- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Conference Call Today
by Michael Collins 27 Feb '13

27 Feb '13

Hello all. Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_27 We will have an update on the mod_ha_cluster discussion and then Ken Rice will be showing off his SwitchPi project. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

Re: [Freeswitch-users] E169 question
by Brian Foster 27 Feb '13

27 Feb '13

> Hi, > I was wondering if I could ask you some dongle questions. I've been > testing out Gsmopen with an E160. Do you have any experience with > that one and why do you prefer E169s? I personally have not. They were, however, recommended to me by a friend who has experience with Huawei-brand dongles. > I'm in NYC. I might be > interested in a group buy, but I'm still trying to figure out if it's > going to reliable for clients. I currently cannot verify anything about the Huawei devices. The group buy actually fell through, and I was planning on getting one or two to play with. If others are interested in doing a group buy I'd be more than happy to facilitate. In your setup, what is your general purpose for having them in production? I planned on using a few for emergency backup in case the Internet or something else went kaput. With the advent of cell phones these days, it's really not an issue now. Otherwise, to me it's a toy. Now if you we're in Europe, there are advantages to using gsmopen and a dongle with a few different mobile providers to get free or reduced rate calling (within the same network). I'm CC'ing the Freeswitch users mailing list, in hopes that someone can better answer your questions. If you haven't already, subscribe to the ML by going to http://lists.freeswitch.org/mailman/listinfo/freeswitch-users. Good luck in your future endeavors, and welcome to the wonderful world of Freeswitch. Regards, Brian Foster Sent from my iPhone > I'm waiting on an antenna at the > moment....I suspect that's my current problem. > > > Thanks in advance! > > Cheers, > > > Peter Hartmann > Hartmann Computer Consulting

2 1

new site, cant call OUT!
by Sean Devoy 27 Feb '13

27 Feb '13

Hi All, (Git head build last week on centos 5.n) This is a new one for me. Calls IN to the new customer with Cisco phones behind NAT are working, but not outbound! I have not been to the site yet. I am told that they can call out once or twice then have to reboot the phone to dial out. Dialing in continues to work! one person says the phone says "Call merged" another says it say "Request merged" on the phone display! I did see in the logs, that when it works, we get: 2013-02-12 11:09:26.383576 [NOTICE] switch_channel.c:968 New Channel sofia/external/105@<some.domain.com> [51884b2d-6aa2-47fe-b687-8604a4e39a7a] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@<some.domain.com> [BREAK] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@<some.domain.com> [BREAK] 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:415 (sofia/external/105@<some.domain.com>) Running State Change CS_NEW 2013-02-12 11:09:26.383576 [DEBUG] switch_core_state_machine.c:433 (sofia/external/105@<some.domain.com>) State NEW 2013-02-12 11:09:26.413581 [DEBUG] switch_core_session.c:975 Send signal sofia/external/105@<some.domain.com> [BREAK] 2013-02-12 11:09:26.413581 [DEBUG] sofia.c:1750 detaching session 51884b2d-6aa2-47fe-b687-8604a4e39a7a 2013-02-12 11:09:26.413581 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'external' for [3014040914@<some.domain.com>] from ip <phones router ip> 2013-02-12 11:09:26.493574 [DEBUG] sofia.c:1842 Re-attaching to session 51884b2d-6aa2-47fe-b687-8604a4e39a7a And when it fails: 2013-02-12 13:32:26.363343 [NOTICE] switch_channel.c:968 New Channel sofia/external/220@<some.domain.com> [91e9eb64-253c-4604-bdfa-4dd18baae18d] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@<some.domain.com> [BREAK] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@<some.domain.com> [BREAK] 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:415 (sofia/external/220@<some.domain.com>) Running State Change CS_NEW 2013-02-12 13:32:26.363343 [DEBUG] switch_core_state_machine.c:433 (sofia/external/220@<some.domain.com>) State NEW 2013-02-12 13:32:26.383344 [DEBUG] switch_core_session.c:975 Send signal sofia/external/220@<some.domain.com> [BREAK] 2013-02-12 13:32:26.383344 [DEBUG] sofia.c:1750 detaching session 91e9eb64-253c-4604-bdfa-4dd18baae18d 2013-02-12 13:32:26.383344 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'external' for [4104934408@<some.domain.com>] from ip <phones router ip>2 2013-02-12 13:32:36.363085 [WARNING] switch_core_state_machine.c:514 91e9eb64-253c-4604-bdfa-4dd18baae18d sofia/external/220@<some.domain.com> Abandoned 2013-02-12 13:32:36.363085 [DEBUG] switch_channel.c:2994 (sofia/external/220@<some.domain.com>) Callstate Change DOWN -> HANGUP Can anyone shed light on what's happening "at that particular juncture" and how I may go about debugging it? I was thinking FS was not finding a connection through NAT to phone, but then why do incoming calls work? Thanks in advance, Sean

4 4

t38 200OK timeout
by royj 27 Feb '13

27 Feb '13

I wonder if anyone faced the issue that posted on jira - http://jira.freeswitch.org/browse/FS-5120 I understand employment of developers and hope Anthony or somebody else will give the time soon, but may be here would be any thoughts. -- regards royj

3 2

Fax document number of pages
by Tihomir Culjaga 27 Feb '13

27 Feb '13

hello, i know its stupid, but please, is there a way to somehow print the number of pages on a document in a <pages>/<total_pages> format instead of p.<pages> ? i cannot find this in mod_spandsp so i assume its somewhere in spandsp library... any idea where ? regards, Tihomir.

