- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeTDM - any good for legacy CAS T1 protocols?
by Bob Hartwig 16 Jun '14

16 Jun '14

Hello all, I have a project coming up that will require me to interface Freeswitch to some rather arcane legacy T1 protocols. Specifically digital loop start, and station-side digital loop start. I'm concerned that the userland support for these may be incomplete or even non-existent in FreeTDM. I've successfully used FreeTDM to connect to PRI, but when I look for span configuration options for CAS T1, the only things that I see that may be applicable are described as "analog" options, and that doesn't sound quite right for loop start on T1 channels (but I could be wrong about that). If I were to use Asterisk for this, I would configure chan_dahdi.conf for " signalling=fxs_ls" and/or "signalling=fxo_ls". Should I pursue using FreeTDM for this, or would I be smarter to incorporate a simple Asterisk gateway to handle this legacy CAS T1 signaling? Thanks, Bob

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Re: [Freeswitch-users] Example of working with upstream distros...
by Brian West 16 Jun '14

16 Jun '14

Results https://github.com/joyent/pkgsrc/commit/df3a615de013424ac366e6e61e22031decd… -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 5, 2014, at 9:55 AM, Brian West <brian(a)freeswitch.org> wrote: > FreeSWITCHers, > I wanted to share a good example of working with an upstream distro on a problem that prevented building FreeSWITCH successfully without some LDFLAGS trickery. > > Unable to build on SmartOS, Patch attached to this jira https://jira.freeswitch.org/browse/FS-6375, The patch wasn't exactly correct so I did some digging. > > If you read the JIRA the links to the mailing list posts to smartos-discuss posts and responses, along with the root issue has been found and a really dirty work around till they update their pkgsrc to not include the 32bit path in 64bit packages so their .la files will work correctly. > > > -- > Brian West > brian(a)freeswitch.org > > <whmcslogo.png> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > >

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call rejection in simultaneous ringing
by Preeti Trivedi 16 Jun '14

16 Jun '14

Hi, I'm trying to ring my extension and mobile ring simultaneously and it worked. Now, I want to stop ringing the other phone when I reject the call from one phone, same as the other phone stops ringing when I answer in one phone. I tried few channel variables but I couldn't get that working. My dial string is as follows: dial_string = string.format("{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@ ${dialed_domain}}sofia/internal/1%s%s%s(a)xx.xx.x.xx ,[leg_delay_start=%d]${sofia_contact(${dialed_user}@${dialed_domain})} Thanks & regards, Trivedi

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DNS name as the public IP respond 403
by Shahzad Bhatti 16 Jun '14

16 Jun '14

Hi, i am trying to use DNS Name and for that i start my Freeswitch Using -nonat and in sip profile create a new profile with the following changes <param name="sip-port" value="8080"/> and <param name="ext-rtp-ip" value="host:sip.mydomain-name.com"/> rest conf is same as internal and when i try to register i got 403 response while on internal profile it works fine i am using FreeSWITCH Version 1.5.8b+git~20131202T224806Z~e127ca7d40~64bit (git e127ca7 2013-12-02 22:48:06Z 64bit) on centos 64-bit reply me what else i am missing regards Shahzad Bhatti

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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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