- FreeSWITCH-users - FreeSWITCH Mailing Lists

Gateway responding to random port (rport problem?)
by Nicolas Brenner 19 Oct '10

19 Oct '10

Hello everyone, I'm having an issue with a VoIP provider where the gateway will respond to any random port instead of 5080. I sent the corresponding trace to the provider and they told me to "disable rport and nat". I removed the auto-nat parameters in the corresponding sip profile, but I keep having the same issue (I don't think it has anything to do with nat anyway). On the trace I noticed that INVITEs sent by FS have an empty rport parameter on the Via header, but according to what I've read, this is standard and it indicates that the response should be sent to the originating port (5080), which is not being done by the gateway ("Avaya SIP Enablement Services"). How may I go about solving this? should I set the rport parameter to something specific (eg. 5080)? or remove it altogether? Thanks, Nico

2 3

Are there any rpath based FS distribution?
by Fabio Pietrosanti (naif) 19 Oct '10

19 Oct '10

Hi all, does anyone know if there is some rpath (www.rpath.com) based FS distribution that can be easily installed as an appliance by using a cdrom? Fabio

1 0

FreeSwitch as SBC, no registration?
by joedotslash 19 Oct '10

19 Oct '10

Hello all, I'm looking to implement the following scenario: Step 1 SIP Server A sends INVITE to port 5060 over IPv6 to FreeSwitch: 2001::1 --udp/tcp--> 2001::2:5060 Step 2 FreeSwitch SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000 1.1.1.1 --udp/tcp--> 1.1.1.2:6000 Step 3 All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path. REGISTER messages WILL NOT be processed by FreeSwitch. They are handle elsewhere. There is more to this but I want to accomplish this first call flow go from there. Is this possible with FreeSwitch? Can someone point me toward a resource on how to get this created? The following looks like a good start but it seems it's setup for users to register. I would prefer not to have this. http://wiki.freeswitch.org/wiki/SBC_Setup Thanks! Joe

3 5

Re: [Freeswitch-users] Compile on Windows
by Tim B 19 Oct '10

19 Oct '10

Thanks guys... I have tried building... Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz and git clone with autocrlf=false still getting errors and failed builds.... ========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ========== Now is there an specifics I need to set in VS2010 for my c++ options, etc? I am a c# developer so I don't do much with c++, so I never messed with any of the options there. I did download the windows 7 sdk so I would have all the windows header files. Tim

6 6

Limit [ERROR] question
by DJB International 19 Oct '10

19 Oct '10

I have noticed once in a while in the log file with the following error: [ERR] switch_limit.c:86 Unset limit backendlist! What exactly would cause this error? Thank you, Dorn B.

2 5

Sofia gateway incoming problem
by Maxim Balabaev 19 Oct '10

19 Oct '10

I`m having really strange problem. I connect new provider, changed all gateway settings but i can`t take calls. While debuggins external and so as global no invite packet is showing up. I even tryed tcpdump - absolutely nothing. Just pings. Outcoming are ok, register is ok. If I switch to asterisk - no problem occures. Support says that server refuses connection, while another gateway works perfectly. I don`t know what to do... May be this can be usefull to. Prov have multiple realms on server and accepts only @myprov.com format. While to asterisk invite is thrown with sipid@myip. I tryed changing server hostname to ip, but still nothing. Here is my conf: http://pastebin.com/qKqAv38a <http://pastebin.com/mMrj33pN> Really appritiate any help.

2 2

Conference leg recording
by Scott Brown 19 Oct '10

19 Oct '10

I'm wondering if there currently is a way to record one leg of a conference call. I have tried several different methods, but always seem to get the entire conference. Thanks! Scott

2 2

digium board
by Nikolay Kondratyev 19 Oct '10

19 Oct '10

Hello all! Can anybody please advise if Digium TE121 board is supported by FS? In the wiki i have seen pages devoted to Sangoma. What will be right choice for E1 PRI interface? Sangoma or Digium? Or may be something else? Thanks in advance, Nikolay.

6 8

nibble to update DB after hangup
by Muhammad Naseer Bhatti 19 Oct '10

19 Oct '10

Hello, I am using nibble_bill to bill a few calls. Trying to implement a scenario where I can update the database or CDR? after the call has been billed. Could not find this info elsewhere. Is there a way we can achieve this with nibble or do I have to run a script, to check balance before and after the call, and right after the hangup update the DB with new info? -B

2 1

FreeSWITCH HA question&talk
by Seven Du 19 Oct '10

19 Oct '10

Hi, I tested FS HA, it works like a charm. however, I'd like to talk about some more: I tested on my local computer by enabling tack-calls. make a call and kill FS, restart FS and execute sofia recover. 1) originate user/1000 &record(/tmp/blah.wav) Recover works, however, it overwrote the recording. Added RECORD_APPEND=true made it work. But for other APPs like socket, it might make a new socket to a outbound socket, can it send a var like channel_recovered so a outbound socket can aware of a recover? 2) call from 1000 to 1001 Recover works. If I also do uuid_record, the recording cannot recover. Seems there are no infos in sip_recovery table. Neither does record_session, though there's an entry in sip_recovery table. I guess it might be possible to get this work. If it can send a channel_recovered event, then it might be possible to continue the record in outside logic(say, event socket). 3) call from 1000 to a gateway outside (NAT-ed), leg-a can recover, leg-b failed with: 2010-10-16 10:28:17.162241 [ERR] sofia_glue.c:3190 AUDIO RTP REPORTS ERROR: [Bind Error!] I guess it is caused by NAT. (since I kill & restart FS manually, it may take 10 seconds). Is it possible to change the router settings? Or could FS send a re-Invite to re-establish media? Below are some fantastic thoughts: 4) N + M redundancy Default FS HA is designed to 1+1 redundancy, it might be interesting for N(active)+M(standby) in large farms. If one active FS fails, bind the float IP to a standby server and go. 5) N (N>2) servers without float IP Let's say each client UA register to 2 (or more) servers, If one server fails, a FS controller then re-call the client in an active call through a pair-server and the client responsible to hangup the originate call and automatically answer the new one. This need client support of course. :) -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

5 7