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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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max simultaneous calls in freeswitch using sip
by bawajan 20 Jan '14

20 Jan '14

Dear Team, I am using freeswitch and want to make 1500 simultaneous calls using mod_sofia. Hence, could you please let me know. Is it possible to do it and also suggest required hardware capacity to handle this. Thanks in advance, Bawajan -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/max-simultaneous-calls-in-fre… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Voicemail prompt customizations
by Siri MM 20 Jan '14

20 Jan '14

Hi, Is it feasible for me to add additional voicemail promtps to the current ones? For example, if voicemail is disabled for a user, but the user tries to access his voicemail box, the prompt just says "Good bye" and hangs up - is there a way I could add some additional recording in this scenario, such as "Voicemail is not enabled, good bye" ? Thanks!

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FTDM+DAHDI+OSLEC problem
by Nandy Dagondon 19 Jan '14

19 Jan '14

Hello everyone, I installed this on CentOS 6.5 x_64. It looks like the DADHI+OSLEC drivers are properly installed but I still have echo problem. When I do "service dahdi status" w/ no FXO call, a sample line displays: 1 FXO FXSKS (In use) (SWEC: OSLEC) When a call is made, the line remains the same. However, on Elastix, (by rebooting on another disk), the display is: 1 FXO FXSKS (In use) (SWEC: OSLEC) (EC: OSLEC) It seems OSLEC doesn't kick-in during calls. I have tried recompiling in this order: dahdi+oslec patch then FS I was baffling because it was working on FS 1.2.15. It's not working on 1.2.17. Unless there's changes in mod_freetdm. I appreciate for advices where to attack this problem. /Nandy

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NDLB-connectile-dysfunction and sending SMS
by Steve Kieu 18 Jan '14

18 Jan '14

Hello everybody, To fix the problem with linphone (wrong contact IP in teh sip packet that makes FS send ACK to local IP address and then linphone drop call after 30secs or so) I added <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> to my directory/default.xml But that changes stop sending SMS from working! when sending SMS (via sip MESSAGE) the sms is not delivered. FS does not say anything in fs_cli - if I load mod_sms and then it print out log saying it pasing the chatplan correctly and ends up with action send() but the actual sip message never gets into the client machine Is it know problem? Is ther any work around that? I am using FreeSWITCH Version 1.0.head on OpenWRT many thanks, regards, -- Steve Kieu

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Join us today for the Friday Bug Hunt!!
by Ken Rice 17 Jan '14

17 Jan '14

Hey Guys, Join us today on 888 (sip:888@conference.freeswitch.org) for the Friday Bug Hunt (formerly The Friday Free For All) Whats the Bug Hunt? Join us live to get help with your Jira Tickets and help us quash tickets that are outstanding!! Time 2PM EST until ? For other ways to join the call see http://fs0.us/call888 Join us today! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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