January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sending BYE due to ACK timeout
by Jason Sia 20 Jan '14

20 Jan '14

Hi, My freeswitch is sending out a BYE signal due to ACK timeout even though there is already audio . The audio gets cut off after around 30 seconds, I think it detects RTP timeout. It's wierd because I already picked up the call and I can already do conversation but after 30 seconds it just cuts it off. Is there any solution for this? Im using freeswitch 1.5 btw. Thanks, Jason

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Freeswitch as outbound proxy server
by Sam 20 Jan '14

20 Jan '14

Hi Guys, I would like to know if we can use freeswitch as outbound proxy or not. If so then, can run script on outbound proxy fs server and validate and reject calls? If we reject the call from outbound proxy fs server then actual proxy/gateway call will automatically reject? Please let me know. Thanks in Advance. ----- Thanks, Samir -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-as-outbound-proxy-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Global Intercept
by Edenilson Perin 20 Jan '14

20 Jan '14

Even after that I answer an extension, if someone use the global intercept, he join the call. How can I make to intercept only when the extension is ringing? Regards,

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add dialstring to deflect
by Sayyed Mohammad Emami Razavi 20 Jan '14

20 Jan '14

Hello, How to set variables at deflecting?! any solution my technical friends?

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Dynamic Xml (extension includes and gateway bridging)
by Yisroel M. Olewski 20 Jan '14

20 Jan '14

Hello everybody I have a .net application that connects to a FreeSWITCH instance and handles incoming calls I do not have easy access to the xml files I have 2 issues which I think might be similar 1. Parking: I need to preconfigure all incoming calls to park right away, otherwise the esl commands are ignored (as http://wiki.freeswitch.org/wiki/Mod_event_socket#Inbound ). How can I do this via code without needing to change the files physically? I haven't found any documentation for anything like this 2. Bridging: I need to be able to dynamically bridge to gateways listed in an external database, so it wouldn't be practical to keep updating the gateway files. Is it possible to bridge via esl to an arbitrary gateway with me providing host and credentials etc? Sincerely, Yisroel M. Olewski CEO +1 (718) 514-2721 +972 (52) 764-9971 Websites . CRM . ERP . Telephony . Inventory . Content Management

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call_timeout (range)
by Mike Kane 20 Jan '14

20 Jan '14

Quick question. What are the lower and upper boundaries for setting "call_timeout"?

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FS does not set response headers in channel variables
by Sayyed Mohammad Emami Razavi 20 Jan '14

20 Jan '14

I call an extension in the trunk, pass some headers to it to validate some transactions. Trunk responses my invite with "SIP/2.0 480 Temporarily Unavailable" (not user response) with some headers like below: recv 906 bytes from udp/[dst.dst.dst.dst]:8090 at 05:02:33.558626: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable From: ""<sip:TCAPI_User@dst.dst.dst.dst:8090>;tag=8SN0g1DmHy0NF To: <sip:91001@dst.dst.dst.dst :8090>;tag=59f1478-33001dac-1f9a-50022-1f428f-b29b56e-1f428f Call-ID: 96685221-f5e9-1231-b1a5-000c29ae8315 CSeq: 54415412 INVITE Via: SIP/2.0/UDP src.src.src.src:5080;rport=5080;branch=z9hG4bKtScUQtUy9rgmp Supported: replaces Info: {"MerCode":"102"} Sign: Ley1/8cOufqkgd7JuGR7f+jiVxSRM= BillPaymentList: {"paymentID":"16596500682","result":"0"} Content-Length: 0 I run /event plain all in my fs console and no variable exists like info, sign and billpayment in channel variables. Any solution to catch response header?! I cannot define or change key of the headers. I should adapt to these keys!!!

