checked 3 versions of Freeswitch for this * Version Feb 2016 shows missed calls on both phones. Even if one phone answers, the other phone one still shows a missed call (reason for upgrading to newer Freeswitch) * Version Aug 2017 never shows missed call, see logs and hangup message below * Version 10/Dec 2017(yesterday) never shows missed call, as above So for the 2 never Freeswitch Versions, here are the logs at hangup 2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel sofia/internal/49@flex.mydomain.de:5060 entering state [terminated][487] 2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup sofia/internal/*49*@flex.mydomain.de:5060 [CS_EXECUTE] *[ORIGINATOR_CANCEL]* 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/*2170*@94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA] *[LOSE_RACE]* 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/2170@94.xx.xxx.xx:42170) Running State Change CS_HANGUP (Cur 3 Tot 22) 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/2170@94.xx.xxx.xx:42170) Callstate Change RINGING -> HANGUP 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/2170@94.xx.xxx.xx:42170) State HANGUP 2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel sofia/internal/2170@94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup sofia/internal/*3275*@94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA] [*LOSE_RACE*] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot create outgoing channel of type [user] cause: [LOSE_RACE] 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate Resulted in Error Cause: 502 [LOSE_RACE] Here is the Cancel message for one of the called phones: U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170 *CANCEL sip:2170*@94.xx.xxx.xx:42170 SIP/2.0. Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp. Max-Forwards: 70. From: "Test" <sip:49@flex.mydomain.de>;tag=mQS1eFDpp1peS. To: <sip:2170@94.xx.xxx.xx:42170>. Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5. CSeq: 116168167 CANCEL. *Reason: SIP;cause=200;text="Call completed elsewhere"*. Content-Length: 0. Any hints why this happens, or anyone has this scenario working? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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No response to registration requests with high port numbers?
by Alex Lake 15 Jan '18

15 Jan '18

Hi, Got a problem with a handset that doesn't seem to be registering on our Freeswitch server. The diagnostics from a member of Cisco support said... "At packet 422 we see the phone "0_4147057716(a)192.168.40.26" send a registration request to 5.9.36.113:5060 from 175.159.77.82:32060 - We do not see a reply to this from the SIP provider. We responses to requests from 0_359240637(a)192.168.40.26 <mailto:0_359240637@192.168.40.26> with source port 5060, which suggests to me that the SIP provider is not replying to the high UDP port requests. As the MX is a NAT device, subsequent SIP requests will always have a high UDP port." I'm not sure where we might configure acceptable ranges of SIP source or if it's something else. Thanks for any suggestions... Alex

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Problem With TLS + Contact Header - Bug?
by Geoff Mina 15 Jan '18

15 Jan '18

Greetings, I have a pretty odd issue that I have no idea how to resolve. We have a Sofia profile that is TLS only. It listens on 5061. This is with an upstream vendor and we want them to be able to send inbound calls to extensions without register. Our FS instance is sending back "transport=udp" and port 5060 in the Contact header of the 200 OK. Even though the invite is for TLS and the sip_profile is tls-only=true. I have attached a SIP trace and a copy of our external.xml sofia profile. I am thinking this is a Freeswitch bug - or maybe just an oddly documented "feature". Any help is really appreciated. In the sip-trace attached, you can see an INVITE followed by a Trying and then an OK. The OK has Contact information for transport=udp and port 5060, which is invalid, so the ACK never gets back to us and we retransmit the 200 OK until it expires. Thanks, Geoff

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Fresh i386 install failing to download packages
by LinMin 15 Jan '18

15 Jan '18

Hi all, Just trying to install on Debian 8 32bit and getting the following messages in apt: Package 'freeswitch-all' has no installation candidate Unable to locate package freeswitch-meta-all System is a Via C7 256mb ram 16gb ssd. Other packages are installing fine. Apt is updated. Doing 'apt-cache search freeswitch' shows some packages, but not all. Have I missed something here? Cheers, Francis

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Freeswitch as sip phone and call transfer
by Mehdi Shirazi 14 Jan '18

14 Jan '18

Hi Is it possible to use Freeswitch as sip softphone and use call transfer ? please note this is not same when Freeswitch connected to provider with sip trunk and makes call transfer.when Freeswitch registers in provider as sip phone , when he receives a call ,Transfer calls via provider network and in this case Freeswitch is not in signaling or media path. My goal is dynamically transfer sip phone calls based on time and caller id via provider network.There is no siptrunk just one sip phone account. thanks

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Freeswitch as sip phone and call transfer
by Mehdi Shirazi 14 Jan '18

14 Jan '18

HiIs it possible to use Freeswitch as sip softphone and use call transfer ? please note this is not same when Freeswitch connected to provider with sip trunk and makes call transfer.when Freeswitch registers in provider as sip phone , when he receives a call ,Transfer calls via provider network and in this case Freeswitch is not in signaling or media path.My goal is dynamically transfer sip phone calls based on time and caller id via provider network.There is no siptrunk just one sip phone account. thanks M.Shirazi

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ESL - Event Variables
by Gregor Nanger 14 Jan '18

14 Jan '18

Hi Guys! I am using ESL. On some events like ChannelState there are only few variables sent to ESL client. Is there a way to for some variables, that I created in channel, to force them to be sent on each event? Thank you for advice, Gregor

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