November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Behavior of deflect
by Andy Spitzer 22 Nov '08

22 Nov '08

Woof! When using "deflect" on an inbound answered call, I notice that the FS channel stays connected as long as the original call exists, and does not send a BYE to the original call even after receiving a NOTFIY with a final response fragment. In addition, while FS gets those NOTIFY's, I haven't found a way to report those as events on the socket. The result is, after a deflect, either the controlling application has to wait for the original phone to hangup (which could be a long time, blocking a channel), or it has to hangup immediately, which seems to confuse some devices as the REFER and implied subscription are still outstanding. Is there a way to get FS to report the results of the REFER to the socket, and/or have it automatically send BYE once the final results are in? Thanks. --Woof!

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Nested conditions not allowed.
by henkoegema 21 Nov '08

21 Nov '08

How do I program following: I call extension 1234. Only if the time is between 9 am and 6 pm then play the file were-sorry.wav I tried: <extension name="tempdisconnected"> <condition field="destination_number" expression="^1234"> <condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$"> <action application="playback" data=".../.../sounds/were-sorry.wav"/> </condition> </condition> </extension> but that doesn't work, because nested conditions are not allowed. How do I solve a problem like this. -- View this message in context: http://www.nabble.com/Nested-conditions-not-allowed.-tp20624343p20624343.ht… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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mod_fax feedback
by Gonzalo Servat 21 Nov '08

21 Nov '08

Hi, Nice work on mod_fax! I just tried sending myself a couple of faxes and they were received successfully. I even tried using tone_detect which correctly detected the fax machine and fired off rxfax. I wanted to report back on a couple of issues: 1) Even though the spool dir is set to /tmp/, I called up rxfax with data="rxfax.tiff" and it saved it in /. Shouldn't it save it in the spool dir? 2) After receiving the fax, the channel is never hanged up. The tone_detect app even detects the busy tone (which calls the hangup application when it detects it) but it just sits there for a long time until I manually uuid_kill it. Log says: 2008-11-17 11:59:08 [DEBUG] switch_core_session.c:489 switch_core_session_perform_receive_message() Send signal OpenZAP/1:1/1 [BREAK] 2008-11-17 12:00:20 [DEBUG] mod_fax.c:163 phase_e_handler() ============================================================================== 2008-11-17 12:00:20 [DEBUG] mod_fax.c:170 phase_e_handler() Fax successfully received. 2008-11-17 12:00:20 [DEBUG] mod_fax.c:181 phase_e_handler() Remote station id: 2008-11-17 12:00:20 [DEBUG] mod_fax.c:182 phase_e_handler() Local station id: Home 2008-11-17 12:00:20 [DEBUG] mod_fax.c:183 phase_e_handler() Pages transferred: 1 2008-11-17 12:00:20 [DEBUG] mod_fax.c:184 phase_e_handler() Total fax pages: 1 2008-11-17 12:00:20 [DEBUG] mod_fax.c:185 phase_e_handler() Image resolution: 8031x3850 2008-11-17 12:00:20 [DEBUG] mod_fax.c:186 phase_e_handler() Transfer Rate: 9600 2008-11-17 12:00:20 [DEBUG] mod_fax.c:188 phase_e_handler() ECM status off 2008-11-17 12:00:20 [DEBUG] mod_fax.c:189 phase_e_handler() remote country: 2008-11-17 12:00:20 [DEBUG] mod_fax.c:190 phase_e_handler() remote vendor: 2008-11-17 12:00:20 [DEBUG] mod_fax.c:191 phase_e_handler() remote model: 2008-11-17 12:00:20 [DEBUG] mod_fax.c:193 phase_e_handler() ============================================================================== 2008-11-17 12:00:23 [DEBUG] switch_ivr_async.c:1228 tone_detect_callback() TONE busy DETECTED 2008-11-17 12:00:23 [DEBUG] switch_core_session.c:667 switch_core_session_queue_private_event() Send signal OpenZAP/1:1/1 [BREAK] ... time goes by ... show channels returns: FreeSWITCH>show channels uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate da211ab0-b4af-11dd-9b26-492b0027ea62,2008-11-17 11:58:46,1226930326,OpenZAP/1:1/1,CS_EXECUTE,OpenZAP,52174195,,inbound_fax,rxfax,rxfax.tiff,XML,internal,PCMU,8000,PCMU,8000 1 total. ... until I finally call: FreeSWITCH>uuid_kill da211ab0-b4af-11dd-9b26-492b0027ea62 +OK In the dialplan, right after calling rxfax I call the hangup application. Thanks to all involved for making FS, mod_fax, spandsp, etc! :) Any help on the above would be greatly appreciated. - Gonzalo