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dialplan is not run on the refer!
by Sayyed Mohammad Emami Razavi 20 Jan '14

20 Jan '14

Hello my friends, Have you any solution for this question?! This is my problem too! I transfer my user to the trunk_2 and when trunk_2 transfers user's call to me with REFER dialplan is not run on this refer. I just read unbridged event after these lines! Some lines read: recv 740 bytes from udp/[21.21.21.21]:8090 at 13:12:05.487698: ------------------------------------------------------------------------ REFER sip:gw+trunk_2@31.31.31.31:5080;transport=udp;gw=trunk_2 SIP/2.0 From: <sip:10*7@21.21.21.21:8090>;tag=59f12d0-33001dac-1f9a-50022-2a4164-6d86d248-2a4164 To: "Outbound Call"<sip:2001@31.31.31.31>;tag=Uca68tZ82mD4c Call-ID: 46945c1e-fc77-1231-f18a-000c29ae8315 CSeq: 1 REFER Via: SIP/2.0/UDP 172.29.0.51:8090;branch=z9hG4bK-2a416d-a50f9461-58324ac0 Refer-To: <sip:10*2*4*0@31.31.31.31:5080> Referred-By: <sip:10*7@21.21.21.21:8090> Max-Forwards: 70 Supported: replaces Contact: <sip:10*7@21.21.21.21:8090> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY Allow-Events: refer Content-Length: 0 ------------------------------------------------------------------------ tport.c:3004 tport_deliver() tport_deliver(0x7f69600043f0): msg 0x7f69600144f0 (740 bytes) from udp/21.21.21.21:5080/sip next=(nil) nta.c:2803 agent_recv_request() nta: received REFER sip:gw+trunk_2@31.31.31.31:5080;transport=udp;gw=trunk_2 SIP/2.0 (CSeq 1) nta.c:3090 agent_check_request_via() nta: Via check: received=21.21.21.21 nta.c:3161 agent_aliases() nta: canonizing sip:gw+trunk_2@31.31.31.31:5080 with contact nta.c:2979 agent_recv_request() nta: REFER (1) going to existing leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f69001622f0): adding notify usage with event refer nua_stack.c:271 nua_stack_event() nua(0x7f69001622f0): event i_refer 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2014-01-20 16:42:05.486392 [DEBUG] switch_core_session.c:998 Send signal sofia/sipinterface_3/10*7 [BREAK] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7f69001622f0): sent signal r_respond 2014-01-20 16:42:05.506399 [DEBUG] sofia.c:6717 Process REFER to [10*2*4*0(a)31.31.31.31] nua_stack.c:573 nua_stack_signal() nua(0x7f69001622f0): recv signal r_respond 202 Accepted nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:405 soa_set_params() soa_set_params(static::0x7f696001d2d0, ...) called nua.c:716 nua_notify() nua: nua_notify: entering nua_stack.c:529 nua_signal() nua(0x7f69001622f0): sent signal r_notify tport.c:3238 tport_tsend() tport_tsend(0x7f69600043f0) tpn = UDP/21.21.21.21:8090 2014-01-20 16:42:05.506399 [DEBUG] switch_ivr.c:1815 (sofia/sipinterface_1/sip:2001@5.238.239.45:5060) State Change CS_EXECUTE -> CS_ROUTING 2014-01-20 16:42:05.506399 [DEBUG] switch_core_session.c:1333 Send signal sofia/sipinterface_1/sip:2001@5.238.239.45:5060 [BREAK] 2014-01-20 16:42:05.506399 [DEBUG] switch_core_session.c:853 Send signal sofia/sipinterface_1/sip:2001@5.238.239.45:5060 [BREAK] 2014-01-20 16:42:05.506399 [NOTICE] switch_ivr.c:1822 Transfer sofia/sipinterface_1/sip:2001@5.238.239.45:5060 to XML[10*2*4*0@context_1] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering tport.c:4026 tport_resolve() tport_resolve addrinfo = 21.21.21.21:8090 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x7f69600043f0): not found by name UDP/21.21.21.21:8090 tport.c:3574 tport_vsend() tport_vsend(0x7f69600043f0): 807 bytes of 807 to udp/21.21.21.21:8090 tport.c:3472 tport_send_msg() tport_vsend returned 807 send 807 bytes to udp/[21.21.21.21]:8090 at 13:12:05.506932: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.29.0.51:8090;branch=z9hG4bK-2a416d-a50f9461-58324ac0;received=21.21.21.21 From: <sip:10*7@21.21.21.21:8090>;tag=59f12d0-33001dac-1f9a-50022-2a4164-6d86d248-2a4164 To: "Outbound Call"<sip:2001@31.31.31.31>;tag=Uca68tZ82mD4c Call-ID: 46945c1e-fc77-1231-f18a-000c29ae8315 CSeq: 1 REFER Contact: <sip:gw+trunk_2@31.31.31.31:5080;transport=udp;gw=trunk_2> Expires: 60 User-Agent: Configured by 2600hz! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ---------- Forwarded message ---------- From: Sayyed Mohammad Emami Razavi <emamirazavi(a)gmail.com> Date: Mon, 20 Jan 2014 19:11:36 +0330 Subject: Process REFER and not Hangup the channel To: freeswitch-users <freeswitch-users(a)lists.freeswitch.org> Hi There, I use FS as proxy to handle a call with REFER to. My question is when get a REFER request, how to do the Transfer regarding the REFER TO before hangup the current channel. My mean before send back the NOTIFY, do the INVITE (based on REFER TO), when the INVITE is 200 OK, then send back NOTIFY. my logs: 11:24:14.141700 [DEBUG] sofia.c:6616 Process REFER to [11004(a)111.22.33.1] 11:24:14.141700 [DEBUG] switch_ivr.c:1773 (sofia/external/6175552222(a)12.13.14.15) State Change CS_EXECUTE -> CS_ROUTING 11:24:14.141700 [DEBUG] switch_core_session.c:1287 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.141700 [DEBUG] switch_core_session.c:830 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.141700 [NOTICE] switch_ivr.c:1779 Transfer sofia/external/6175552222(a)12.13.14.15 to XML[11004@public] 11:24:14.161701 [DEBUG] switch_ivr_play_say.c:1682 done playing file local_stream://moh 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD DONE [sofia/external/+18550000000@1.2.3.4:5060] 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:622 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.161701 [DEBUG] switch_channel.c:2979 (sofia/external/+18550000000@1.2.3.4:5060) Callstate Change HELD -> HANGUP the Transfer to XML[11004@public] not DONE yet, the (sofia/external/+18550000000@1.2.3.4:5060) already Hangup, How to delay the Hangup after Transfer is done?

