June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Need to deploy 5 inbound and 5 outbound
by Amin Dhamani Askhanuman 16 Jun '14

16 Jun '14

Hi, Currently we are using freeswitch with 1 inbound and 1 outbound number now we want to add 5 more inbound and outbound numbers to existing configuration and call from 5 inbound and outbound will route to certain extensions. Can you guide me how I can do that. -- Best regards, Amin Dhamani Amin Dhamani | System Administrator | +66 911167159 | Krits Tower, 1032/1-5, 14 Rama 4, Bangkok 10120 *www.askhanuman.co.th <http://www.askhanuman.co.th/>* | www.facebook.com/AskHanuman | amin(a)askhanuman.com This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________ -- This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________

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Voice Mail issue
by Amin Dhamani Askhanuman 16 Jun '14

16 Jun '14

Hi, Currently we are facing issue with voicemail. Sound quality is not good. Even caller can't hear the greeting properly. Can you guide how to resolve it. -- Best regards, Amin Dhamani Amin Dhamani | System Administrator | +66 911167159 | Krits Tower, 1032/1-5, 14 Rama 4, Bangkok 10120 *www.askhanuman.co.th <http://www.askhanuman.co.th/>* | www.facebook.com/AskHanuman | amin(a)askhanuman.com This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. _____________________________________________________________________________ -- This communication may contain information that is legally privileged, confidential or exempt from disclosure. If you are not the intended recipient, please note that any dissemination, distribution, or copying of this communication is strictly prohibited. Anyone who receives this message in error should notify the sender immediately by telephone or by return e-mail and delete this communication entirely from his or her computer. ______________________________________________________________________________

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Setting conference parameter regardless if it is already created or not
by Dmitriy Shumaev 16 Jun '14

16 Jun '14

Hi. I need to set a conference parameter - "caller_id_name". So in my dialplan I do next: <action application="set" data="api_result=${conference($1 set caller_id_name ${Orig_Ptt_Ext})}"/> <action application="set" data="api_result=${conference($1 set caller_id_number ${Orig_Ptt_Ext_UUID})}"/> <action application="answer"/> <action application="conference" data="$1@simple++flags{nomoh}"/> . But it is not right, because the conference possibly does not exist at that time (will be created at " conference "). Then I tried to do next: <action application="answer"/> <action application="conference" data="$1@simple++flags{nomoh}"/> <action application="set" data="api_result=${conference($1 set caller_id_name ${Orig_Ptt_Ext})}"/> <action application="set" data="api_result=${conference($1 set caller_id_number ${Orig_Ptt_Ext_UUID})}"/> But it is also not right, bacause dialplan stops at doing "conference". So, what am I suppose to do to make it right? May be there is some way to pre-create conference (without real members)? May be there is a way to configure some actions to be done after creation of conference - something like "execute_on_conference_create"? With best regards, Shumaev D.A., programmer @ KBR Ltd.

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HA and conference events
by Mike Kane 16 Jun '14

16 Jun '14

Hello all. Quick question. We successfully configured db functionality using Postgres to support HA, which is working quite well. The one anomaly we see is that when in a conference and the originator would like to add a member, originator sends a "Refer" to Fs conference server. That triggers FS conf to generate "Notify's" that are sent just after the "Refer" with a sipfrag 100/Trying and it's sipfrag 200/Ok response. This functionality works with sqlite, but not with Postgres. Any ideas on what we maybe missing? Below describes the messages flow. Member A joins FS conference Member A would like member B to join so member A sends a "Refer" to FS conference server requesting conference server to invite member B. FS conference server sends invite to member B FS conference server sends member A a "Notify" with sipfrag 100/Trying (this does not happen now) FS conference server sends member B and invite per the "Refer" sent to it (normal call flow for member B continues as he/she joins) FS conference server sends member A a "Notify" with sipfrag 200/Ok (this does not happen)

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Extend DTMF console logging
by Grant Bagdasarian 16 Jun '14

16 Jun '14

Hello, In switch_channel.c line: 471 there is a function call to log the received DTMF. I believe there is one in switch_rtp.c as well. Is it possible to extend this to also include on which channel the DTMF was received? There is a channel reference available in the function, so it could be extended, I'm just not sure how to since I don't know C :) if (!sensitive) { switch_log_printf(SWITCH_CHANNEL_CHANNEL_LOG(channel), SWITCH_LOG_DEBUG, "RECV DTMF %c:%d\n", new_dtmf.digit, new_dtmf.duration); } FS_cli currently only logs something like: switch_channel.c:471: RECV DTMF 1:200. It would be convenient to see something like: switch_channel.c:471 sofia/outbound/1000@10.10.10.10:5060<mailto:sofia/outbound/1000@10.10.10.10:5060> RECV DTMF 1:200 Can someone show me how to? Or provide patch? Maybe add it to the master branch? Thanks Regards, Grant

