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modfifo not calling all registered members
by efern75＠hushmail.com 30 Jun '13

30 Jun '13

Good evening all, Have an issue with modfifo not calling all members. This has been happening for a while but can't find out why. I have 10 yealink phones/1 snom phone that have each 3 individual registered extension. (one phone has ext: 101,201,301, the next phone has 102,202,302, etc). I have all 33 extensions registered according to "sofia status profile internal reg". I add each extension to fifo via a script that just runs: fifo_member add FIFO0 {fifo_member_wait=nowait}user/EXTENSION Below is the pastebin of "fifo list" after loading the above command against all registered phones (33 extensions,11 phones) http://pastebin.com/diNU6hbm This is how the fifo.conf.xml file looks like (default): <configuration name="fifo.conf" description="FIFO Configuration"> <settings> <param name="delete-all-outbound-member-on-startup" value="false"/> </settings> <fifos> </fifos> </configuration> When a call comes in and is sent to the FIFO queue via: <action application="fifo" data="FIFO0 in"/> Only 23 channels/call setups are made to the possible 33 extensions that are in fifo list. below is the show channels of this setup: http://pastebin.com/s7TvxGMa Here is a list of the 10 non called extension: 102,305,205,123,308,201,101,206,306,222 After hanging up, this is the "fifo list" dump for reference: http://pastebin.com/9JgD49fL I have ran "fifo debug 7" and /debug 7 at the console and did some test calls and nowhere does it light up with the extensions that are missing for the fifo call setup =\ I have also for example ran ngrep against a phone that only has two out of three INVITES sent (based on show channels) and can verify that only 2 INVITES are being sent. (just wanted to make sure that the phone wasn't rejecting the invite on the missing show channels extension due not not being very fast at seeing it happen). I am running: OS: Debian 6.0.7 (x64) FS: 1.2 latest git (1.2.10+git~20130628T192118Z~929f52b15c) Also, i have shutdown freeswitch, deleted the entire /usr/local/freeswitch/db/ directory content; started freeswitch, waited for all extensions to fill in eventually, and ran the fifo_member add command and can still duplicate this issue =\ Thank you for your assistance on this!!!! Eric

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Limit usage not decreased after attended transfer
by Stanislav Sinyagin 29 Jun '13

29 Jun '13

hi, My user wants to have only one incoming call on a number of his SIP accounts, and no concurrent calls. So, on incoming calls I do <action application="limit" data="hash ${domain} ${availability_username} 1"/> and on outgoing calls I do <action application="limit" data="hash ${domain_name} ${availability_username} -1"/> Problem: when an inbound call arrives, and the user transfers it to another PSTN number and hangs up, limit_usage still shows 1 active conversation, until the transferred call ends. That means, the user already hung up, but he cannot accept new calls until the original call finishes. Any idea how to eliminate this situation and free up the limit counts after transfer? thanks stanislav

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[CRIT] mod_nibblebill.c:558 Failed to log to database!
by mrajib321＠yahoo.com 29 Jun '13

29 Jun '13

Hi, Is anyone running or tested mod_nibblebill on the latest release of FreeSWITCH 1.2.10. A simple internal profile call between two SIP clients completes successfully but fails to debit the balance. I have the following configuration settings Dialplan configuration <action application="set" data="nibble_rate=0.20"/> <action application="set" data="$${nibble_account}"/> User directory configuration <variable name="nibble_account" value="1"/> Nibblebill module configuration <param name="odbc_dsn" value="odbc://freeswitch:freeswitch:freeswitchrw"/> <param name="db_table" value="accounts"/> <param name="db_column_cash" value="cash"/> <param name="db_column_account" value="id"/> ODBC driver has been tested sucessfully with mySQL using isql command. Running FreeSWITCH 1.2.10 on Linux machine FreeSWITCH Version 1.2.10+git~20130625T154058Z~30be15fd52 Any pointers on how to fix the error would be appreciated. Thanks. I received the following errors: 2013-06-28 14:24:49.488199 [ERR] mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE account_code='1'] 2013-06-28 14:24:49.488199 [INFO] mod_dialplan_xml.c:558 Processing 1600 <1600>->1601 in context default 2013-06-28 14:24:49.488199 [ERR]mod_dptools.c:1351 No variable name specified. 2013-06-28 14:24:49.508248 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1601@192.168.104.191:5073 [e0a7b224-c48e-449d-b8ca-280ea016af07] 2013-06-28 14:24:49.528204 [NOTICE] sofia.c:6443 Channel [sofia/internal/sip:1601@192.168.104.191:5073] has been answered 2013-06-28 14:24:49.528204 [NOTICE] sofia_glue.c:4291 Pre-Answer sofia/internal/1600(a)192.168.104.191! 2013-06-28 14:24:49.528204 [NOTICE] switch_ivr_originate.c:3427 Channel [sofia/internal/1600(a)192.168.104.191] has been answered 2013-06-28 14:24:54.568203 [NOTICE] sofia.c:716 Hangup sofia/internal/1600(a)192.168.104.191 [CS_EXECUTE] [NORMAL_CLEARING] 2013-06-28 14:24:54.588237 [NOTICE] switch_ivr_bridge.c:721 Hangup sofia/internal/sip:1601@192.168.104.191:5073 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2013-06-28 14:24:54.588237 [ERR] mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE account_code='2'] 2013-06-28 14:24:54.588237 [ERR]mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE account_code='2'] 2013-06-28 14:24:54.588237 [INFO] mod_nibblebill.c:521 Beginning new billing on 51157e70-fb95-4c90-91f6-3575f097fe00 2013-06-28 14:24:54.588237 [CRIT]mod_nibblebill.c:558 Failed to log to database! 2013-06-28 14:24:54.588237 [ERR]mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM accounts WHERE account_code='1']

