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Sangoma + Atcom Card on the same machine
by Anita Hall 27 Feb '12

27 Feb '12

Hi I am trying to configure Sangoma A104d + Atcom Card (4-port E1) on the same machine. Is it even possible using FreeTDM ? $lspci 01:06.0 Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card 01:07.0 Bridge: PLX Technology, Inc. Device 4002 (rev 01) For Sangoma card, I want to use the more stable libsng_isdn from Sangoma, while for Atcom card I will have to use libpri. Or may be, I should use libpri for both ? regards, Anita

1 1

How to avoid hearing fake ringback tone after bridging?
by dma 27 Feb '12

27 Feb '12

Hi All, The caller is able to hear the real outbound media, including ringback tone, network IVR/announcement, and etc. if the lua code is: session:answer() freeswitch.bridge(session, session2) However, the caller can only hear only fake ringback tone when "session:answer()" is removed. The SIP trace shows "SESSION PROGRESS" is received from the outbound channel. The setting of the call is: bypass_media = true bypass_media_after_bridge = true instant_ringback=false ignore_early_media=false Is there a solution that allow the caller to hear the real outbound media before the call is answered? Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-avoid-hearing-fake-rin… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

IVR Architecture
by Anita Hall 26 Feb '12

26 Feb '12

Hi We are considering writing our own module for processing IVR in C. We are considering using mod_event_socket and then modify it to consume and produce the events internally instead of passing them to the socket. Is this a good idea? I know FreeSWITCH already has a mod_dialplan_xml which can use mod_curl to communicate to the outside world for I/O and DB access. But ... 1) Does XML have all the power flexibility that will be needed for powerful IVR applications? 2) Does mod_dialplan_xml use a separate thread for each IVR execution ? What will be the overhead of interpreting the XML ? 3) Any more ideas ? Thanks :) regards, Anita

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Bug: sleep can't be called inline, contrary to doc
by Harry Coin 26 Feb '12

26 Feb '12

The wiki writes: Applications that are allowed to be run *inline* are: ... * sleep <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep> ... But the log shows: ... Dialplan: loopback/ResCont6507800-b Action sleep(2000) INLINE 2012-02-25 15:29:52.107985 [ERR] mod_dialplan_xml.c:69 This application cannot be executed inline ... -HC

5 8

toll_allow question
by Craig Smith 26 Feb '12

26 Feb '12

Can someone explain how the name="toll_allow" value="domestic,international,local" gets used in the default config? and how to test for it? Thanks.

2 1

failed to find agent when using mod_callcenter
by fieldpeak 26 Feb '12

26 Feb '12

Hi friends, i'm using mod_callcenter, when i call a number of Queue, FS failed to find the agent. while i have register a softphone (nubmer 1001) in the system, also i already assing this agent to this queue in "callcenter.conf.xml", i doubt maybe it is wrong format for "user/1001@localhost" in "callcenter.conf.xml", then i tried user/1001@default. it still did not help, could anybody help me? thanks a lot! *log:* 2012-02-15 23:47:52.491927 [DEBUG] mod_callcenter.c:1036 Updated Agent admin@localhost set state = Receiving 2012-02-15 23:47:52.494305 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables 2012-02-15 23:47:52.494305 [DEBUG] switch_ivr_originate.c:2286 Parsing session specific variables 2012-02-15 23:47:52.494305 [DEBUG] switch_event.c:1170 Parsing variable [call_timeout]=[10] 2012-02-15 23:47:52.494305 [WARNING] mod_dptools.c:3016 Can't find user [1001@default] 2012-02-15 23:47:52.494305 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2012-02-15 23:47:52.494305 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] 2012-02-15 23:47:52.494305 [DEBUG] mod_callcenter.c:1641 Agent admin@localhost Origination Canceled : SUBSCRIBER_ABSENT *reg user info:* Call-ID: e8534b480a48a927@SG9tZS1QQy14NjQ. User: 1001(a)192.168.200.252 Contact: "1001" <sip:1001@x.x.x.x :8420;rinstance=7e0b8cda3d10ac00;fs_nat=yes> Agent: eyeBeam release 3015c stamp 27107 Status: Registered(AUTO-NAT)(unknown) EXP(2012-02-15 23:53:10) EXPSECS(87) Host: DEMO IP: y.y.y.y Port: 8420 Auth-User: 1001 Auth-Realm: x.x.x.x MWI-Account: 1001(a)192.168.200.252 callcenter.conf.xml: <configuration name="callcenter.conf" description="CallCenter"> <settings>  </settings> <queues> <queue name="7_Queue_test@localhost"> <param name="strategy" value="longest-idle-agent"/> <param name="moh-sound" value="$${hold_music}"/> <param name="time-base-score" value="system"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> </queues>    <agents> <agent name="admin@localhost" type="callback" contact="[call_timeout=10]user/1001@localhost" status="Available (On Demand)" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> </agents>  <tiers> <tier agent="admin@localhost" queue="7_Queue_test@localhost" level="1" position="1"/> </tiers> </configuration> -- Regards, Charles

