October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Asterisk vs Freeswitch
by Ujjval Karihaloo 02 Oct '09

02 Oct '09

Is there benchmark test results on how many simultaneous calls Freeswtich can do (with RTP anchored through it) vs the Asterisk. For any hardware/CPU/Mem that anyone may have performed this performance testing. Any numbers on average how much Freeswitch scores over the Asterisk in terms of capacity will help.

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looking for qualified and cheap TISP
by Erwin Davis 02 Oct '09

02 Oct '09

Hi, I installed internal freeSWITCH in my LAN and want to see if I can dial out / dial in. Could anyone suggest one Telephone Service provider which is capable of connecting with FreeSWITCH and CHEAP/even FREE if possible? At this moment, I want to prove it is working with the real outside world. Thanks, e

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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
by Mike van Lammeren 02 Oct '09

02 Oct '09

On Mon, Sep 28, 2009 at 9:05 PM, "Even André Fiskvik" <grevenx(a)me.com > wrote: > From: "Even André Fiskvik" <grevenx(a)me.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 28 Sep 2009 22:52:13 +0200 > Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey > I have been working with a similar setup myself, but for some reason I > ended up ditching theUltraMonkey setup because I just couldn't get it to > work right. > > It's been quite a while since my effort, so I don't remember what the exact > issue was. > I got registrations to work, but had some other sip-dialog issues. > > We have since then changed over to running OpenSIPs as a loadbalancer in > front of > multiple FreeSWITCH instances. This setup is still in testing, but > seemlingy works fine > (and if it doesn't, it's my own fault for writing a bad opensips config). > > After we have done some more testing I can create a wiki-page with config > details. > > > Best regards, > Even André > > Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren

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Call Forward All/Busy/No-Answer
by Jerry Richards 02 Oct '09

02 Oct '09

How would I configure FS to Call Forward All or Call Forward when Busy or Call Forward when No-Answer? Can this be done at the server, rather than at the phone? Best Regards, Jerry

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FreeSWITCH Weekly Conference Starting, Please Call In!
by Michael Collins 02 Oct '09

02 Oct '09

Hey folks, the weekly conference call is starting. Please see the agenda for instructions on dialing: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_02 Looking forward to speaking with you all! -Michael

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Dialplan Issue
by Shelby Ramsey 02 Oct '09

02 Oct '09

Hello: I asked this on IRC yesterday and I think I confused everyone involved. So I apologize in advance here for reposting the question and if I wasted anyone's time. So here is the issue I'm having. I'm trying to use FS as a redirect server (specifically to serve up LNP queries via 302 redirects). But I'm having an issue where based on the string in the dialplan FS will respond with a 500 internal error message instead of a 300 redirect. The call flow should be this: -- remote party sends an Invite to my FS instance -- FS should respond with a 302 The following works as expected (FS will send a 302 when it receives an Invite): <action application="redirect" data="sip:${destination_number}@ ${network_addr};rn=${rn};npdi=yes"/> However if I do this (which is the way the response should look) FS will respond with a 500 internal server error: <action application="redirect" data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/> So the issue is the placement of the user params .... if they are before the @ FS will send a 500 internal server error ... if they are after the @ FS will send a 302. Unfortunately placing the user params after the @ doesn't quite conform to the way other devices expect to receive the 302 for this application. Any help would be greatly appreciated. Shelby PS ... hats off to the author of mod_memcache ... that is extremely useful!

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conference participant from behind NAT
by RobertT 02 Oct '09

02 Oct '09

I am a bit confused with what's going on in a following scenario. I have a public FS server with a public conference, that clients are connecting to with my softphone. All of this softphones have STUN option enabled and working, effectively resolving client's public IP address. They also have ICE enabled (but I guess it's not relevant here, since FS doesn't do ICE). Also, media trafic is secured with SRTP. The problem is when one client connects from port-restricted NAT into a conference he can hear sound for some time and he can be heard by other participants, but after awhile sound is gone and neither he hear anything nor he can be heard. Where is the problem? Is it NAT, closing RTP port after some silence period from client? I tried to start conference with waste flag, but without success eventually. The very same person can be contacted through this FS with direct call (being established in proxy_media mode) without any problems, but this is where ICE stuff starts doing its' "magic", I guess. Maybe I should try the same with SRTP disabled? Any help would be apreciated! Best regards, Robert.

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Minimum audio length for uuid_record
by Dan Le 02 Oct '09

02 Oct '09

We're running into a problem with the minimum file size when recording using uuid_record. It seems if the audio is too short it deletes the audio file. Is there a way to override that? We're running the FS 1.0.4 tag, on a windows xp 64bit machine. We start uuid_record from a lua script, recording at 8khz, if the recording is shorter than about 5s, the file gets blown away, but we may want to keep recordings that are 3-4s long. Thanks, Dan

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bgapi jobid to uuid
by Matt Riddell 01 Oct '09

01 Oct '09

Hi, I decided to go with a linked list for current channels and maintain that through state changes. So, basically it works like this: 1. Originate a call using bgapi (we get a jobid in response) 2. Receive an event with the jobid and a uuid 3. Lookup the linked list for the jobid, set the uuid 4. Receive hangup etc (with uuid) remove from linked list The problem is that sometimes I receive the hangup with the uuid before I'm told what the match between the jobid and the uuid are. Any ideas? -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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REGISTER fails with 407 after minutes of success register
by Fernando Testa 01 Oct '09

01 Oct '09

Hi all, I have a FS that registers on an Ericsson pabx as gateway under sip_external. This gateway start registering on the Ericsson ok, but after a while, around 50mins, it fails with the logs below. If I hit *sofia profile external restart* on fs_cli then the gateway returns to register with success (that means, we get 200 OK from Ericsson). This happens with FS 1.0.4 release tarball, and trunk r15011. I found similar situations on these links, but not actually found a solution. Any help is very welcome. *OS* CentOS 5.3 x86_64 4Gb RAM log at http://pastebin.freeswitch.org/10517 conf/sip_profiles/external/ericsson.xml: <gateway name="ericsson_1064"> <param name="username" value="1064"/> <param name="realm" value="10.227.0.3"/> <param name="password" value="5080"/> <param name="register" value="true"/> <param name="register-proxy" value="10.227.0.3"/> <param name="expire-seconds" value="60"/> <param name="retry-seconds" value="10"/> </gateway>

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