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Using Postgres in core and channels keep on piling up
by Muhammad Naseer Bhatti 16 Jun '14

16 Jun '14

Hi, I am using Postgres in the core as well as for mod_db because the limit subsystem needs to be shared with other hosts for mod_db <param name=“odbc-dsn” value="pgsql://hostaddr=remote_host port=15432 dbname=freeswitch user=freeswitch password=freeswitch options='-c client_min_messages=NOTICE'" /> and with switch.conf <param name=“core-db-dsn” value=“pgsql://hostaddr=remote_host port=15432 dbname=freeswitch user=freeswitch password=freeswitch options=‘-c client_min_messages=NOTICE’” /> DB host has a latency of 27 ms from FreeSWITCH server. Channels keep on piling up until the max_session limit is reached and the switch does not accept any more channels. Seems like the channel info is not being released from the database. Even now I have stopped sending any calls, the switch still shows 7000+ channels connected. Offcourse that’s the database only. But why this is happening? freeswitch@internal> show calls count 7260 total. freeswitch@internal> status UP 0 years, 0 days, 0 hours, 24 minutes, 54 seconds, 940 milliseconds, 636 microseconds FreeSWITCH (Version 1.4.6 git 9479729 2014-06-03 19:35:16Z 64bit) is ready 12884 session(s) since startup 0 session(s) - peak 10000, last 5min 0 0 session(s) per Sec out of max 500, peak 89, last 5min 0 10000 session(s) max min idle cpu 0.00/99.83 Current Stack Size/Max 240K/8192K show channels shows 0d01315c-ed00-11e3-a7c7-dfbcc3672627,inbound,2014-06-05 18:23:42,1402007022,sofia/Profile_250.218/vGriffins@88.208.219.118:5050,CS_EXECUTE,vGriffins,123456789,88.208.219.118,4414732506,,,RINGING,,,,,vBilling,,,,,,,,,,,,,,,,,,,,, But when I try to kill that uuid, freeswitch@internal> uuid_kill 0d01315c-ed00-11e3-a7c7-dfbcc3672627 -ERR No such channel! I am not sure what’s going on except something is not allowing for the channels to be cleared from the database. I also have tried rtp-timer-name =soft and session-timeout=10 with enable-timer=true, but that does not helps. What should I look for now? Thanks. Sent with Unibox

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BLF troubleshooting
by Nandy Dagondon 16 Jun '14

16 Jun '14

Hi guys, BLF on our receptionist phone no longer displaying properly on registration status of phones - they always display green even if the phones are not registered e.g. physically disconnected from the network. It should display orange. However, they're displaying on active calls - flashing red when ringing and solid red when answered. Can anyone give me an idea how to attack this problem? Just a hint. I see the databases sip_subscriptions, sip_registrations, sip_presence, sip_dialogs, etc. Which important Events to watch? Just a hint. Thanks. /Nandy

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Autopsy of GCC 4.6.3 compilation bug on Raspberry Pi (FS-6494 and FS-6524)
by William King 16 Jun '14

16 Jun '14

http://quentusrex.com/freeswitch-raspberrypi-gcc-compiler-bug.html -- William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king(a)quentustech.com

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Why do I get this odd *98 code in my dialplan?
by Sean Devoy 15 Jun '14

15 Jun '14

Hi All, My user has a CISCO SPA504G with the operator console. She is a receptionist and receives and transfer many calls a day just fine. Occasionally in the log file I see entries like this: Processing Front Desk <10>->*98 in context from-internal-ESTA Now the phone is configure with Vertical Service Activation Code *98 for Blind Transfer code. It may be something the user is doing incorrectly, but I have better luck getting information from a gold fish. Any ideas how this "code" is appearing as an extension? Sean

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callcenter_config agent / tiers reload
by Alvaro Mari-Thompson 14 Jun '14

14 Jun '14

Hi all, In my continuing quest to get some basic ring-all call queuing up and running, I've abandoned mod_fifo and started looking at mod_callcenter which easily does everything I need. The stumbling block I've had is about dynamic configuration reloading. Our freeswitch instances are configured to use xml_mod_curl to retrieve configs, as all our configuration is kept in a third-party system. In the CLI, I am able to call callcenter_config queue reload <name> which works as intended, but this will not reload the agents or tiers. Having tried both callcenter_config agent reload <agent> and callcenter_config tiers reload <queue> <agent>, I get an -ERR no reply from the console each time. I've looked through mod_callcenter.c and can quite clearly see in code that these should be both working, code example can be found here https://fisheye.freeswitch.org/changelog/freeswitch/?cs=9ecbb54 Is this a known issue? I can see JIRA issue on this matter at https://jira.freeswitch.org/browse/FS-5850 but it seems to indicate this was all complete towards the end of last year, which is why it's puzzling me that those commands are not working. Regards, Alvaro

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