3 5

sending multiple voicemail attachments to different emails
by Jason Holden 26 Feb '13

26 Feb '13

Hi, Tried the recommendation from the wiki for setting up the information in the directory and separating entries by a , but this does not work.

2 1

dialplan action when sending a REFER to FreeSWITCH
by Hussein Said 26 Feb '13

26 Feb '13

Hi, I have an application that routes a call to an ad-hoc conference. That application then hunts an extension to answer the call by sending a REFER to FreeSWITCH which will send an INVITE from the conference to join. I need a way to make the "hunted" extension the moderator so that it can control the conference. I did set a moderator controls profile and can see it works if I change the dial plan(public.xml) to set the "member-flag=moderator". Also, "mod-wait|rfc4579" are set in the conference.conf.xml. My challenge is that when I hunt for an extension and find one, I send a REFER to FREESwitch so that it sends an INVITE to the extension that needs to answer the call, and I could not find a variable that worked in the dialplan that I can use as a condition to set the "member-flag=moderator" for the channel that is about to join the conference. Is what I am attempting feasible? Thanks

2 1

Possible BUG in G729 negotiation
by Flavio Goncalves 26 Feb '13

26 Feb '13

Hi, I have searched the list, but I couldn't find a final answer for this case. I'm receiving calls from a provider using G729 annex B. INVITE a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:19 CN/8000. and FS is answering 200OK m=audio 31388 RTP/AVP 18 101 19. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:19 CN/8000. a=ptime:20. If freeswitch supports only G729A, shouldn't FS refuse the call with 488 or suppress the line annexb=yes? The audio quality is really bad when I receive calls in this way. After removing the annexb on the gateway the audio is fine, but unfortunately I can't remove from it from all VoIP providers terminating calls. For outbound calls I simply removed the annexb, but for inbound there is no way to do it. Flavio

3 6

user and extension creation
by Venkateshwaran Thirugnanam 26 Feb '13

26 Feb '13

Hi All, I had installed freeswitch,I need to create a user and extension to test the connection,Please guild me to create users and extension for them to make a call to their extension... Regards, Kumaran T

4 4

FreeSWITCH Weekly News and Notes
by Michael Collins 26 Feb '13

26 Feb '13

Greetings! First item in the news today is a happy report from long-time FreeSWITCH user Henry Gavin. Henry runs a company in the U.K. called SureVoIP<http://www.surevoip.co.uk>. He is pleased to report that "thanks to FreeSWITCH and FusionPBX" his company is once again a finalist for the annual ITSPA awards<http://www.surevoip.co.uk/blog/2013/02/25/surevoip-and-the-surevoip-api-fin…>. Congrats to Henry for leveraging FreeSWITCH in a successful business endeavor. Another annual event is the Google Summer of Code<http://www.google-melange.com/gsoc/homepage/google/gsoc2013>(GSoC). FreeSWITCH will once again apply as a mentoring organization. Please start thinking of project ideas that we can include in our organization's application. Applications will be submitted starting March 18 and no later than March 29. Ken Rice and I will be coordinating this process. Stay tuned for more details. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_20>we did a nice tour of the CudaTel <http://www.cudatel.com> Communication Server. In the coming weeks we will have more presentations for GUI front-ends that community members have built. On this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_02_27>we will have Ken Rice give us an update on his new project: SwitchPi <http://switchpi.org>. If you like DIY projects then you'll appreciate what Ken has done with integrating the Raspberry Pi with FreeSWITCH and some other items to create something new. We look forward to seeing it in action. Don't forget about the FreeSWITCH HA discussion on Tuesday evening at 8PM EST. Last week's discussion was very fruitful. Eliot Gable gave us all a lot of information about the different approaches that he can take for building mod_ha_cluster <http://wiki.freeswitch.org/wiki/Mod_ha_cluster>. We look forward to his report on the potential of using OpenMPI<http://www.open-mpi.org/>. For those who can't make it to the HA discussion please join weekly conference call on Wednesday where we will have a brief recap of the HA call. In ClueCon <http://www.cluecon.com>news we have uploaded two new videos: * What's new in sipXecs 4.6 <http://youtu.be/mz14UqXRQDU> - Douglas Hubler * Challenges and Opportunities in Open Source VoIP<http://youtu.be/QnDYE-DfjgU>- Travis Cross Stay tuned for more ClueCon 2012 videos and ClueCon 2013 announcements. Have a great week and we look forward to talking to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

3 2