3 4

Woomera
by Cliff Wells 26 Feb '12

26 Feb '12

I've been trying to discover a way to use woomera from FreeSWITCH. The information is sketchy at best. Currently we have a setup like the following: sangoma_mgd <-> woomera <-> asterisk <-> sip <-> FreeSWITCH I'd like to remove Asterisk from the equation, since basically all it does is act as a bridge (and consumes additional resources). I've tried mod_woomera that is available from the Project DiaStar, but it only worked partially (it seems to only support connecting to a single woomera server, and we will have many). I have also read in old (circa 2008) emails that mod_opal would one day lead to a new mod_woomera or perhaps support woomera directly, but haven't found much additional information. Is there any existing solution that I can apply to this problem? Regards, Cliff

4 8

Help with early end to ringback when calling PSTN?
by Harry Coin 26 Feb '12

26 Feb '12

Hello all, great project! In the simple case of dialing out through a small 4 port fxo (grandstream-4104) there is an unwelcome ringback silence gap of almost exactly 10 seconds while the fxo goes offhook and issues its dtmf, then the audio resumes with whatever audio the PSTN sends in response to the dialled number. I've tried to continue the ringback music using ignore_early_media=true with no success. Tried transfer_ringback, instant_ringback, no joy. In the following short fs_cli trace no matter what I try the music stops at the point below noted as 12:19:44... and there is silence until 12:19:54... when the PSTN provides audio. What can I do to keep the ringback music going until the log point noted at ...:54xxxx below? 2012-02-23 12:19:44.000317 [DEBUG] mod_dptools.c:1176 sofia/internal/011(a)gate.quietfountain.com SET [bypass_media]=[true] EXECUTE sofia/internal/011(a)gate.quietfountain.com bridge({ignore_early_media=true}sofia/internal/9935551212@fxoports1to4.quietfountain.com|sofia/internal/9915551212@fxoports1to4.quietfountain.com) 2012-02-23 12:19:44.000317 [DEBUG] switch_ivr_originate.c:1882 Parsing global variables 2012-02-23 12:19:44.000317 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2012-02-23 12:19:44.000317 [NOTICE] switch_channel.c:911 New Channel sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [ab71adf5-4a5e-e111-b2c4-00104b98432d] 2012-02-23 12:19:44.000317 [DEBUG] mod_sofia.c:4496 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State Change CS_NEW -> CS_INIT 2012-02-23 12:19:44.000317 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Running State Change CS_INIT 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State INIT 2012-02-23 12:19:44.021687 [DEBUG] mod_sofia.c:85 sofia/internal/9935551212(a)fxoports1to4.quietfountain.com SOFIA INIT 2012-02-23 12:19:44.021687 [DEBUG] mod_sofia.c:125 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State Change CS_INIT -> CS_ROUTING 2012-02-23 12:19:44.021687 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State INIT going to sleep 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Running State Change CS_ROUTING 2012-02-23 12:19:44.021687 [DEBUG] switch_channel.c:1840 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Callstate Change DOWN -> RINGING 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State ROUTING 2012-02-23 12:19:44.021687 [DEBUG] mod_sofia.c:148 sofia/internal/9935551212(a)fxoports1to4.quietfountain.com SOFIA ROUTING 2012-02-23 12:19:44.021687 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-02-23 12:19:44.021687 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State ROUTING going to sleep 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Running State Change CS_CONSUME_MEDIA 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State CONSUME_MEDIA 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State CONSUME_MEDIA going to sleep 2012-02-23 12:19:44.021687 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] send 1397 bytes to udp/[192.168.29.55]:5060 at 18:19:44.034612: ------------------------------------------------------------------------ INVITE sip:9935551212@fxoports1to4.quietfountain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr Max-Forwards: 69 From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com> Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE Contact: <sip:mod_sofia@192.168.29.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7f17ad7 2011-10-01 14-05-14 +0200 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 445 X-FS-Support: update_display P-Asserted-Identity: "Net Ops Desk L1" <sip:x011@192.168.29.1> v=0 o=011 8004 8000 IN IP4 192.168.29.57 s=SIP Call c=IN IP4 192.168.29.57 t=0 0 m=audio 5012 RTP/AVP 0 8 4 18 112 97 103 102 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:103 AAL2-G726-40/8000 a=rtpmap:102 G729E/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 a=ptime:20 ------------------------------------------------------------------------ 2012-02-23 12:19:44.021687 [DEBUG] sofia.c:5174 Channel sofia/internal/9935551212(a)fxoports1to4.quietfountain.com entering state [calling][0] recv 353 bytes from udp/[192.168.29.55]:5060 at 18:19:44.040757: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com> Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE User-Agent: Grandstream GXW4104 (HW 1.0, Ch:2) 1.3.4.10 Content-Length: 0 ------------------------------------------------------------------------ SNOOZE WITH UNWELCOME SILENCE..... recv 503 bytes from udp/[192.168.29.55]:5060 at 18:19:54.350776: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com>;tag=7be5ae114be4f046 Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE User-Agent: Grandstream GXW4104 (HW 1.0, Ch:2) 1.3.4.10 Contact: <sip:192.168.29.55:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 ------------------------------------------------------------------------ 2012-02-23 12:19:54.332358 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.332358 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [DEBUG] sofia.c:5174 Channel sofia/internal/9935551212(a)fxoports1to4.quietfountain.com entering state [proceeding][180] 2012-02-23 12:19:54.353976 [NOTICE] sofia.c:5253 Ring-Ready sofia/internal/9935551212(a)fxoports1to4.quietfountain.com! 2012-02-23 12:19:54.353976 [NOTICE] mod_sofia.c:2393 Ring-Ready sofia/internal/011(a)gate.quietfountain.com! recv 797 bytes from udp/[192.168.29.55]:5060 at 18:19:54.362180: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com>;tag=7be5ae114be4f046 Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE User-Agent: Grandstream GXW4104 (HW 1.0, Ch:2) 1.3.4.10 Contact: <sip:192.168.29.55:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer, 100rel, path Content-Length: 224 v=0 o=system 8002 8000 IN IP4 192.168.29.55 s=SIP Call c=IN IP4 192.168.29.55 t=0 0 m=audio 5012 RTP/AVP 0 8 4 18 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 ------------------------------------------------------------------------ 2012-02-23 12:19:54.353976 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/011(a)gate.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [NOTICE] switch_ivr_originate.c:481 Ring Ready sofia/internal/011(a)gate.quietfountain.com!

