December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Flowroute is testing CNAM lookup
by Steven Schoch 27 May '14

27 May '14

I am one of the Beta testers of Flowroute's CNAM lookup for inbound calls. So far, it seems to be working quite well. -- Steve

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Compiling Freeswitch for Android?
by GillesToo 25 Apr '14

25 Apr '14

Hello Searching the archives only seems to return information on how to use SIPDroid as an SIP client to connect to a Freeswitch server. Since Android is built on Linux, and Freeswitch has been compiled for compact devices, I was wondering if someone had tried compiling it for Android? I could use a small IVR on a smartphone to handle incoming calls. Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compiling-Freeswitch-for-Andr… Sent from the freeswitch-users mailing list archive at Nabble.com.

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DID providers - any thoughts? (jan 2013)
by Cal Leeming [Simplicity Media Ltd] 22 Apr '14

22 Apr '14

Hello everyone, I've been recently reviewing several DID providers, and looking for some feedback. I appreciate this isn't totally on topic for FreeSWITCH, but over the last few months I've found the community feedback to be invaluable - so I'm hoping it might open up a good discussion. I would be really interested to hear other peoples experiences with their DID providers, as much detail as possible. There are several providers we've looked at, but they are either reselling others services, or do not deliver the service quality we are looking for. The criteria I've been looking for is; * Able to provide either US or UK numbers * Exposes an API that allows control of purchasing and routing * Guarantees premium call quality, good latency, and inbound CLI. * Does NOT operate on a market/exchange basis * Trading as a telecoms provider for more than 5 years * Has a reputable name within the community or industry So far, I have only found the following providers that meet this criteria; * Hello Telecom - UK numbers only, ITSPA registered, no charges for minutes, rental or concurrent channels. No minimum charges. Excellent call quality/latency, no CLI problems, and almost no downtime in the last 6 months. However, they do not offer a sane purchasing API which makes automation difficult, and interop with FS has been somewhat painful. * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but still very fair. Currently at inter-op stage, so can't comment further - but they seem very good so far. * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, difficult to justify unless you already have existing traffic. * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over 9000 pages of inter-op and around £10k in set up fees. Thanks in advance Cal

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Cisco phone registration
by andpe 20 Apr '14

20 Apr '14

Hi I have a problem with the registration of the Cisco 79xx phones. The SIP Message Header (MH) which sends the phone, in the fields "TO" and "FROM", places the IP address instead of the domain name. The configuration of the phone is set to the domain name (xx.yy.com.) Is there any way to set the phone to send the domain name in the field instead of an IP address? Is there a firmware version that sends MH correctly? Andy

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avmd from mod managed
by Malay Thakershi 24 Mar '14

24 Mar '14

Hello, Is there a way to do AVMD from mod_managed? I see JavaScript example technique from http://wiki.freeswitch.org/wiki/Mod_avmd but it seems signature of StreamFile is different in mod_managed. In JavaScript, second argument to StreamFile is a function (OnInput) where detection is done while is mod_managed, it is sample start number. So is AVMD possible from mod_managed? Thank you. Malay

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Cannot Make any call [CS_NEW] [WRONG_CALL_STATE]
by crazygabry 13 Mar '14

