June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

t38modem without registration...
by Klaus Hochlehnert 13 Jun '09

13 Jun '09

Hi all, maybe someone can help me here. For some compatibility issues I need to use an older version of t38modem which doesn't support SIP registration. Incoming faxes work well, but I don't know how to set this up for outgoing faxes. So, how can I setup this with FS for outgoing faxes without SIP registration? Thanks, Klaus

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Re: [Freeswitch-users] No VoIP audio after upgrading to latest svn
by John Wehle 13 Jun '09

13 Jun '09

> I think possibly that the configs changed, specially the auto-nat stuff Yep ... a closer look at the packet trace showed FreeSWITCH settings the Contact as 10.10.10.1 instead of the actual IP address of the machine. > If you have modified those two files then I recommend looking at the new > default config versions of those two files and integrating your changes into > the new ones. Yep ... that's my SOP. Looking at the internal.xml supplied with the new FS I see: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> Once I commented out those entries everything worked fine. I'm kind of surprised that this default changed ... the older FS came with these commented out and worked fine in the simple configuration where the server and phones are on the same network segment. In any case my config has been adjusted, things are working, it's Friday, and I get to go home so life is good. :-) -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Remove voicemail prompts
by Mark Campbell-Smith 13 Jun '09

13 Jun '09

Hi! How can I configure voicemail so that I do not get the options such as "record your message at the tone" and "mark this message as urgent" Thanks!

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Parrot (mod_parrot) support in FreeSWITCH
by Diego Viola 13 Jun '09

13 Jun '09

Hi everyone, I just found this project which seems to be a new VM for running languages... the parrot project aims to create a virtual machine for Perl 6 and other dynamic languages. You can take a look at it here: http://www.parrot.org/ It already supports many different languages: http://www.parrot.org/languages And it has a Apache module: http://www.parrot.org/mod_parrot I wonder that makes it embeddable... would it be possible to create a mod_parrot for FreeSWITCH? Please take this only as a feedback and not as anything else. Diego

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Re: [Freeswitch-users] No VoIP audio after upgrading to latest svn
by John Wehle 13 Jun '09

13 Jun '09

Yet more information ... a packet trace of a openzap to Grandstream call shows: Source Destination Packet FreeSWITCH Grandstream SIP Request: INVITE ... Grandstream FreeSWITCH SIP Status: 100 Trying Grandstream FreeSWITCH SIP Status: 180 Ringing Grandstream FreeSWITCH SIP Status: 200, with session description FreeSWITCH Grandstream SIP Request: ACK ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... ... FreeSWITCH Grandstream RTP Payload type=ITU-T G.711 PCMU ... FreeSWITCH Grandstream SIP Request: BYE ... Grandstream FreeSWITCH SIP Status: 200 OK The interesting thing is I don't see the Grandstream attempt to send audio. Is there something that FreeSWITCH needs to say to the Grandstream in order for the phone to send audio? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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No VoIP audio after upgrading to latest svn
by John Wehle 13 Jun '09

13 Jun '09

Upgraded from Apr 3 svn to svn 13769. Calling from openzap to 9999 (music on hold) works. Calling from openzap to 9995 (5 sec echo test) works. Calling from openzap to vmail works. Calling from Grandstream to 9999 (music on hold) works. Calling from Grandstream to 9995 (5 sec echo test) doesn't work ... call goes through however silence is heard. Calling from Grandstream to vmail doesn't work ... call goes through however vmail disconnects apparently due to receiving silence. Calling from Grandstream to openzap doesn't work ... call goes through and the Grandstream can hear what is said on the openzap side, however openzap hears silence from the Grandstream. Calling from Grandstream to Grandstream doesn't work ... call goes through however both sides hear silence. Suggestions on how to proceed? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Re: [Freeswitch-users] No VoIP audio after upgrading to latest svn
by John Wehle 13 Jun '09

13 Jun '09

BTW: in all cases show channels says PCMU 8000 is being used for the read and well as write codec. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Need to hire an experienced FreeSwitch Developer
by Herb Levitin 12 Jun '09

12 Jun '09

I need to hire an experienced FreeSwitch developer to build a small chat bridge that supports VoIP and 1-4 PRI's of TDM. Please contact me at herb(a)powercom.com or (805)845-8906.

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RFC2833 double-digits
by Ben Jones 12 Jun '09

12 Jun '09

Hi all, We're running into a problem with rfc2833. Here's the situation: A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH. The FS console shows it receives the digits once. We send the call to a tier-1 carrier The digits are played read back, but often the first several digits are repeated, i.e., 'Sorry, 1, 1, 2, 2, 3, 4 is not a valid password' when the user entered '1234.' When I was testing this recently, the call was sent to a Cisco gateway, which the documentation* states has problems with rfc2833, just like Sonus gateways, although that should be fixed past revision 10744. When using inband, everything works great. We are using start_dtmf_generate in the outbound dialplan. DTMF to tier-1 providers is inband. My initial thought was that the tier-1 was receiving rfc2833 and inband, but it's not consistent. I'm sure more information will be requested of me to help troubleshoot, so please let me know. Any advice would be very much appreciated. *http://wiki.freeswitch.org/wiki/RTP_Issues -- Ben J. -- Support Engineer II Teliax, Inc - Denver, CO tips and faqs at help.teliax.com

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INFO: Important story you need to Digg and read right away
by Michael Collins 12 Jun '09

12 Jun '09

Gang, There have been crazy rumors flying around. Tony sets the record straight. Please go here now and digg this story: http://digg.com/software/Anthony_Minessale_of_FreeSWITCH_Discusses_Barracud… Thanks! -MC

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