June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

21 Jun '12

Is there a way such that if my script is "speak"ing to a user that they input DTMF and the speak is terminated immediately?

4 7

Rtp traffic
by Juan Pablo L. 21 Jun '12

21 Jun '12

Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft), so i m requesting the information to configure a gateway on the FS side for outbound and inbound, but they are requesting that the RTP traffic be sent to one IP (to the UMG) and the SIP traffic to a differentIP.I have been reading documentation but i have not found how to accomplish this. can someone please give me a hint ? is this possible ? anyone with experience integrating to a huawei msc?

2 1

302 redirect problem
by Miha 21 Jun '12

21 Jun '12

Hi, In sip_profiles I have set to manual redirect. When call goes in public dialplan and than phone redirect call, FS automatically send redirect call to contex redirect. Is this normal? How to set that call will go to internal dialplan? Thanks! 2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:410 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) State Change CS_REPORTING -> CS_DESTROY 2012-06-20 14:55:16.126100 [INFO] mod_dialplan_xml.c:485 Processing 0038618100100 <0038618100100>->*051357952 in context redirected* 2012-06-20 14:55:16.126100 [DEBUG] switch_core_session.c:1228 Send signal sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264 [BREAK] 2012-06-20 14:55:16.126100 [DEBUG] switch_core_session.c:1428 Session 2 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) Locked, Waiting on external entities 2012-06-20 14:55:16.126100 [NOTICE] switch_core_session.c:1446 Session 2 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) Ended 2012-06-20 14:55:16.126100 [NOTICE] switch_core_session.c:1448 Close Channel sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264 [CS_DESTROY] 2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:514 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) Callstate Change HANGUP -> DOWN 2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:517 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) Running State Change CS_DESTROY 2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:527 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) State DESTROY 2012-06-20 14:55:16.126100 [DEBUG] mod_sofia.c:374 sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264 SOFIA DESTROY 2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:86 sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264 Standard DESTROY 2012-06-20 14:55:16.126100 [DEBUG] switch_core_state_machine.c:527 (sofia/nat/sip:014310000.fs_kabelvoip2@xxx.xxx.xxx.xxx:1264) State DESTROY going to sleep 2012-06-20 14:55:16.126100 [WARNING] mod_dialplan_xml.c:515 *Context redirected not found*

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Debian Packages OR Salt Configuration Files
by Avi Marcus 21 Jun '12

21 Jun '12

Hi - I know the centos yum packages are being worked on... is there debian packages somewhere? Also, is anyone using salt (http://saltstack.org/) to admin their servers and/or freeswitch? It's a remote execution tool for a cluster + configuration version control. I've started using it and it seems pretty awesome! If someone has config files, freeswitch modules etc that they use with it that they'd be willing to share that would be great! -Avi

2 1

My First FreeSwitch OR Is this easy to do?
by Matt Neimeyer 20 Jun '12

20 Jun '12

I apologize for the length I'm not sure yet what is "relevant" My partner and I run a small business out of our home and we're getting big enough that I want to take advantage of some "modern phone technology" primarily we're interesting in an auto-attendant at this point but more will come later I'm sure. We have two analog lines coming into the house (provided via Verizon FiOS) and then in every room we have two line capable analog phones. In our FiOS settings I have both lines ring both lines simultaneously (to emulate a hunt group) so if someone is on line 1 the incoming call still rings audibly on line 2 instead of just buzzing call waiting. I will be getting a (hand me down) TDM1600P V2 card which, in its current configuration, has four FXO and four FXS ports. So my plan was to install FreeSwitch and "splice in-line" where the phone lines come out of the FiOS box. So two lines would go into the FreeSwitch Box and two "extensions" would come out the other side. Whenever I rewire the house I'll decide then whether to run a data connection to each "endpoint" and get SIP phones, add more FXS modules to the TDM1600P, some mix, or whatever but I'm starting small (I think). As far the auto-attendant goes 95% of what I need now is to filter "non-humans". So I would set up a simple menu "Press 1 for Tech Support, Press 2 for Billing, Etc)". This should filter autodialers that want us to update our Google listing and so on... but I assume we can still have all of the options (until we hire on more staff) ring both extensions. Assuming this lets unanswered calls go to voice mail... On our analog phones though I'd really love to modify the Caller ID to show which option they selected. So instead of "Bob Customer / 234-567-8901" it might show "TS - Bob Customer / 234-567-8901" if they selected Tech Support. If possible I'd also like to "fill gaps" in Caller ID from a database because a few of our customers only show the number and not a name. Short term, I'm currently planning on downloading the Blue.box ISO and using that as my start point. Medium term I'd like to set up email notification of voice mails. Long term I'd love to integrate Google Voice and/or Skype. Really long term I might need to "convert" to a multi-tenant setup for Personal, Company (now with Employees) and Side Project Company. I think that's it... Any advice? Comments? Matt

