April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

frame per packet
by Kenneth Taylor 27 Apr '11

27 Apr '11

Hi, I'm building a voip client for android phones over gsm and consider using FS as sever. In order to save cellular costs I want to put more than one frame per Rtp packet. Is there any easy solution to do it in FS? Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go to lower levels? I don't think I should do it in the codec, it may cause problems in the rtp sequence number and probably a lot more. TNX, Ken

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mod_callcenter and uuid-standby
by Peter P GMX 27 Apr '11

27 Apr '11

Hello, I am trying to use the mod_callcenter functionality. This works nicely so far so thank you to everybody involved for programming this nice module! But I am stuck somehow with uuid-standby. Can anybody explain how uuid-standby works? Another question: In the Asterisk based callcenter solution named "Vicidial", an agent can be held permanently in a conference, waiting for calls who are bridged to his uuid in the conference. Can this be haviour be done with mod_callcenter? Best regards Peter

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FreeSWITCH Conference Call Today - Safi Systems IVR Builder
by Michael Collins 27 Apr '11

27 Apr '11

Hello all! We are happy to have Zac Wolfe from Safi Systems joining us today to talk about the IVR building system they have created. There is a gotomeeting.comlink on the agenda page here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_27 We look forward to this discussion! -Michael

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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

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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

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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.

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