February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_pocketsphinx
by roy 19 Jun '11

19 Jun '11

Hi, Trying to get mod_pocketsphinx working from the current git tree, FreeSWITCH version: 1.0.head (git-2629a57 2010-06-16 22-42-15 +0200) however, I got this error during startup: <snip> 2010-06-17 11:18:08.032645 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: ngram_model_get_counts** </snip> Lib paths were already defined in /etc/ld.so.conf as: include /usr/local/freeswitch/lib /usr/local/freeswitch/mod # ls -al mod/mod_pocketsphinx.* -rwxr-xr-x 1 root staff 1467 Jun 17 10:49 mod/mod_pocketsphinx.la -rwxr-xr-x 1 root staff 905769 Jun 17 10:49 mod/mod_pocketsphinx.so Was wondering if someone was able to get this working. Manually downloading the current pocketsphinx and sphinxbase and renaming them into 0.4.99 and rebuilding didn't help either. Would appreciate if anyone can point me into the right direction. TIA. Roy

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Music On Hold
by Chia-Yen Wu 16 Jun '11

16 Jun '11

Hello, I am using freeswitch as an IVR server. I would like to be able to put the call on hold and play music while performing some other tasks in the lua. when the task is complete , the music stop and the IVR continue but i cant find a way to do that, is there any function can do "music on hold" in dialplan ? Thank you

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mod_fsv, record and playback options
by William Kendi ... 27 May '11

27 May '11

Greetings, Currently, I am trying to record and play videos through FreeSWITCH using the "mod_fsv" module. Some options like timeouts and interrupts by dtmf digits seems to be lacking in the "mod_fsv" module and I cant even stop the video recording using the "break" application. The video recording stop when the user disconnects. Anyone has any ideas about how to stop the FSV video recording without disconnecting the user? Thanks in advance.

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T.38 via UPDATE request
by Peter P GMX 19 May '11

19 May '11

Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter Here's the UPDATE message: UPDATE sip:mod_sofia@82.115.xx.xxx:5080 SIP/2.0. Call-ID: 5283fe4e-334f-122d-d1b9-001517956764. Contact: <sip:212.91.xxx.xxx:5060>. Content-Type: application/sdp. CSeq: 16340063 UPDATE. From: <sip:06912345678@sip.telco.de>;tag=00-08135-017e041a-4d21f6037. Max-Forwards: 31. Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>. To: "030987654321" <sip:030987654321@82.115.xx.xxx>;tag=pH663F4S02erm. User-Agent: Cirpack/v4.41e (gw_sip). Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31. Content-Length: 300. . v=0. o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx. s=SIP Call. c=IN IP4 212.91.xxx.xx. t=0 0. m=image 6860 udptl t38. a=sendrecv. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1000. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy.

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freeSWITCH segfault on openwrt
by Woody Dickson 18 May '11

18 May '11

Hi, I manage to get freeSWITCH to compile and run on openwrt, but then freeSWITCH keeps crashing when the number of calls reaches 25 consistently. Content of the core file returns nothing when opened with gdb. Does anyone have any clue as to why this is happening and how to fix it? thanks, woody

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How to read RTP package?
by Frankie Yiu 13 May '11

13 May '11

Hi there, I am new to FreeSwitch, and I have a question to ask. If I want to read the RTP package in C++ or C#, what function should I call to get it? Currently I have can make a call to a phone and play an audio. Thanks in advance. Frankie ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

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How to filter Skypopen chat message in ESL mode?
by xuyan yang 27 Apr '11

