Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Hello Takeshi, Thanks for your hint... it worked out... so to be precise: VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message. Finally, here is what i u

Re: [Freeswitch-users] Yet another question about A500 + FS

Thank you for you reply Mike! 'ignore_early_media=true' variable setting is the solution, but I figured it out shortly after posting to the ML. Best regards, V. Panayotov On Mon, Aug 24, 2009 at 10:46 PM, Michael Jerris wrote: > Do you have an answer in the dialplan for that extension? Also,

[Freeswitch-users] compile freeswitch, lua, luasql for 64-bit arch

Hi - Does anyone know how to compile freeswitch for a 64-bit architecture on mac os x? MySQL on MAC OS X only comes pre-compiled for 64-bit. I compiled luasql against this libmysqlclient.dylib. Now when I call lua from freeswitch I get: 2009-08-24 21:06:44.370439 [ERR] mod_lua.cpp:182 error lo

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

Windows installer does not work for me. I've reinstalled various times, same results. I can correctly create a number, but when I try to create a device for that number, it tells me that cannot locate the device, and the password for vicemail will be invalid. After that, it begins to give the ph

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi wrote: > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga wrote: >> >> sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l >> 4000 >> scenario file:  uac_

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga wrote: > > sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s > 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l > 4000 > scenario file:  uac_redirect.xml > FS dialplan:   public.xml > SIP trace: 

Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

Hi Raymond, I'm not planning to have more than 10 concurrent calls on this devices, but I'm also curious as you about how many calls can it handle. When I get to that point I will post the test results on this list. Regards, Rogelio On Aug 23, 2009, at 8:23 PM, Raymond Chandler wrote: Coul

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Is there any way to make FS complain about what header is wrong ? T. On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga wrote: > this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0 > > > > http://sipp.sourceforge.net/doc3.0/reference.html > > *[branch]* - Provide a branch value

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0 http://sipp.sourceforge.net/doc3.0/reference.html *[branch]* - Provide a branch value which is a concatenation of magic cookie (z9hG4bK) + call number + message index in scenario. An offset (like [branch-N]) can be appended i

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] shouldn't that be Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga wrote: > ... documentation hate that :)) ... but thats my life actually... thats > what i do f

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

... documentation hate that :)) ... but thats my life actually... thats what i do for living :) of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS. well ... my soul? .. is it really necessary? T. On

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace wrote: > It does the same thing, does anything get set during the install that > would remember or cache the old network settings? I can access anything from > the FS server on any of several networks and vice-versa but the SIP will not > register, aga

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l 4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log Here it is... sorry for not including at first... T.

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both. If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance? -- Anthony Minessale II F

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Hello Brian, Dave Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed branch within ACK message to meet INVITE's onebut it is still not enough... Also i checked RFC and this is how should it be ... (ACK without contact taking care to have correct TAGs and branch)... w

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

This is what I was asking! :D When the installer finished it started the whole thing and everything got loaded fine, but when I restarted my system it didn't (and did not anymore). Well, I will try to install everything from scratch again and see... On Mon, Aug 24, 2009 at 20:30, Brian West wrote

Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH

It is very nice to hear that, great work! Thanks to support FS. Telles Moises Silva wrote: So, I finally took some days to put up OpenR2 working with OpenZAP, which means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has support for. Including Mexico, Brazil, Argentina and other

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts. Doesn't mak

Re: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0

oh yeah, it's only the port they let you pick it looks like they don't let you pick ip in the abyss code, it would require an intrusive patch into that depend lib to allow you to set that. On Mon, Aug 24, 2009 at 4:17 PM, Peter P GMX wrote: > Any clue which one? I could not identify it, also l

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace wrote: > “Registration error: 408-Request Timeout” > Sorry for the the typo - 408 = timeout. (480 = temp unavail) Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there... Try:

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

"Registration error: 408-Request Timeout" Mike Peace Network Analyst EDCO, The Document People Direct 417-447-3367 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, August 24, 2009 4:18 PM To: fr

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace wrote: > I haven’t changed any of the conf files. > > What happens when you try to register? Do you get 480? (timeout) Or something else? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitc

Re: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0

Any clue which one? I could not identify it, also looked into the sources. Best regards Peter Anthony Minessale schrieb: > there is a configuration option in the xml file to control which ip it > binds to. > > > > On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX > wrote