3 3

bypass_media->true state causes sofia bridge under loopback to fail
by Harry Coin 26 Feb '12

26 Feb '12

I'm thinking this is a bug: A local extension calls internal dialplan ring group. If the local extension calls the internal dialplan at 1234 directly, whether bypass_media=true or bypass_media=false it works. However when that same extension is called via bridge loopback/1234 in a ring group it fails with 'normal temporary failure' only when bypass_media=true; it works when bypass_media is false. What am I missing? I'm going to explore the 'bow out' notion next but I don't see that it applies. detail: The ring group lua eventually calls continue_on_fail=true hangup_after_bridge=true bridge loopback/1234 then, internal dialplan at 1234 does bypass_media=true originate_continue_on_timeout=true hangup_after_bridge=false continue_on_fail=true bridge sofia/internal/5551212(a)voip2pstn.com will always fails with 'Normal temporary failure' and then fall through to voicemail. If bypass_media=true -> bypass_media=false both cases work normally. Using FreeSWITCH version: 1.0.head (git-7f17ad7 2011-10-01 14-05-14 +0200) P.S. Thanks for the pointer to 'jira', I filed the previous issue there. I'm new enough I'll wait to file this until I get some sense of whether it's my mistake somehow before filing one about this. -- Harry Coin Quiet Fountain LLC 2118 Lundy Ln Bettendorf, Iowa 52722 USA Office: 563 332 4970x101 Fax: 563 332 9633 Mobile: 563 650 7800

3 2

Exporting variables through mod_fifo/mod_callcenter & codec issue in mod_fifo
by Marcin Gozdalik 25 Feb '12

25 Feb '12

Hi I am trying to export some variables to consumers (mod_fifo) or callback agents (mod_callcenter). The basic scenario I am trying to achieve is to associate some data with path taken in IVR (in form of exported variable) and pass information on which option was chosen in IVR to the agent. I took a look at the source to find out how mod_fifo/mod_callcenter originate calls to consumer/callback agents and it seems the calls are not associated with A-leg (somebody calling to IVR and transferred to queue). Is it on purpose? Is there a workaround for that? I also spotted strange behavior of mod_fifo when originating calls to consumers. It sets absolute_codec_string for calls originated to consumers to variable_sip_use_codec_name from A-leg, adding PCMA/PCMU if not already present. I find the logic there quite puzzling. If somebody is calling to queue using e.g. G.729, mod_fifo will restrict codecs available in leg to consumer to only G.729, PCMA, PCMU. Why is this narrowing of choice of codecs implemented? Best regards -- Marcin Gozdalik

1 1