- FreeSWITCH-users - FreeSWITCH Mailing Lists

webphone app
by budi wibowo 27 Apr '11

27 Apr '11

looking for webphone sip based on flash. any info, please share thx budi wibowo

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Python vs LUA
by Anton VG 27 Apr '11

27 Apr '11

Hello! Cannot get answer for myself for languages processing overhead/preference, I personally would like to use Python for scripting under Freeswitch, but considering some recent list activity about low CPS while using python scripts, etc querstioning, maybe it's better to use LUA instead? WIKI or list does not answer this questions in satisfying manner unfortunatelly. For DB intensive scripts - DB connections pool is a very important stuff, having direct influence performance and CPS specifically. It looks that lua might be lighter and more tightly integrated to FS, but if processing overhead difference is just a few times python is still preferred for me (and many others) since much more powerfull/libraryreach/etc. I'ts would be just nice to clear this question. 1. Is there any comparisions on plain processing overhead? 2. Any data on memory consumtption per active python process? 2. Is there a freeswitch DB pool connection interface like for LUA in mod_python? 3. If not - what would be the known technique to use external DB connection pool from inside the FS python script? 4. Seems logical, but please confirm/reject - once executed (compiled to the bytecode) - the second time the python script under FS will be executed as precompiled bytecode? Regards, Anton.

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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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Out-of-memory problem (dmesg: [1950219.067153] Killed process 12772 (freeswitch))
by Markus Mueller 27 Apr '11

27 Apr '11

Hello Freeswitch, after running Freeswitch for some weeks the system resulted in a out-of-memory problem. No ssh-login is possible, only on console. Please see http://projekte.priv.de/freeswitch/dmesg for the kernel messages. The memory statistics is interesting: In the http://projekte.priv.de/freeswitch/memory , wich is /proc/meminfo, you can see that SUnreclaim is very big... I am using Debian Squezze with the dahdi-source package to access the, libpri-1.4.12-beta3 and the git fetched with "git clone git:/ /git.freeswitch.org/freeswitch.git " from "Wed Mar 2 20:50:33 2011 -0600" (last entry in git log). I am glad for any hits or suggestions! Regards, Markus Mueller

2 1

FreeTDM: ISDN-Port (D-Channel?) useable only one time
by Markus Mueller 27 Apr '11

27 Apr '11

Hi Freeswitch users, if I get an incomming call (FAX), the call is answered and successfully processed, this channel don't get any new calls in until the channel has been reseted by "ftdm stop" and "ftdm start". Please see the log file on http://projekte.priv.de/freeswitch/log You can see there that a fax is comming in on three ISDN lines, but only accepted on one of them. On the first fax it comes in on all three ISDN lines, on the second fax only on two ISDN lines, the third fax is only comming in on one line and all further faxes can't be accepted. After resetet with "ftdm stop" and "ftdm start" it is comming in again. Further config files can be found on http://projekte.priv.de/freeswitch/freetdm.conf.xml http://projekte.priv.de/freeswitch/dialplan.xml http://projekte.priv.de/freeswitch/freetdm.conf http://projekte.priv.de/freeswitch/dahdi_system.conf I am using Debian Squezze with the dahdi-source package to access the, libpri-1.4.12-beta3 and the git fetched with "git clone git:/ /git.freeswitch.org/freeswitch.git " from "Wed Mar 2 20:50:33 2011 -0600" (last entry in git log). Als hardware I have three singleport HFC Cards: 04:00.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) 04:01.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) 04:02.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) root@sip:/usr/local/freeswitch/conf# dahdi_scan [1] active=yes alarms=OK description=HFC-S PCI A ISDN card 0 [TE] name=ZTHFC1 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 04 Slot 01 basechan=1 totchans=3 irq=16 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding=AMI framing=CCS [2] active=yes alarms=OK description=HFC-S PCI A ISDN card 1 [TE] name=ZTHFC2 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 04 Slot 02 basechan=4 totchans=3 irq=17 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding=AMI framing=CCS [3] active=yes alarms=OK description=HFC-S PCI A ISDN card 2 [TE] name=ZTHFC3 manufacturer=Cologne Chips devicetype=HFC-S PCI-A ISDN location=PCI Bus 04 Slot 03 basechan=7 totchans=3 irq=18 type=digital-TE syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI framing_opts=CCS coding=AMI framing=CCS root@sip:/usr/local/freeswitch/conf# I am glad for any hits or suggestions! Regards, Markus Mueller