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Process REFER and not Hangup the channel
by Sayyed Mohammad Emami Razavi 20 Jan '14

20 Jan '14

Hi There, I use FS as proxy to handle a call with REFER to. My question is when get a REFER request, how to do the Transfer regarding the REFER TO before hangup the current channel. My mean before send back the NOTIFY, do the INVITE (based on REFER TO), when the INVITE is 200 OK, then send back NOTIFY. my logs: 11:24:14.141700 [DEBUG] sofia.c:6616 Process REFER to [11004(a)111.22.33.1] 11:24:14.141700 [DEBUG] switch_ivr.c:1773 (sofia/external/6175552222(a)12.13.14.15) State Change CS_EXECUTE -> CS_ROUTING 11:24:14.141700 [DEBUG] switch_core_session.c:1287 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.141700 [DEBUG] switch_core_session.c:830 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.141700 [NOTICE] switch_ivr.c:1779 Transfer sofia/external/6175552222(a)12.13.14.15 to XML[11004@public] 11:24:14.161701 [DEBUG] switch_ivr_play_say.c:1682 done playing file local_stream://moh 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD DONE [sofia/external/+18550000000@1.2.3.4:5060] 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:622 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.161701 [DEBUG] switch_channel.c:2979 (sofia/external/+18550000000@1.2.3.4:5060) Callstate Change HELD -> HANGUP the Transfer to XML[11004@public] not DONE yet, the (sofia/external/+18550000000@1.2.3.4:5060) already Hangup, How to delay the Hangup after Transfer is done?

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Diff between the transfer and bridge application?
by Fredrik Åhlén 20 Jan '14

20 Jan '14

Hi, I am playing around with my demo installation of FreeSwitch and have a question regarding dialplans and dialplan applications. I’ve looked in the wiki but I am not sure of the practical differens between the transfer and bridge application? For example, you can “route” an inbound call to extension 1001 with both these configs: 1. <action application="transfer" data="1001 XML default"/> 2. <action application="bridge" data="user/1001"/> Is the only diff that the transfer application executes the dialplan for the new extension but not in the bride case? With Best Regards, Fred

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