June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

XML_CURL and static XML (loaded in memory)
by Vik Killa 11 Jun '12

11 Jun '12

Is it possible to load freeswitch with static XML and then load mod_xml_curl? It seems FreeSWITCH does not work correctly when this is done. Here is an example: In freeswitch.xml, I have: <section name="directory" description="User Directory"> <X-PRE-PROCESS cmd="include" data="static/directory.xml"/> </section> This loads my directory configuration statically in memory. After I start FreeSWITCH, I load mod_xml_curl with 'fs_cli> load mod_xml_curl' (mod_xml_curl is not loaded when freeswitch starts) Here is my XML_CURL configuration: <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="default"> <param name="gateway-url" value="192.168.0.158" bindings="directory" /> </binding> </bindings> </configuration> FreeSWITCH does not seem to function properly after XML_CURL is loaded. If I run the 'user_data' API, FS hangs there for a long time and eventually returns the password from memory (it does not make an XML_CURL request) Here is example: fs_cli> user_data 1000(a)domain.com param password I guess the ultimate question is this: Is there any way to setup XML_CURL to failover to static XML (loaded in memory)? Thanks!

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checking voicemail without ID
by Vik Killa 11 Jun '12

11 Jun '12

Why is it that a caller can enter a non-existent voicemail box and made up PIN and login to the non-existent mailbox? I've tested this with *98 and entering random numbers followed by a password of random numbers.

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Garbled Leg Recording G729 with Asynchronous ptime
by Colin Sindle 11 Jun '12

11 Jun '12

Hi, Please, I was wondering if there is anything obvious that we are missing when trying to record a G729 call with asynchronous pimes (in: 20ms, out: 60ms). The problem case call flow is: inbound leg (ptime 20 ms), start recording, playback a short message, bridge to outbound (destination can only handle a ptime of 60 ms). We are using mod_com_g729's codec. The audio while on the call in normally fine (rarely there is one way speech depending on inbound/outbound provider), but *the recording has a garbled outbound (Callee) leg* (inbound/Caller voice is fine.). We have tried all the combinations of G729@20i, G729@60i, rtp-autofix-timing, and default-ptimes to no avail. We are using git HEAD. I do see this on fs_cli: 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.543269 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xa8894898 (nil) 2012-06-07 16:56:45.583270 [WARNING] mod_sofia.c:1158 *Asynchronous PTIME not supported, changing our end from 20 to 60* 2012-06-07 16:56:45.583270 [DEBUG] sofia_glue.c:2931 Changing Codec from G729@20ms@8000hz to G729@60ms@8000hz 2012-06-07 16:56:45.623271 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xa86fe340 0xa86f1e78 Should this be working, are we missing something obvious? Thank you, Kind regards, Colin

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Time needed for ringing a call group
by Robert Longfield 11 Jun '12

11 Jun '12

I have support call group that works as it should. When a caller selects the support call group it rings all the support extensions as it should. The problem arises when a support member is logged in to their extension but is not at their computer. The call will continue to ring the call group forever. I would like to change this so it will only ring for a set amount of time. Currently if the support group is busy or they manually decline the call the incoming call will be transferred to the general extension. I would like to mimic this behaviour when the call goes unanswered after 20 secs or so. Here is the Call Group Settings: <extension name="group_dial_support"> <condition field="destination_number" expression="^2001$"> <action application="set" data="call_timeout=2000"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="group/support@$${domain}"/> <action application="answer"/> <action application="sleep" data="100"/> <action application="bridge" data="sofia/internal/1000@$${domain}"/> </condition> </extension> I hope I’ve explained my problem clearly. -Rob Robert Longfield SEO Specialist KL Insight http://klinsight.com/ Tel: 613-344-2116 | Fax: 613.634.7029 993 Princess Street, Suite 212 Kingston, ON K7L 1H3, Canada Notice of Confidentiality: The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review re-transmission dissemination or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this in error please contact the sender immediately by return electronic transmission and then immediately delete this transmission including all attachments without copying distributing or disclosing same.