1 0

Freetdm - Wanpipe - ftmod_sangoma_isdn
by Dennis Young 26 Apr '11

26 Apr '11

All, Last month I upgraded from using ftmod_sangoma_boost to ftmod_sangoma_isdn, as a result, I'm no longer getting caller_name on inbound calls. I can see the caller_number but the caller_name is blank. I'm running windows 32-bit, sangoma A102 card, wanpipe_6_42_1, libsng_isdn-7.1.0, GIT-Head from 4/14/11. I'm sure it's a configuration issue because it was working with boost. Can someone point me in the right direction? ------- wanpipe.conf ----------- [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 ------ freetdm.conf ----------- [span wanpipe wp1] trunk_type => T1 group => grp1 b-channel => 1:1-23 d-channel => 1:24 [span wanpipe wp2] trunk_type => T1 group => grp1 b-channel => 2:1-23 d-channel => 2:24 --------- freetdm.conf.xml -------- <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> </settings> <config_profiles> <profile name="twtelecom"> <param name="facility" value="no" /> </profile> </config_profiles> <sangoma_pri_spans> <span name="wp1" cfgprofile="twtelecom"> <param name="switchtype" value="national" /> <param name="interface" value="cpe"/> <param name="dialplan" value="XML"/> <param name="context" value="public" /> </span> <span name="wp2" cfgprofile="twtelecom"> <param name="switchtype" value="national" /> <param name="interface" value="cpe"/> <param name="dialplan" value="XML"/> <param name="context" value="public" /> </span> </sangoma_pri_spans> </configuration> ...dly Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law. Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.

2 1

fs prepaid wholesale
by budi wibowo 26 Apr '11

26 Apr '11

hi any idea to implement prepaid wholesale for FS? thx budi

4 4

absolute_codec_string question
by Nicolas Brenner 26 Apr '11

26 Apr '11

Hi, I'm trying to force the use of a specific codec when originating a call from the command line. This is my codecs config in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,SPEEX,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,SPEEX,G729"/> If I just originate a call: originate {ignore_early_media}sofia/gateway/mygateway/444444 &bridge(user/1001) I get the following SDP: v=0 o=FreeSWITCH 1303727406 1303727407 IN IP4 127.0.0.1 s=FreeSWITCH c=IN IP4 127.0.0.1 t=0 0 m=audio 17054 RTP/AVP 0 8 98 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 If I originate a call with the absolute_codec_string variable set like this: originate {ignore_early_media,absolute_codec_string='G729,PCMU'}sofia/gateway/ mygateway/444444 &bridge(user/1001) I get the following SDP: v=0 o=FreeSWITCH 1303719860 1303719861 IN IP4 127.0.0.1 s=FreeSWITCH c=IN IP4 127.0.0.1 t=0 0 m=audio 24894 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 And it doesn't matter the value I give to the absolute_codec_string, I will get the same SDP, except if I specify SPEEX, then I'll get the same SDP as the one above. Am I missing something, or is this the way it's supposed to work? I also tried with the codec_string variable, and I get the same results. I'm using the following version from the git repository: head-git-244fd68 2011-03-21 14-27-57 -0400 Thanks for your help, Nicolas

7 14

continue_on_fail for condition only where 100 not received?
by peely 26 Apr '11

26 Apr '11

Hi, I can't seem to get a continue_on_fail code which would only retry if no response whatsoever is received from an endpoint. Does anyone know how I could route to a secondary endpoint ONLY if the initial endpoint does not respond with 100? I guess as a secondary part of the question I would need to know if I can change the INVITE timer to a smaller amount so the second device is attempted in less than 60 seconds. Basically I have a bunch of routing devices within the network I want to route to with a failove option, these devices onward route so I trust if that device is OK not to route to the second device, given these devices log sessions themselves it's vital that if one device receives an INVITE none of the others do for the same session. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/continue-on-fail-for-conditio… Sent from the freeswitch-users mailing list archive at Nabble.com.

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