- FreeSWITCH-users - FreeSWITCH Mailing Lists

hangup/disconnect on transfer
by Hector Geraldino 31 Oct '11

31 Oct '11

Hi, I'm facing a problem that I'm not even sure that belongs to the FS boundaries but I'm asking anyway: I need to know when the legB of the call transfers the legA to another extension. Here's the scenario: PSTN -> Cisco Unified CM -> Sip Trunk (FreeSwitch) -> ESL application An incoming call is effectively bounded by a Cisco Call Manager platform to a SIP trunk managed by FreeSwitch, and controlled by a Java application. I can answer the call, play prompts and do some more stuff. After that, I'm originating a call to another extension (managed by the CUCM, not freeswitch) to be answered by an agent, and bridging both legs using uuid_bridge. So what I've is a classic call center scenario where I connect incoming calls to agents, and I calculate the call handling time based on the time between the call was bridged to an operator and the time the call was dropped because one of the two legs hanged up. Also after receiving the disconnection notice I perform some cleanup tasks. Great. So far so good. But, there's a problem that begins when the agent (legB) transfers the call to another extension (or even an external phone number) using the transfer function on their IP Phone. As this phone is not 'controlled' by FreeSwitch there's no event registered for this call transfer. So what is happening is that both legs are still connected to FS even when the agent (legB) has transferred the call. This off course is an undesired behavior, because I'm saying that 'agentB spent 30 minutes handling call X' when he actually transferred this call to someone else after maybe just 20 seconds. So what I need is an event or something that helps me to detect when a call has been transferred, not using the 'transfer' tool on freeswitch but transferred outside FreeSwitch. The only notification that FreeSwitch receives after a call transfer is performed is this SIP message (with its response): recv 805 bytes from udp/[192.168.23.104]:5060 at 14:40:04.945886: ------------------------------------------------------------------------ UPDATE sip:mod_sofia@192.168.5.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.23.104:5060;branch=z9hG4bK78836e9096b From: <sip:5562@192.168.23.104>;tag=63092ed1-42e0-4489-9bb9-bd2b77b7fcf3-28767966 To: "" <sip:5512@192.168.5.72>;tag=cFQ5y9jHy0a0p Date: Mon, 31 Oct 2011 14:39:41 GMT Call-ID: e062f678-7e70-122f-f890-000c298202db User-Agent: Cisco-CUCM8.0 Max-Forwards: 70 Supported: timer,resource-priority,replaces Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 UPDATE Contact: <sip:5562@192.168.23.104:5060> Supported: X-cisco-srtp-fallback Supported: Geolocation P-Asserted-Identity: "Leona C" <sip:5655@192.168.23.104> Remote-Party-ID: "Leona C" <sip:5655@192.168.23.104>;party=calling;screen=yes;privacy=off Content-Length: 0 ------------------------------------------------------------------------ send 624 bytes to udp/[192.168.23.104]:5060 at 14:40:04.946088: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.23.104:5060;branch=z9hG4bK78836e9096b From: <sip:5562@192.168.23.104>;tag=63092ed1-42e0-4489-9bb9-bd2b77b7fcf3-28767966 To: "" <sip:5512@192.168.5.72>;tag=cFQ5y9jHy0a0p Call-ID: e062f678-7e70-122f-f890-000c298202db CSeq: 101 UPDATE Contact: <sip:mod_sofia@192.168.5.72:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 1800;refresher=uas Min-SE: 120 Content-Length: 0 ------------------------------------------------------------------------ Is there any way to react to this incoming message and generate an event or action? A channel variable or something that I may be missing? Any thoughts? Thanks for your kindly attention, Hector

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[Freeswitch-users] CHANNEL_CALLSTATE=>switch_channel_perform_set_callstate
by David 31 Oct '11

31 Oct '11

Hi, how can i push variables to the CHANNEL_CALLSTATE=>switch_channel_perform_set_callstate EVENT for a conference outbound channel? Greez David

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No caller id & DTMF detection fail issue
by Vellayappan.N 31 Oct '11

31 Oct '11

Hi, I am using Digium X100P SE single port fxo analog card for making pstn calls. I can able to ring to my SIP number from my mobile/land line number. But I am not getting the caller id number who is actually calling, instead I am getting only "0000000000" always. My service provider is Airtel and I am in India. Freeswitch is detectling only few (two or three) dtmf digits. Can you please help me to get the incoming caller-id number while calling from PSTN number. -- Regards Vellayappan

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b-leg pending
by fieldpeak 30 Oct '11