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I've been bored about no Audio(NAT, Firewall or ...)!
by Sayyed Mohammad Emami Razavi 29 Jun '13

29 Jun '13

I can not detect my solution! i have no audio on FS and i can not detect in which problem is! any suggestion?! anybody helps me?! RTP packets sent from FS don't reach my client! RTP packets saved with tcpdump in FS server after being decoded via WireShark play appropriate sound(ivr-demo) for me! My FS NET architecture: FS <> internet <> NAT <> Client (linphone, 3cx ...)

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Re: [Freeswitch-users] [Freeswitch-dev] Github and FreeSWITCH
by Brian West 29 Jun '13

29 Jun '13

I will investigate that... just wanted to make sure everyone was aware we use Jira and our wiki so we don't have fragmentation. On Jun 21, 2013, at 8:13 AM, Mathieu Parent <math.parent(a)gmail.com> wrote: > Hi, > > 2013/6/21 Brian West <brian(a)freeswitch.org>: >> Please use http://jira.freeswitch.org to publish patches. > > Then, why don't you simply remove the ability to request pull/fill > issues/update wiki? > > Regards > -- > Mathieu Parent

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Getting a lua script to run early
by Alex Lake 29 Jun '13

29 Jun '13

I want to run a little lua script early in the dialplan processing. I need to get in front of some of the X-PRE-PROCESS stuff, notably <X-PRE-PROCESS cmd="include" data="public/*.xml"/> Can I use "inline" to do this, I wonder? I've already put <X-PRE-PROCESS cmd="include" data="ddicheck.xml"/> in, but the extension it contains that runs some lua doesn't execute until after all the other main extensions have been considered for execution.

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Friday Free For All
by Ken Rice 28 Jun '13

28 Jun '13

sip:888@conference.freeswitch.org... Where is everyone? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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originate_disposition , hangup_cause - return empty
by Lloyd Aloysius 28 Jun '13

28 Jun '13

Hi All fs_cli shows the reason the for the bridge fails is *mod_dptools.c:3096 Originate Failed. Cause: NO_ANSWER* When I try to get the reason for the bridge fail from lua. All the following return empty or nil value local last_bridge_hangup_cause=session:getVariable("last_bridge_hangup_cause") local bridge_hangup_cause=session:getVariable("bridge_hangup_cause") local hangup_cause=session:getVariable("hangup_cause") local originate_disposition = session:getVariable("originate_disposition") Does this seem to be a bug? or I am accessing a wrong variable? Thanks Lloyd

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One way audio or no audio at all on Polycom after make current
by Emrah 28 Jun '13

28 Jun '13

Hi guys, Has anything changed recently on NAT traversal helpers? I have Polycom phones that used to work with inherit_codec=true, both for voice and video. After a make current, the video works consistently and the voice is either one way or inexistent. When it's one way, the caller does not hear the remote party. This seems to be related to Polycom phones only. Trying to use early media for a forced codec negotiation does not help. Please let me know if you have any idea or how to get more data to further troubleshoot this issue. Thanks!

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reading gateway configuration via mod_xml_curl
by mbo 28 Jun '13

28 Jun '13

I've configured freeswitch that it loads gateway configuration via mod_xml_curl. So far thats working fine. But I see an issue when I do changes to a gateway configuration, like changing registration credentials, etc. I can do this in my database and then perform "sofia profile external rescan" in the fs_cli. This forces to requery the data via mod_xml_curl, but I see the following warning in the cli: [WARNING] sofia.c:2840 Ignoring duplicate gateway 'gw1' So finally the changes are not applied. Is there an other way to force reload of the gateway configuration? Thanks Markus

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