13 Mar '14

Hello everybody, i cannot figure out this problem.. i have a multi tenant freeswitch installation. with every domain i can register users but when i try to make any call i get this behaviour: 2013-12-18 00:30:12.654751 [DEBUG] switch_core_session.c:1048 Send signal sofia/internal/1000@MYDOMAIN:9060 [BREAK] 2013-12-18 00:30:12.654751 [DEBUG] switch_core_session.c:1048 Send signal sofia/internal/1000@MYDOMAIN:9060 [BREAK] 2013-12-18 00:30:12.654751 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000@MYDOMAIN:9060) Running State Change CS_NEW 2013-12-18 00:30:12.654751 [DEBUG] switch_core_state_machine.c:486 (sofia/internal/1000@MYDOMAIN:9060) State NEW 2013-12-18 00:30:12.654751 [DEBUG] switch_core_session.c:1048 Send signal sofia/internal/1000@MYDOMAIN:9060 [BREAK] 2013-12-18 00:30:12.654751 [DEBUG] sofia.c:1845 detaching session 8f2d7628-677b-11e3-af8c-a115136ede7e 2013-12-18 00:30:22.714696 [WARNING] switch_core_state_machine.c:567 8f2d7628-677b-11e3-af8c-a115136ede7e sofia/internal/1000@MYDOMAIN:9060 Abandoned 2013-12-18 00:30:22.714696 [NOTICE] switch_core_state_machine.c:570 Hangup sofia/internal/1000@MYDOMAIN:9060 [CS_NEW] [WRONG_CALL_STATE] 2013-12-18 00:30:22.714696 [DEBUG] switch_channel.c:3211 Send signal sofia/internal/1000@MYDOMAIN:9060 [KILL] 2013-12-18 00:30:22.714696 [DEBUG] switch_core_session.c:1383 Send signal sofia/internal/1000@MYDOMAIN:9060 [BREAK] 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000@MYDOMAIN:9060) Running State Change CS_HANGUP 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:731 (sofia/internal/1000@tMYDOMAIN:9060) State HANGUP 2013-12-18 00:30:22.714696 [DEBUG] mod_sofia.c:413 Channel sofia/internal/1000@MYDOMAIN:9060 hanging up, cause: WRONG_CALL_STATE 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:58 sofia/internal/1000@MYDOMAIN:9060 Standard HANGUP, cause: WRONG_CALL_STATE 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:731 (sofia/internal/1000@MYDOMAIN:9060) State HANGUP going to sleep 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:744 (sofia/internal/1000@MYDOMAIN:9060) Callstate Change DOWN -> HANGUP 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1000@MYDOMAIN:9060) State Change CS_HANGUP -> CS_REPORTING 2013-12-18 00:30:22.714696 [DEBUG] switch_core_session.c:1383 Send signal sofia/internal/1000@MYDOMAIN:9060 [BREAK] 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1000@MYDOMAIN:9060) Running State Change CS_REPORTING 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:816 (sofia/internal/1000@MYDOMAIN:9060) State REPORTING 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:102 sofia/internal/1000@MYDOMAIN:9060 Standard REPORTING, cause: WRONG_CALL_STATE 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:816 (sofia/internal/1000@MYDOMAIN:9060) State REPORTING going to sleep 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:493 (sofia/internal/1000@MYDOMAIN:9060) State Change CS_REPORTING -> CS_DESTROY 2013-12-18 00:30:22.714696 [DEBUG] switch_core_session.c:1383 Send signal sofia/internal/1000@MYDOMAIN:9060 [BREAK] 2013-12-18 00:30:22.714696 [DEBUG] switch_core_session.c:1591 Session 32 (sofia/internal/1000@MYDOMAIN:9060) Locked, Waiting on external entities 2013-12-18 00:30:22.714696 [NOTICE] switch_core_session.c:1609 Session 32 (sofia/internal/1000@MYDOMAIN:9060) Ended 2013-12-18 00:30:22.714696 [NOTICE] switch_core_session.c:1613 Close Channel sofia/internal/1000@MYDOMAIN:9060 [CS_DESTROY] 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:618 (sofia/internal/1000@MYDOMAIN:9060) Callstate Change HANGUP -> DOWN 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:621 (sofia/internal/1000@MYDOMAIN:9060) Running State Change CS_DESTROY 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/1000@MYDOMAIN:9060) State DESTROY 2013-12-18 00:30:22.714696 [DEBUG] mod_sofia.c:323 sofia/internal/1000@MYDOMAIN:9060 SOFIA DESTROY 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:109 sofia/internal/1000@MYDOMAIN:9060 Standard DESTROY 2013-12-18 00:30:22.714696 [DEBUG] switch_core_state_machine.c:631 (sofia/internal/1000@MYDOMAIN:9060) State DESTROY going to sleep i reloaded mod_sofia and xml. Can you also tell me if i have to reload sofia or just reload xml in order to add a domain? Thanks in advice -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Cannot-Make-any-call-CS-NEW-W… Sent from the freeswitch-users mailing list archive at Nabble.com.

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i've problem with TLS
by dwi yulianto 07 Mar '14

07 Mar '14

i've folowed tutorial from http://wiki.freeswitch.org/wiki/SIP_TLS but when im use phonerlite as softphone i've problem when enable TLS in phonerlite, and when i see in FS_CLI from my freeswitch server i got this error tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0xb67215c8): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0xb67215c8): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) anyone know the solution from that error? thanks.

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On-conference distortion for FreeTDM channels, but not SIP channels
by Melanie Treitinger 03 Mar '14

03 Mar '14

Nobody has an idea on this? Here again Bob's Message from 2013-09-20: I've tried everything I can think of on this, so now I'm reaching out to you good people. I have a FreeSWITCH system that has the latest version of FreeTDM, Dahdi, libpri, and a Rhino R4T1 quad T1 board with up-to-date firmware. When I call into one of the PRI channels, a "playback"-played file sounds clean, but when I go into conference, the conference audio is distorted. This does not happen with a SIP channel. I would classify the distortion as a "crackle", but it is very regularly periodic. Repeating ticks at (roughly) 10-20 Hz repeat rate, I would say. I can only hear them when audio is present, i.e. the audio seems to be modulated with the ticks. And that conference audio can be the MOH guitar recording. I've eliminated the following: - T1 clocking - Echo Cancellers (I've turned on and off all combinations of hardware and software EC) - T1 board firmware (it's the latest, and the problem only happens on-conference). - Packet length - it's the same (20 mS) for the conference engine, FreeTDM, and Dahdi - mu-law vs. a-law encoding - behaves identically. - Comfort noise - happens with and without. - conference sampling rate (tried 8000 and 16000, even though I'm using TDM, which is 8000). - FreeSWITCH version - happens on 1.2 stable, the latest master branch, and an older version that we were using. If you haven't heard of anything like this, could you please give me your thoughts on why this could happen only on-conference, not while playing WAV files, and only with FreeTDM channels? Thanks! Bob

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You guys are terrible
by Jose Suero 28 Feb '14

28 Feb '14

I'm swimming in a freeswitch log looking for a problem when all of the sudden this comes up: 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's commin' for you... Didn't find the freeswitch problem, but now have a machete masked fellow on his way :-) Made me smile

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FreeSwitch Proxy + RTPProxy Media server
by adahary 09 Feb '14

09 Feb '14

I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. <action application="set"> </action> <action application="set"> </action> Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Med… Sent from the freeswitch-users mailing list archive at Nabble.com.

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