2 4

segfault on IPv6
by kokoska rokoska 21 Nov '08

21 Nov '08

Hi all, yesterday I start to experiment with FreeSWITCH and IPV6 and first what I saw is segfault :-) 1. I'm using FreeSWITCH 1.0.trunk (10479) runnig one of Sofia profiles on IPv6 (sip and rtp) on Centos 5.2, 32 bit. 2. As client phone I try Sip Communicator on WinXP, 32 bit. (BTW: Do you know some other soft phone supporting IPv6?) 3. When soft phone send REGISTER (mupltiply times, I really don't know why :-), FS responds 100, 401 and on second run also 200 OK. And as soon as 200 OK is sent, FreeSWITCH dies with segmentation fault. Could someone, please, tell me, what should I do to properly report this issue? Or, may be, I am missing something. Any help is very welcome :-) Best regards, kokoska.rokoska

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Supress Unregister at external gateway
by Peter P GMX 21 Nov '08

21 Nov '08

I have the following scenario for a 2-server FS system with failover functionality: * I have a number of Gateways registered at FS1 * I have another number of Gateways registered at FS2 * In case I want to do maintenance on FS1 I would like all external gateways to be registered on FS2 and then shutdown FS1 * This is not of a problem if I have only a handful of gateways, but in our case they're about 10.000 * The goal is to minimze downtime, when I migrate e.g. 5000 gateways from FS1 to FS2 * 1st method: The normal way would be to register (via xml-rpc or socket) the new Gateway on FS2 and then deregister it on FS1. FS1 sends an unregister (register with expires=0) to the gateway though (after the register from FS2), so this Gateway unfortunately no longer sends calls to any of the 2 FS. * 2nd method: The other way is to deregister them first on FS1 and then register them all on FS2. Because deregistering and registering takes time with ~5000 Gateways (+ we have to ensure that the new register has to take place after unregister), this results in a significant downtime. So my question: Is there any way to suppress deregister messages to the gateway? Best regards Peter

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playback_terminator for phrase
by Woody Dickson 21 Nov '08

21 Nov '08

Hi, I have my dialplan set as follows: <action application="set" data="playback_terminators=#" /> <action application="phrase" data="vm_count,4:new:9:old" /> But the playback terminators does not work when the phrase is being played. Is the the wrong way of specifying the terminator key for phrase? Thanks, Woody

2 1

Unable to pass arguments to phrase for playAndGetDigits
by Juan Backson 21 Nov '08

21 Nov '08

Hi, I tried to use the Phrase API for playAndGetDigits, but it only works if the phrase does not take any input. When an input is specified, it gives error: 2008-11-21 22:42:55 [ERR] switch_ivr_play_say.c:201 switch_ivr_phrase_macro() Can't find macro ivr_prompt,welcome. 2008-11-21 22:42:55 [WARNING] switch_ivr_play_say.c:360 switch_ivr_phrase_macro() Macro [ivr_prompt,welcome] did not match any patterns Here is the lua script: playAndGetDigits(1 5 3 10000 # phrase:ivr_prompt,welcome:menu /mnt/app/prompt/wav/unvailable.wav 9|8|7|6|5|4|3|2|1|100|805|102|103|506|128) Is this something wrong with my lua script? Thanks, JB

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Why does a SIP forked dial select just the first 183?
by Iñaki Baz Castillo 20 Nov '08

20 Nov '08

Hi, I read in: http://wiki.freeswitch.org/wiki/Dialplan_Recipes the following: ----------------------- = Forked dial example = Forked dial is when you want to attempt to ring 2 destinations at the same time. Freeswitch will attempt to call both bridge options simultaneously. The first bridge leg that replies with a 183 (session progress) message will win the call, and the other bridge leg is dropped. This only will work with late_negotiation enabled. ----------------------- While the above could be valid when sending calls to varios (just two?) PSTN gateways, I wonder why I couldn't send a call to two local users (200 and 201) and wait until one of them *answers*. My expected behaviour is: - I call to FS extension 500. - FS does a bridge to 200 and 201. - 200 replies 180 "Ringing" - 201 replies 183 "Session Progress" with early media (not very common in a SIP phone but possible anyway) - After a while 200 replies a 200 OK so FS sends a CANCEL to 201 and replies 200 OK to the caller. Is this common behaviour not possible in FS? is there othe way to get it instead of using "bridge" app with two destinations in "data"? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Video broken with SRTP
by Artem Makhutov 20 Nov '08