30 Oct '11

Hi friends, i found when i make a call to pstn from a registed user via FS, after the call hangup (no matter caller hangup or callee hangup, and no matter the callee answer or not), the a-leg always was not released. however, if i use two registed users call for each other, there is no problem. can anyone help advise how to fix it, thanks! below is my dialplan, <extension name="CallToPSTN" > <condition field="destination_number" expression="^(0\d+)$" > <action application="bridge" data="sofia/internal/$ 1(a)213.102.56.206" /> </condition> </extension> -- Regards, Charles

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Re: [Freeswitch-users] transfer_ringback and ivr
by Madovsky 29 Oct '11

29 Oct '11

I found a way to replace transfer_ringback and produce a non blocking tone in case of a-leg answered already (like after an ivr option) <action application="displace_session" data="$${sounds_dir}/extras/ringback.wav m"/> <action application="set" data="send_silence_when_idle=1400"/> if you wish to use sleep before answer or bridge you can use it, slience_when_idle is done to send RTP packets and let the a-leg to continue to hear the tone <action application="sleep" data="2000"/> unfortunately displace_session doesn't have any stop option, so it needs to use uuid_displace <action application="set" data="api_res=${uuid_displace ${uuid} stop $${sounds_dir}/extras/ringback.wav}"/> or use it with execute_on_answer for example if you don't when the b-leg will answer... Hope it helps Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, October 29, 2011 4:41 AM Subject: transfer_ringback and ivr I didnt' succeed to make transfer_ringback work after an ivr and play_and_get digits and before transfer app. must I set something else or is it the wrong variables to create a ringback if a-leg answered already ? Thanks Franck

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transfer_ringback and ivr
by Madovsky 29 Oct '11

29 Oct '11

I didnt' succeed to make transfer_ringback work after an ivr and play_and_get digits and before transfer app. must I set something else or is it the wrong variables to create a ringback if a-leg answered already ? Thanks Franck

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FreeTDM woes
by Trever L. Adams 29 Oct '11

29 Oct '11

I have a span of 4 FXS modules. They are all configured to use context Internal-FXS. Channel 5 will dial. All the others won't even stop sending the dialtone when one attempts to dial. Channel 5 can call channels 1-3 (there is no 4 at the moment, I have tried it where 4 isn't used and 5 becomes 4, same problem), but if 5 hangs up, the called channel continues to ring. zt.conf has echo_cancellation_level at 16. Yet, I constantly get messages like: Disable echo cancel for 1:5 I get messages like this continually: [ERR] ftmod_zt.c:1188 [s1c2][1:2] Failed to process event from channel Finally, with the help of someone on this list I now have my call screening setup to work for the appropriate caller id names and numbers when it has been blocked. The problem is, the screening part completely fails. These are the actions from the file, which are only slightly modified from the example at http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_13:_Call_Screening . It asks for your name, it records. But it doesn't give ANYTHING to the dialed party. <action application="set" data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="phrase" data="voicemail_record_name"/> <action application="playback" data="tone_stream://%(500, 0, 640)"/> <action application="set" data="playback_terminators=#*0123456789"/> <action application="record" data="${call_screen_filename} 7 200 2"/> <action application="set" data="group_confirm_key=1"/> <action application="set" data="fail_on_single_reject=true"/> <action application="set" data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="FreeTDM/1/1:_:FreeTDM/1/2:_:FreeTDM/1/3:_:FreeTDM/1/4:_:FreeTDM/1/5:_:FreeTDM/1/6"/> <action application="voicemail" data="default $${domain} ${dialed_number}"/> <action application="hangup"/> I don't think the problem is the phrase:screen_confirm as I have tried removing it. I do set dialed_number to my phone number at the top of this extension so it is set. It should be noted that all calls to this FXO port are routed as the action above shows. The non-blocked-caller_id route has the exact same bridge line. Am I just plain missing something about the FreeTDM module? Is it this broken? Any help would be greatly appreciated. Trever -- "Karl's version of Parkinson's Law: Work expands to exceed the time alloted it."