27 Apr '11

hi, I am writing ESL program on FreeSWITCH Version 1.0.head (git-8825b6e 2010-11-28 17-15-39 -0500) I need to handle skype chat message with a inbound ESL connection. But there are 1 or 2 esl events received randomly for each message. the first short one is alway generated, but the second one is random. So when trying to filter according to UUID, nothing is caught in many cases. If I make filter based on skype ID, duplicated messages are often heard. I do not why the behavior of the second event is random. How should I setup the filter to get 1 and only 1 event for each chat message? Thanks. the first is a short one with the following header: [Event-Name] = [MESSAGE] [Event-Calling-Function] = [incoming_chatmessage] [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT] [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500] [Content-Length] = [1] [subject] = [SIMPLE MESSAGE] [FreeSWITCH-IPv4] = [192.168.0.3] [hint] = [niqizhi] [from] = [niqizhi] [Event-Date-Local] = [2010-12-01 21:02:37] [proto] = [skype] [FreeSWITCH-IPv6] = [::1] [id] = [5334] [Event-Calling-File] = [mod_skypopen.c] [Event-Date-Timestamp] = [1291237357051788] [FreeSWITCH-Hostname] = [EYSRV] [login] = [interface1] [during-call] = [true] [Event-Calling-Line-Number] = [2915] [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a] the second is much longer: [Caller-Source] = [mod_skypopen] [Event-Calling-Function] = [incoming_chatmessage] [Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT] [Answer-State] = [answered] [FreeSWITCH-IPv4] = [192.168.0.3] [Channel-State] = [CS_EXECUTE] [Channel-Read-Codec-Bit-Rate] = [256000] [FreeSWITCH-IPv6] = [::1] [Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500] [Channel-Read-Codec-Rate] = [16000] [Caller-Destination-Number] = [5655] [Caller-Channel-Transfer-Time] = [0] [Channel-Call-State] = [ACTIVE] [Caller-Channel-Progress-Media-Time] = [0] [FreeSWITCH-Hostname] = [EYSRV] [Caller-Channel-Answered-Time] = [1291237326697085] [login] = [interface1] [during-call] = [true] [Channel-Name] = [skypopen/interface1] [Caller-Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500] [Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500] [Channel-Read-Codec-Name] = [L16] [Caller-Channel-Name] = [skypopen/interface1] [Caller-Caller-ID-Number] = [niqizhi] [Event-Date-Timestamp] = [1291237357051788] [Channel-State-Number] = [4] [Event-Calling-Line-Number] = [2888] [chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a] [Event-Name] = [MESSAGE] [Content-Length] = [1] [subject] = [SIMPLE MESSAGE] [Caller-Caller-ID-Name] = [niqizhi] [from] = [niqizhi] [Caller-Dialplan] = [XML] [Caller-Channel-Hangup-Time] = [0] [id] = [5334] [Caller-Profile-Index] = [1] [Caller-Direction] = [inbound] [Caller-Username] = [skypopen] [Channel-Write-Codec-Name] = [L16] [Call-Direction] = [inbound] [Caller-Screen-Bit] = [true] [hint] = [niqizhi] [Caller-Privacy-Hide-Number] = [false] [Event-Date-Local] = [2010-12-01 21:02:37] [proto] = [skype] [Caller-Channel-Created-Time] = [1291237326468855] [Event-Calling-File] = [mod_skypopen.c] [Caller-Channel-Progress-Time] = [0] [Caller-Privacy-Hide-Name] = [false] [Channel-Write-Codec-Rate] = [16000] [Caller-Context] = [default] [Channel-Write-Codec-Bit-Rate] = [256000] [Presence-Call-Direction] = [inbound] [Caller-Profile-Created-Time] = [1291237326468855]

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Difference Between "realm" and "challenge-realm" in sip_profiles
by Jerry Richards 21 Apr '11

21 Apr '11

Hello All, What is the difference between the parameters "realm" and "challenge-realm" in the conf/sip_profiles tree? Thanks, Jerry

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Mitel 3300 ICP "Multicall" Equivalent
by Alex Crow 18 Apr '11