20 Nov '08

Hello, I have tested video with FreeSWITCH and found out that video with SRTP is broken. Without SRTP the video works fine. I have attached a capture of the SIP SDP negotiation. I am pretty new to FreeSWITCH so I don't know where exacly the problem is. It looks like FreeSWITCH does not understand m=video 2166 RTP/SAVP 125 126 and trims the additional parameters for it. You can see this in the last line of the log. Can somebody please check it? Thank You, Artem Makhutov INVITE sip:1001@mydomain.org SIP/2.0 Via: SIP/2.0/UDP 192.168.10.224:36188;branch=z9hG4bK-d8754z-370a5b6ed6093a0b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:1000@192.168.10.224:36188> To: "1001"<sip:1001@mydomain.org> From: "1000"<sip:1000@mydomain.org>;tag=81370820 Call-ID: MWJjZjUyNWE0ZTAxNzk2MjQ1MmJhOGQ1MzVkN2U4Njc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1000",realm="mydomain.org",nonce="2e358562-1b1c-48db-b5ac-30184bf50bdc",uri="sip:1001@mydomain.org",response="a1a8d0af96473701e29164539053acfc",cnonce="79768bcc6cba39eb3ea53dca7efbc844",nc=00000001,qop=auth,algorithm=MD5 User-Agent: eyeBeam release 1101l Content-Length: 977 v=0 o=- 9 2 IN IP4 192.168.10.224 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.10.224 t=0 0 m=audio 24832 RTP/SAVP 8 101 a=alt:1 1 : uAe2P6iR j3nStAHd 192.168.10.224 24832 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:dEf9n0TRvJa69HPNvBztLXiIpzQZBXLEVqAWrasz a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0lqyUnCN1VCD139FjAyWfMpcx818SFR7gV8FHEGF a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:2CE064FD329D4469BE4B7B207D91F854 m=video 2166 RTP/SAVP 125 126 a=alt:1 1 : nUHelaPs CFWlR1JP 192.168.10.224 2166 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:/eFXXGOOsbZDjZ8kdd0+Xe05yD0+9zMmqMVVU1ep a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:LaFls9X+TF/0Qj2P4DqwjccTcB+tTssk+nwe+S67 a=fmtp:125 profile-level-id=42e00a; max-br=888; max-mbps=11880 a=fmtp:126 profile-level-id=42e01e; packetization-mode=1; max-br=888; max-mbps=11880 a=rtpmap:125 H264/90000 a=rtpmap:126 H264/90000 a=sendrecv a=x-rtp-session-id:0C4E8C6261F341F8A0F5F2DADBDD16E6 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.224:36188;branch=z9hG4bK-d8754z-370a5b6ed6093a0b-1---d8754z-;rport=36188 From: "1000" <sip:1000@mydomain.org>;tag=81370820 To: "1001" <sip:1001@mydomain.org> Call-ID: MWJjZjUyNWE0ZTAxNzk2MjQ1MmJhOGQ1MzVkN2U4Njc. CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Content-Length: 0 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.224:36188;branch=z9hG4bK-d8754z-370a5b6ed6093a0b-1---d8754z-;rport=36188 From: "1000" <sip:1000@mydomain.org>;tag=81370820 To: "1001" <sip:1001@mydomain.org>;tag=4Qp6tZyg66X4e Call-ID: MWJjZjUyNWE0ZTAxNzk2MjQ1MmJhOGQ1MzVkN2U4Njc. CSeq: 2 INVITE Contact: <sip:mod_sofia@192.168.10.2:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.224:36188;branch=z9hG4bK-d8754z-370a5b6ed6093a0b-1---d8754z-;rport=36188 From: "1000" <sip:1000@mydomain.org>;tag=81370820 To: "1001" <sip:1001@mydomain.org>;tag=4Qp6tZyg66X4e Call-ID: MWJjZjUyNWE0ZTAxNzk2MjQ1MmJhOGQ1MzVkN2U4Njc. CSeq: 2 INVITE Contact: <sip:mod_sofia@192.168.10.2:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 403 v=0 o=FreeSWITCH 4152559439128132736 353175461369499874 IN IP6 2001:6f8:115f:1:213:8fff:fee9:e263 s=FreeSWITCH c=IN IP4 192.168.10.2 t=0 0 a=sendrecv m=audio 29416 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:SfGg1pL13i5Ak878yCV/6awj282NgAifEqPoswar m=video 0 RTP/SAVP 19

3 4

using curl to load switch.conf
by Juan Backson 20 Nov '08

20 Nov '08

Hi, I tried using curl to configure switch.conf to set the min rtp port and max rtp port. The request did arrive in webserver and the response is correctly returned. However, Freeswitch still did not use the port in that specified range. So, I manually modified the switch.conf under autload_configs and then it works. Does anyone know what could be the problem? Thanks, JB

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