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Script for ASR & ACD Monitoring
by Faisal Rehman 28 Oct '11

28 Oct '11

Hi Everyone, I want to write a script that reads the Master.csv & generate an alert if the ASR & ACD is low from the desired value. Anybody has written such script, please help me write it or provide any sample. Thanks Faisal Rehman

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Re: [Freeswitch-users] Feedback
by Michael Giagnocavo 28 Oct '11

28 Oct '11

Wow that looks very impressive. I'd love to see it. Have you considered putting it up on GitHub? That gives you a nice UI, bug tracking, and wiki. -------- Original message -------- Subject: [Freeswitch-users] Feedback From: Chad Vogel <cvogel(a)lyonl.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> CC: Hello, I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users. Here are some of the changes we have made: * We converted FS core to C++ and having it compiling to .NET 4.0 CLI * FS now runs inside its own namespace * We replace FS_CLI with a PowerShell shell app * Supports SQL Server support via the Native SQL Server Client API * Support for windows clustering (up to 32 node active/active cluster) * Ties more directly into the Win32 API and has less reliance on open source Libs * Replaces OpenSSL sockets with Windows encrypted sockets * Added windows performance monitoring * No longer need to use mod_managed for managed modules * Memory management relies on .Net garbage collection * Added support for Microsoft Speech * VoiceXML 3.0 and SCXML support * No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages) * Fixed RTP clock timing issues in virtualized environments * New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 64, Issue 174
by Dip Mehta 28 Oct '11