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Controlling caller_id_name
by Torstein Knutsen 11 Jun '12

11 Jun '12

Hi I'm trying to control the "caller_id_name" variable used in conferencing. My dialplan looks like this : <extension name="public_extensions"> <condition field="destination_number" expression="^(0meet)$"> <action application="conference" data="72013@webconf"/> When I call into the conference, I do an API "conference xml_list" which gives details of the conferencees .. I want to be able to change the caller_id_name in the listing, but all my efforts for doing this via dialplan fails. Is this even possible somehow ? I tried to add this to the dialplan before executing the conference : <action application="export" data="caller_id_name=Hobbes" /> <action application="set" data="caller_id_name=Hobbes" /> <action application="export" data="effective_caller_id_name=Hobbes" /> <action application="export" data="effective_caller_id_name=Hobbes" /> <action application="set" data="origination_caller_id_name=Hobbes" /> <action application="export" data="origination_caller_id_name=Hobbes" /> But to no avail... Any pointers would be appreciated. Thank you Torstein

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Re: [Freeswitch-users] Gateway failover and ports limitation
by awais nazir 11 Jun '12

11 Jun '12

Hi, Provider expects me to control limit at my end. I have to do the scenario with static dialplan not with database/odbc. Thanks in advance for helping. On Mon, Jun 11, 2012 at 2:56 PM, Peter Olsson <peter.olsson(a)visionutveckling.se> wrote: If the provider only allows you to use 30 channels, I guess they will send a specific error to you if you try to do more calls, so just handle that error as you handle 480, 503 etc. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För awais nazir Skickat: den 11 juni 2012 11:43 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Gateway failover and ports limitation Hi Can u help me a bit here. I have a scenario like GW_priority_1_30_ports GW_priority_2_60_ports So if first gatway is down or fails with giving cause code (480 or 503 ) OR its 30 channels are occupied then next carrier should be selected. And so on. I can see failover config in dial plan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Fa… . But blending it with limit and doing failover on Gateway down and on the basis of some cause code is confusing. Your help and hints are appreciated. --waisee !DSPAM:4fd5bc0432761819510119!

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Re: [Freeswitch-users] Gateway failover and ports limitation
by Peter Olsson 11 Jun '12

11 Jun '12

If the provider only allows you to use 30 channels, I guess they will send a specific error to you if you try to do more calls, so just handle that error as you handle 480, 503 etc. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För awais nazir Skickat: den 11 juni 2012 11:43 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] Gateway failover and ports limitation Hi Can u help me a bit here. I have a scenario like GW_priority_1_30_ports GW_priority_2_60_ports So if first gatway is down or fails with giving cause code (480 or 503 ) OR its 30 channels are occupied then next carrier should be selected. And so on. I can see failover config in dial plan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Fa… . But blending it with limit and doing failover on Gateway down and on the basis of some cause code is confusing. Your help and hints are appreciated. --waisee !DSPAM:4fd5bc0432761819510119!

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Gateway failover and ports limitation
by awais nazir 11 Jun '12

11 Jun '12

Hi Can u help me a bit here. I have a scenario like GW_priority_1_30_ports GW_priority_2_60_ports So if first gatway is down or fails with giving cause code (480 or 503 ) OR its 30 channels are occupied then next carrier should be selected. And so on. I can see failover config in dial plan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Fa… . But blending it with limit and doing failover on Gateway down and on the basis of some cause code is confusing. Your help and hints are appreciated. --waisee

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how to use codec g729 on freeswitch ?
by Samira Mh 11 Jun '12

11 Jun '12

hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ?

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(no subject)
by Tom C 11 Jun '12

11 Jun '12

On the "Performance testing and configurations" page of the wiki, under Recommended SIP Settings, it says the following: libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles Is this still true? It appears to me that every call, and maybe every channel, is getting its own thread. Or is the above comment referring to call management rather than actually processing audio?

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