December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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 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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Invalid Application hash
by David Wafula 29 Mar '11

29 Mar '11

Hi all, Notice the last line in the log below when i dial 1001. This is from the latest git as of this morning. Any pointers? thanks. Dialplan: sofia/internal/1000(a)146.141.76.153 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(dialed_extension=1001) Dialplan: sofia/internal/1000(a)146.141.76.153 Action export(dialed_extension=1001) Dialplan: sofia/internal/1000(a)146.141.76.153 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1000(a)146.141.76.153 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1000(a)146.141.76.153 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(call_timeout=30) Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(continue_on_fail=true) Dialplan: sofia/internal/1000(a)146.141.76.153 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1000(a)146.141.76.153 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1000(a)146.141.76.153 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1000(a)146.141.76.153 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1000(a)146.141.76.153 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1000(a)146.141.76.153 Action answer() Dialplan: sofia/internal/1000(a)146.141.76.153 Action sleep(1000) Dialplan: sofia/internal/1000(a)146.141.76.153 Action voicemail(default ${domain_name} ${dialed_extension}) 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000(a)146.141.76.153) State Change CS_ROUTING -> CS_EXECUTE 2010-10-25 10:47:47.193206 [DEBUG] switch_core_session.c:1057 Send signal sofia/internal/1000(a)146.141.76.153 [BREAK] 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/1000(a)146.141.76.153) State ROUTING going to sleep 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:318 (sofia/internal/1000(a)146.141.76.153) Running State Change CS_EXECUTE 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/1000(a)146.141.76.153) State EXECUTE 2010-10-25 10:47:47.193206 [DEBUG] mod_sofia.c:239 sofia/internal/ 1000(a)146.141.76.153 SOFIA EXECUTE 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000(a)146.141.76.153 Standard EXECUTE EXECUTE sofia/internal/1000(a)146.141.76.153 set(open=true) 2010-10-25 10:47:47.193206 [DEBUG] mod_dptools.c:1028 sofia/internal/ 1000(a)146.141.76.153 SET [open]=[true] 2010-10-25 10:47:47.193206 [ERR] switch_core_session.c:1807 Invalid Application hash -- David Wafula

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Re: [Freeswitch-users] outgoing callerid freetdm TDM400
by François Legal 28 Mar '11

28 Mar '11

That looks like the problem I had with my FXS ports sending CID not understood by european phones. In which country are you / what kind of analog phones are you using ? François On Wed, 8 Sep 2010 13:16:08 +1000, Brian May <brian(a)microcomaustralia.com.au> wrote: > On 8 September 2010 11:21, Moises Silva <moises.silva(a)gmail.com> wrote: >> Pastebin a debug log for a call. Also, before placing the call, create the >> directory /var/log/freetdm and then execute this in FreeSWITCH CLI: >> ftdm trace /var/log/freetdm/test <span> <chan> > > Hopefully I got this right ;-) > > <http://pastebin.com/dEXkkNaK> > > To me everything looks good in this listing. > > Not sure what I can do with the freetdm trace files...

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Re: [Freeswitch-users] mod_callcenter features
by Rob Hutton 14 Mar '11

14 Mar '11

1. Try something like: reloadxml callcenter_config queue load|reload queue_name 2. I don't believe so. Hit Moc on the IRC channel. I am trying, also... -- Thanks, Rob On Friday 31 December 2010 00:32:46 Rajkumar K wrote: > Hi, > > 1. In mod_callcenter, there is no command like "callcenter_config queue add" > for adding queues dynamically. We need to configure it in > callcenter.conf.xml file and run "callcenter_config queue reload <queue > name>". So is there any other ways to add queues. > > 2. The mod_callcenter automatically bridges the call once it identifies the > available agent. But the requirement is to identify the available agent and > leave the bridge control to our dial plan. There we need to other operations > (like announcing caller name to agent) instead of bridging the call. > > regards > rajkumar k > > On Thu, Dec 30, 2010 at 6:12 PM, Rob Hutton <justlikeef(a)gmail.com> wrote: > > > Some of the API commands are on in the Wiki under the heading API: > > > > http://wiki.freeswitch.org/wiki/Mod_callcenter > > > > Everything starts with callcenter_config, so you can feel your way around > > the command line and everything is fairly self explanatory. > > > > What do you mean control the call flow? Can you give an example of what > > you are trying to accomplish? > > > > -- > > Thanks, > > Rob > > On Monday 27 December 2010 04:42:29 rajkumar wrote: > > > > > > Hi, > > > > > > I am developing an application with mod_callcenter. I need to know the > > > following about mod_callcenter. > > > > > > * Is it possible to add/update/delete the queue configurations > > dynamically > > > without using static xml configurations. > > > * How can I control the call flow (for playback message and recording) > > > before and after bridging the call. > > > > > > Thanks in advance > > > > > > regards > > > rajkumar k > > > > > >

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Call back from voice mail creates loop
by Tim St. Pierre 11 Mar '11

11 Mar '11

Hello, I'm trying to set up the option in voicemail so that users can return a call directly from VM. In my voicemail.conf.xml profile I have specified the dialplan as XML and the context as "internal", which is what I use for calls coming from our endpoints or authenticated users. In our internal dialplan, we do some pattern matching and call an appropriate lua dialplan script to do build a set of instructions. Problem is, the lua scripts are looking at the destination_number variable to decide where to send the call. This variable is still set to "voicemail" from when the user called into their voicemail. Shouldn't the voicemail app have set that to the number we want to call back? I used the info app to look at the variables, and I didn't see anything obvious. Is destination_number the correct variable to use in a lua script that generates a dialplan? -Tim

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mod callcenter
by Sam 10 Mar '11

10 Mar '11

Hello, Was testing callcenter module and found out that at times it gives error " invalid application callcenter " and after reloading the module it works fine. Some time also happens that if I reload the module it do not reloads the parameters of the callcenter like the agents & tires. It just unload & reloads even if there are changes to the specifications of agents. Regds Sam

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