March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Complete wholesale app in freeswitch
by David Villasmil 21 Jul '11

21 Jul '11

Hello Guys, I'm finishing a "complete" wholesale application created on freeswitch and I was wondering whether it would be a good idea to put it up on the wiki. I just don't know how. Features include all the following parameters configurable via web interface: - Profile creation based on server IP where traffic is received. You can have multiple IPs, system will create multiple profiles/diaplans so it can differentiate. - i.e. offer to the same customer a "gold" routing on IP1 and cheap routing on IP2 - Customer add/modify/delete - IP source - Rates for client routes based on areacode - Prepaid or postpaid. - When cutomer balance is 0, no more calls are allowed. - limit max channels - Media by-pass - When by-passed, customer and provider will exchange RTPs directly. Else, server will be in the middle. - Provider add/modify/delete - costs for provider routes based on areacode - limit max channels - Routing based on areacode, gives great granularity. - Routes can be assigned multiple gateways/providers which can in turn be distributed based on weigth. Includes overflow to next configured GW. - Basic financial report generation (totals) by customer/provider - Basic traffic ASR/ACD report (totals) by cutomer/provider - Basic user administration. (No access level, only total access) - CDR export to csv file. I also have a prepaid card app... no web interface on that one though... Thanks all David

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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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mod_say in different codecs
by David Notivol 12 May '11

12 May '11

Hi all, I'm trying to setup an IVR server, and I'm using the session:say function from a LUA script. My question is if there's any way of having mod_say playing audios different than audio files; I mean audios encoded in G729, G711ulaw, G711alaw, etc. to avoid having FS making transcoding every time I run a say command. Checking the folders tree for the sounds, I can see the place for the different languages, voices, and wav qualities (8k, 16k...) is clearly specified; but is it a way to place files encoded in different codecs? Thanks in advance. Regards, David Notivol

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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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Outbound caller id for a specific gateway?
by Dmitry Bely 27 Apr '11

27 Apr '11

My VoIP provider requires a specific caller ID set for an outbound call otherwise the call is rejected. Currently I set it just before bridge <action application="set" data="effective_caller_id_name=..."/> <action application="bridge" data="..."/> But it's tedious as there is a number of bridge commands in the dialplan and I still have to explicitly specify the caller id for "originate" command in the FreeSWITCH console. Is it possible to force an outbound caller id on a gateway basis? - Dmitry Bely

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