28 Oct '11

Thanks MC. You got it right. Appreciate your reply. Cheers ! Dip Mehta On Sat, Oct 29, 2011 at 12:07 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Hangup Causes (Michael Collins) > 2. Re: Dialing issues (Michael Collins) > 3. Re: High load on database server (john(a)mobex.biz) > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 28 Oct 2011 11:10:21 -0700 > Subject: Re: [Freeswitch-users] Hangup Causes > Although not easy on a production system, your best bet is to catch the > behavior in the act. Usually this means that you have to rely upon humans to > report the symptoms to you in a timely and accurate manner. > > One way you can help track these down is to use something like pcapsipdump > and do a ring buffer that covers at least several days. Then, when you hear > about a symptom you can go back and look at the corresponding pcap file > which will have both signaling and audio. From there you can hopefully trace > the root of the problem. > > -MC > > http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump > > On Fri, Oct 28, 2011 at 9:19 AM, Jason Moran <jmoran(a)secureachsystems.com>wrote: > >> I have some questions about the types of statuses we are getting (Yes, >> I’m already familiar with http://wiki.freeswitch.org/wiki/Hangup_causes ) >> **** >> >> ** ** >> >> We just deployed FS to production a couple of days ago and here is a >> sample of our Hangup Causes:**** >> >> CALL_REJECTED 1**** >> >> NO_ANSWER 738**** >> >> NO_USER_RESPONSE 175**** >> >> NORMAL_CLEARING 1667 (Expected that this is only Early Hangups)**** >> >> NORMAL_TEMPORARY_FAILURE 4**** >> >> NORMAL_TEMPORARY_FAILURE (Rang) 37 (Rang means this status >> happened more than 5 seconds after the session was started)**** >> >> Complete 224 (Complete is our own status instead of >> NORMAL_CLEARING, meaning WE hung up because we finished the entire message). >> **** >> >> ORIGINATOR_CANCEL (Rang) 3**** >> >> RECOVERY_ON_TIMER_EXPIRE (Rang) 9 **** >> >> UNALLOCATED_NUMBER 2**** >> >> USER_BUSY 35**** >> >> ** ** >> >> My questions:**** >> >> **1. **NORMAL_TEMPORARY_FAILURE happens a lot, and often returns 30 >> or more seconds after starting. Could this be a “silent call” (etc)? We have >> gotten 3-4 complaints of silent calls and I’m hoping to find the cause.** >> ** >> >> **2. **Is there anything here (or ideas elsewhere?) to explain why >> a call will be silent once in a while?**** >> >> **3. **Is this spread of causes normal or does it seem strange**** >> >> **4. **We’ve had a few people note that the message on their >> voicemail is over before they can hear it - even though we play the message >> twice (to make short messages longer). Does anybody know of a solution for >> this?**** >> >> ** ** >> >> Thanks! -Jason**** >> >> ** ** >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 28 Oct 2011 11:15:13 -0700 > Subject: Re: [Freeswitch-users] Dialing issues > This is a newfies configuration issue. I've only looked briefly at the > newfies dialer (I don't want to get anywhere near it because it's AGPL - > yuck) and I do know that it requires you to specify your gateway in some > manner. It does not look like you have done that. I'd review this > information from the newfies-dialer.org documentation site: > > Newfies-Dialer’s Gateway > > Log into the Newfies-Dialer interface using your root username and password > previously created in these instructions. > > Click Admin, and locate and enter the “Dialer Gateways” panel on the > dashboard, then add a dialer gateway. Only the fields in bold need to be > filled in. That is to say, the name of gateway, for identification, and the > Gateway’s field, e.g. > > sofia/gateway/myprovider/ > > “myprovider” is the name of the gateway which you can take from the gateway > name when you typed “sofia status” when configuring the Freeswitch Gateway. > Finally, check the status is active, and save. The trunk should now appear > in the gateway list. > -MC > > On Fri, Oct 28, 2011 at 1:28 AM, Dip Mehta <dipkumar.mehta(a)gmail.com>wrote: > >> Hi, >> >> I am trying to integrate Newfies Dialer with Freeswitch. >> >> This is my sofia status >> >> freeswitch@127.0.0.1@internal> sofia status >> Name Type >> Data State >> >> ================================================================================================= >> *:XX.XX.XX.XX* alias >> internal ALIASED >> external profile sip:mod_sofia@* >> :XX.XX.XX.XX*:5080 RUNNING (0) >> external::example.com gateway >> sip:joeuser@example.com NOREG >> * external::XX.XX.XX.XX gateway sip:password@XX.XX.XX.XX >> REGED* >> internal profile >> sip:mod_sofia@199.167.145.102:5060 RUNNING (0) >> internal-ipv6 profile sip:mod_sofia@[::1]:5060 >> RUNNING (0) >> >> ================================================================================================= >> 3 profiles 1 alias >> >> >> Looks like the Service provider is now registered with the Freeswitch. I >> am asuming as it says REGED in the status. >> >> However, when i start my campaign through the dialer, I am unable to make >> the outgoing calls which results in these errors. >> >> 2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:1884 Parsing >> global variables >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_request_uuid]=[48df3dbe-013e-11e1-8b92-bc305bec2901] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_hangup_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/hangupcall/] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [origination_caller_id_number]=[8888888888] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [bridge_early_media]=[true] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [hangup_after_bridge]=[true] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_app]=[true] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [absolute_codec_string]=[PCMA] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [originate_timeout]=[1800] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_event.c:1521 Parsing variable >> [ignore_early_media]=[true] >> 2011-10-28 04:24:41.992674 [ERR] switch_core_session.c:426 *Could not >> locate channel type* *XX.XX.XX.XX Customers contact number* >> 2011-10-28 04:24:41.992674 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type [*XX.XX.XX.XX Customers contact number*] >> cause: [CHAN_NOT_IMPLEMENTED] >> 2011-10-28 04:24:41.992674 [DEBUG] switch_ivr_originate.c:3348 Originate >> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:1884 Parsing >> global variables >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_request_uuid]=[53a1c460-013e-11e1-8b92-bc305bec2901] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_answer_url]=[http://*XX.XX.XX.XX*:9080/api/dialer_cdr/answercall/] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_hangup_url]=[http://*XX.XX.XX.XX* >> 2:9080/api/dialer_cdr/hangupcall/] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [origination_caller_id_number]=[8888888888] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [bridge_early_media]=[true] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [hangup_after_bridge]=[true] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_app]=[true] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [absolute_codec_string]=[PCMA] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [originate_timeout]=[1800] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_event.c:1521 Parsing variable >> [ignore_early_media]=[true] >> 2011-10-28 04:25:00.052673 [ERR] switch_core_session.c:426 Could not >> locate channel type *XX.XX.XX.XX Customers contact number* >> 2011-10-28 04:25:00.052673 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type [*XX.XX.XX.XX Customers contact number*] >> cause: [CHAN_NOT_IMPLEMENTED] >> 2011-10-28 04:25:00.052673 [DEBUG] switch_ivr_originate.c:3348 Originate >> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] >> >> >> Please advice. >> >> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: <john(a)mobex.biz> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Date: Fri, 28 Oct 2011 11:36:45 -0700 > Subject: Re: [Freeswitch-users] High load on database server > I'm probably in over my head, but i could use help. > > it appears my Freeswitch went down and came back up (possible from volume) > when it did a large volume accoutn is getting bolcked 407 messages: > > Im getting the following message log: > > 2011-10-28 14:24:54.778973 [DEBUG] sofia.c:7099 IP 66.42.61.24 Rejected by > acl "domains". Falling back to Digest auth. > 2011-10-28 14:24:54.778973 [WARNING] sofia_reg.c:1400 SIP auth challenge > (INVITE) on sofia profile 'internal' for [18125362643(a)216.134.222.93] from > ip 66.42.61.24 > > That IP address is set up as 66.42.61.24/32 in the cidr. > > What do i do to get them back up? > > Thanks > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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