5 7

FreeSWITCH, OSX, Libtool, Macports
by Terry Barnum 20 Jun '12

20 Jun '12

I recently started investigating FS and am curious about what the specific problem is with Macports and FS. I've been using Macports for a couple years and find it to be a great, actively maintained package manager that uses /opt/local, making it a very reliable way to keep production software working through Apple system updates. I've found the Macports team to be very responsive so if there are specific things that I can pass along, it might get fixed. Is there a link or thread that documents what's incompatible with FS? Thanks, -Terry

3 13

Re: [Freeswitch-users] Users need a red flashy light thingy
by toddb＠toddbailey.net 20 Jun '12

20 Jun '12

3 2

Users need a red flashy light thingy
by Sean Devoy 20 Jun '12

20 Jun '12

Freeswitch is so much better than my customers deserve. I have the DialPlan set up nicely so they can dial *72 to forward all DID calls to their answering service and *73 to turn it off. Today she Dialed *72 to turn it off 4 different times, and since that is the code to turn it on --- it did NOT turn off. She was livid and sent me hate mail about how "terrible IP phones are and they never work ." I even consoled her saying how *72 and *73 have no inherent indicators to show which means on and off. I have offered the obvious solutions of SPEED DIAL buttons that are labeled "Enable" and "Disable". Now my question. Anyone have any ideas how to use a line button, or possibly MWI to indicate that lines are forwarded (a db value). MWI is clearly in use so line button is my preference, but if MWI had a flash function, etc???? I have unused line buttons - How can I make them show busy (like shared call appearance) from the switch? Maybe a dummy user/extension I do something clever with? If it matters the phones are Cisco 504G (4 line). Thanks in advance, Sean

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Re: [Freeswitch-users] Users need a red flashy light thingy
by toddb＠toddbailey.net 20 Jun '12

20 Jun '12

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Re: [Freeswitch-users] gateway registration using TLS
by Mi Ke 20 Jun '12

20 Jun '12

Dear Sean, It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;) Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ? Thanks / MiKe ----- Original Message ----- From: Sean Devoy Sent: 06/20/12 07:03 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] gateway registration using TLS Mi Ke, You should put your actual remote provider’s domain name or tip address in where it says “remote_ip_here” and I assume you have entered you actual username and password where it says “yyy” and “zzz”. It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/> Hope that helps, Sean From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Mi Ke *Sent:* Wednesday, June 20, 2012 11:46 AM *To:* Freeswitch Users *Subject:* [Freeswitch-users] gateway registration using TLS Hi All, Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config: <profiles> <profile name="external"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <gateways> <gateway name="xxx"> <param name="register" value="true"/> <param name="username" value="yyy"/> <param name="realm" value="remote_ip_here:5061"/> <param name="password" value="zzz"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> <param name="retry-seconds" value="60" /> </gateway> </gateways> <settings> ... <param name="context" value="external"/> <param name="sip-port" value="5060"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="contact-params" value="tport=tcp"/> <param name="tls-version" value="sslv23"/> <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/> <param name="tls-sip-port" value="5061"/> <param name="dialplan" value="XML"/> ... </settings> </profile> </profiles> But that does not work... tport_connected( 0x7f9b44014110): events CONNECTED tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 ) tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061 tport_vsend returned 600 tport_set_events(0x7f9b44014110): events IN tport(0x7f9b44014110): reset timer tport_wakeup(0x7f9b44014110): events IN HUP ERR tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil) nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061 nua(0x7f9b440196a0): event r_register 503 Service Unavailable Any ideas what I'm doing wrong? Thanks / MiKe

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