18 Apr '11

Hello list, Our company is considering a migration from Mitel 3300 ICP to an alternative platform, and of course FreeSwitch is a major contender, being as we are not enamoured with "black boxes" and proprietary protocols. There is a function in the Mitel that ideally we'd like to replicate (on whatever hardphones are most appropriate). "Multicall" allows you to assign one number to a button (other than the "main line" or default extension) on several phones (up to 31 on the 3300) such that a call (or any number of concurrent calls up to the number of phones configured with that button) to that number can be signalled on all phones and picked up by anyone with that button configured on that phone. We use that for calling direct to regional teams. No barging is allowed, and the button only changes state (from off to flashing) when a call needs answering on the multicall number. Said button assignment can be set to either ring or not. Once the call has been answered by one user of the multicall the light goes off on other configured extensions and another incoming call can be answered in the same way by another extension, regardless of the status of all other extensions with the same button configured. I was looking at the SCA/SLA stuff on the wiki, which is a close as I could find to this behaviour, but barging is unwanted and we require the line appearance, ringing and answering to be as described above. My question, given the above, is: 1. If SCA is able to do this, what is the best phone to support it (Polycom seems good from the wiki) 2. If SCA is not appropriate, I have the feeling that queues would be the next best option - if these can be assigned to a button on the phone so much the better. 3. BLF doesn't look right as it seems it needs at least one extension with that as its "primary" number. There is some flexibilty here, but to ease transition it would be best to keep the functionality as close as possible. IMHO the Mitel way is really odd but it's got a lot of traction having been used here for 7 years+. Thanks Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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bind_meta_app blocks DTMFs?
by Mariusz Czulada 15 Apr '11

15 Apr '11

Hi All, I'm trying to implement subconference feature, to allow some participants go (or rather "gosub") to a another conference room to discuss something then get back to the main conference. I use FreeSWITCH Version 1.0.head (hacked-20110112T172836Z). Bellow some configure extracts: >From conference.conf.xml: ... <caller-controls> ... <group name="just_exit"> <control action="hangup" digits="#"/> <control action="hangup" digits="1"/> <control action="mute" digits="0"/> </group> ... </caller-controls> ... <profiles> .. <profile name="confX_M"> ... <param name="caller-controls" value="just_exit"/> ... </profile> <profile name="confX_S"> ... <param name="caller-controls" value="just_exit"/> ... </profile> </profiles> ... >From dialplan (own) public/confX.conf.xml: ... <extension name="confX_join"> <condition field="destination_number" expression="^confX_join$"/> <condition field="${confX_num}" expression="^1(\d{4})$"> <action application="bind_meta_app" data="1 a s execute_extension::subX_1"/> <action application="bind_meta_app" data="2 a s execute_extension::subX_2"/> <action application="bind_meta_app" data="3 a s execute_extension::subX_3"/> <action application="conference" data="confX_${cpnfX_num}@confX_M"/> </condition> </extension> ... <extension name="subX_1"> <condition field="destination_number" expression="^subX_1$"> <action application="set" data="pass_rfc2833=true"/> <action application="conference" data="confX_${cpnfX_num}_1@confX_S"/> </condition> </extension> <extension name="subX_2"> <condition field="destination_number" expression="^subX_2$"> <action application="set" data="pass_rfc2833=true"/> <action application="conference" data="confX_${cpnfX_num}_2@confX_S"/> </condition> </extension> <extension name="subX_3"> <condition field="destination_number" expression="^subX_3$"> <action application="set" data="pass_rfc2833=true"/> <action application="conference" data="confX_${cpnfX_num}_3@confX_S"/> </condition> </extension> ... During connection user is first handled by IVR, where he enters conference number (assume: "1234") which is stored in "confX_num" variable. Then call is transfered to "confX_join". Here I assign actions for *1, *2 and *3 DTMF sequences then join main conference ("confX_1234_M"). When some user presses i.e. "*2" then he also enters subconf 2 ("confX_1234_2"). Those users attached to "confX_1234_2" are at the same time still attached also to "confX_1234_M", but found muted and deaf to other participants. Then, leaving subconference they will be back "alive" in master conference, which I tested by kicking them from "confX_1234_2" manually. Now my problem begins. Then user is on "confX_1234_M" dialing "#" forces user to leave conference, as defined in "just-exit" controls. But when user enters also "confX_1234_2" DTMF codes are no longer processed, neither by master conference nor by subconference. I set "pass_rfc2833=true" but it is not helpful. Did I missed to set something [1] in the dialplan or other config files, or [2] actually no DTMF codes can be processed while in "*2" leg? And if [2]==true: would it be possible to develop such DTMF handling? BTW: DTMFs are not processed while in subconference but if user is kicked off from subconf back to main conference, one can use DTMF codes again to (un)mute (0), join sub (*2) or leave (# or 1). TIA for your help or tips. Regards, Mariusz

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