December 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference "kick all"
by Mike Kane 18 Dec '13

18 Dec '13

Hello, I'm having an issue with conference "kick all". When I release a call before the joining member joins the conference, the conference room does not send a canel or bye to the joining member that hasn't answered yet. It will kick all members currently joined in the conference room. Is is functionality supported, or will kick all just react to calls already in progress in the conference room? Thanks Mike

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FW: how to dial from different gateways according prefix numbers?
by netandtech 18 Dec '13

18 Dec '13

From: netandtech [mailto:bill_vus@hotmail.com] Sent: Wednesday, December 18, 2013 5:27 PM To: freeswitch Subject: how to dial from different gateways according prefix numbers? The following is the extension number that I use in the default.xml What I want to achieve is to Route the calls for landlines to different gateways and calls to mobiles from another gateway. <extension name="1002">   <condition field="destination_number" expression="^69(\d+)$"> <action application="transfer" data="sofia/gateway/myvoip/$0"/> </condition> <condition field="destination_number" expression="^22810(\d+)$"/> <action application="transfer" data="sofia/gateway/voipdito2/$0"/> </extension>

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H323 Configuration
by Richard Mace 18 Dec '13

18 Dec '13

Hi, Within the h323.config.xml file there is an entry for: <listeners> <listener name="default"> <param name="h323-ip" value="$${local_ip_v4}"/> <param name="h323-port" value="1720"/> </listener> </listeners> is the "h323-ip" value the IP address of the H323 server? Richard

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Freeswitch channel is down but stays alive
by Tornike Zedginidze 18 Dec '13

18 Dec '13

Hi, In very rare case, some channels stay active till I execute uuid_session_heartbeat or any uuid command. I run Freeswitch 1.2.14 and there is an LUA listener handling calls' dialplans and hangups. In approximately 12 000 calls, 1 channel stays alive even it's state is down. You can see CLI output here: http://pastebin.com/Z8ycPX3g What do you suggest, where can the issue be? Also not only hearbeat, but any uuid command finishes the call. Thanks

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FS sending INFO requests
by Grant Bagdasarian 17 Dec '13

17 Dec '13

Hello, Why does FS keep sending INFO requests? How do I disable these? Regards, Grant

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number translation in freeswitch
by Prakash Ganesan 17 Dec '13

17 Dec '13

Hi, Is it possible to do a 1 to 1 mapping in a database in free switch and get the number translated?. for example 123456789 is mapped to 987654321. if an INVITE comes with 123456789 to FS, it should translate to 987654321 in RURI and To header before it sends the invite out. this trranslation should be for 100,000 number present in the database? is it a vaible solution in FS? Many thanks Prakash

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Fwd: mod_nibble hangup problems
by Jason Sia 17 Dec '13

17 Dec '13

Hi, When the call is cut off by mod_nibble it sends a hangup signal. However, it is resulting to a multiple re invite. Is there a misconfiguration somewhere? I just copied the examples from the user guide. Thanks, Jason

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autofix incorrect packetization
by Hristo Trendev 17 Dec '13

17 Dec '13

Hi, I get choppy audio in conferences from endpoints that send G722@30mspackets. The endpoint in question is a FRITZ!Box CPE and this seems to be a known problem with these type of devices. They simply ignore the offered ptime in SDP and stick to sending 30ms RTP packets no matter what. The last time I tested with a FRITZ!Box CPE was in February and I am absolutely sure that FS used to auto fix this type of problems in the past and when this was happening I would see the following lines in the logs: "Asynchronous PTIME not supported, changing our end from 20 to 30" "Changing Codec from G722@20ms@8000hz to G722@30ms@8000hz" and also something to do with timer resolution being increased to deal with 30ms intervals. This doesn't happen any more (v1.2.14) and all I can hear is choppy audio from these CPEs. The only relevant lines I could find in the logs were: "mod_conference.c:3590 Setup timer soft success interval: 20 samples: 160" "switch_core_io.c:759 Engaging Read Buffer at 640 bytes vs 240" I also verified with a packet trace that FS is indeed receiving RTP packets with payload of 240 bytes, which would correspond to 30ms and is sending packets with 160 bytes of payload (20ms), so as far as I can tell this would actually classify as async ptime. I only see the problem one way - the choppy audio comes from the remote party, which is sending the 30ms packets into the conference. The remote party however still hears normal audio coming from the conference (although it is packetized and sent out at 20ms). Could it be that for some reason FS or the conference application is mistakenly treating the 30ms packets as 20ms ones when mixing the audio into the conference? I do recognize that this is caused by misbehaving end device and it is not a FS issue, but I just wanted to know if someone else is also seeing similar problems or if this can somehow be related to a recent change in the FS code? I have updated to v1.2.15 so I can retest, but I will only get a chance to do it on Friday. Best Hristo

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List Etiquette (just a reminder)
by Ken Rice 17 Dec '13

17 Dec '13

Hey guys, just a reminder (as someone just did this) Please do not hijack threads... If you have a new problem start a new thread but sending a new message to FreeSWITCH-users(a)lists.freeswitch.org If you have just subscribed to the list or have not posted to the list, chances are you are still moderated. We do this to reduce spam and it works quite well. If you post to the list and it doesn¹t come right through chances are you are moderated and one of the list mods will get to checking the mod queue shortly. Please keep in mind that most of the list mods are based in the USA. That means between 10PM and 8AM Eastern Time in the US, and on weekends there may be a delay in getting the queue processed. You don¹t have to post 4 times (and you most certainly should not post 4 times attempting to hijack other threads.) Have a Great Week! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Mod LCR SQL sub-query for better routing
by Victor Chukalovskiy 17 Dec '13

17 Dec '13

Hello, For those interested, I added a piece to mod lcr wiki. It makes sorting / routing logic better than default logic: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL_with_sub-query_-_for_rea… Why it helps: Rates rates can often be given both on per-NPA or per-NPANXX level depending on the carrier and on the NPA. Moreover, some carriers may have NPANXX rate lower than the corresponding NPA rate, while others will have it inverse. Neither simple ORDER BY rate, prefix; nor ORDER BY prefix, rate; give the truly cheapest route. The LCR logic should be two-step process to accommodate this. Cheers, -Victor

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