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

I haven't changed any of the conf files. Mike Peace Network Analyst EDCO, The Document People Direct 417-447-3367 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, August 24, 2009 3:46 PM To: freeswitc

Re: [Freeswitch-users] Couple of questions

On 25/08/09 2:05 AM, Anthony Minessale wrote: > I updated testclient.c so you can see how now. Awesome awesome awesome! Thanks heaps man - everybody here has been great! -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily

[Freeswitch-users] pass-thru issue (G729)

Hello All, I'm having an issue with g.729 pass-thru. On an older 1.0.3 install it's all good, but on a newer machine, more cores, etc. the quality of voice on g.729 is very poor. Anyone here dealt with something like this? With all other codecs are perfect, even in pass-thru... Thanks in ddvance

Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Did you happen to hard code an IP in your vars.xml? /b On Aug 24, 2009, at 3:42 PM, Mike Peace wrote: I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Hi Tihomir - I'm no SIP guru, but the things which look suspicious about the ACK to me are: - Via header - different branch - Contact header - differs from INVITE --Dave > Hi Anthony, > > I'm aware it is generating 30 retries per a call and this is killing > me ... > > I lost my entire working

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp ;tag=[call_number] To: sut [peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Le

[Freeswitch-users] Moved Freeswitch to new subnet.

I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to a new subnet on out network that we have tons of boxes on and have all the routes already configured on and

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Hello Brian, it doesn't work .. tried this today as well: freeswi...@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 20:28:09.367300: INVITE sip:30003016094191...@10.4.4.251SIP/2.0 Via: SIP/2.0/UDP 10.

Re: [Freeswitch-users] Cannot create outgoing channel type [error]cause: [FACILITY_NOT_SUBSCRIBED]

Make sure your x-lite is not on port 5060 /b On Aug 24, 2009, at 3:25 PM, Jerry Richards wrote: > Yes. This a stand-alone Windows XP machine. > > Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.o

Re: [Freeswitch-users] Cannot create outgoing channel type [error]cause: [FACILITY_NOT_SUBSCRIBED]

Yes. This a stand-alone Windows XP machine. Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, August 24, 2009 12:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Cannot create outgoing channel type [error]cause: [FACILI

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

mr fritz is lying somewhere get a pcap of the traffic from fritz to FS and look at the size of the audio packets if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252) then it's 30ms if it's saying 20 but it means 30 you should leave the last change in place and also add in an

Re: [Freeswitch-users] Problem with cnam.js?

Every page on the wiki should be editable. If you don't already have an account, go to: http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup Mike On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote: I think there’s something wrong with the script at http://wiki.freeswitch.org/w

Re: [Freeswitch-users] Yet another question about A500 + FS

Do you have an answer in the dialplan for that extension? Also, check out the ignore_early_media variable. Mike On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote: > Hi, > > I managed to get our A500 running with FreeSWITCH 1.0.4 stable using > wanpipe 3.4.4 drivers. But now I have another

[Freeswitch-users] Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]

Hello All, I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine for the first time using the Getting Started Guide. I can register three lines (1000, 1001, and 1002), but when I attempt to call one phone to the other I hear the operator say: "The person at extension 1000

Re: [Freeswitch-users] Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]

Are you trying to test everything on the same machine? /b On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote: > Hello All, > > I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP > machine > for the first time using the Getting Started Guide. I can register > three > lines (

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

In your scenario you need to add [peer_tag_param] at the end of the to on the Ack. /b On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote: > > > > recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: > >

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

That is the remote sdp, not the local sdp. They are sending ptime 20, not us. Are they actually sending 20 ms packets or are they sending 30? MIke On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote: > Hello Anthony, > > I set p...@30i,p...@30i and I can see in the logs that PCMA is used. > Howev

[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Hi Anthony, I'm aware it is generating 30 retries per a call and this is killing me ... I lost my entire working day to figure out what is missing in the damn ACK message SIPp is sending back... ACK looks quite ok to me. pls can you help ? freeswi...@l01sipindir1> recv 573 bytes from udp/[10.4

Re: [Freeswitch-users] topology hiding leaking information in SDP

so problem solved , Thanks you anthony and Brian From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Monday, August 24, 2009 7:58:47 PM Subject: Re: [Freeswitch-users] topology hiding leaking information in SDP if you establish an audio path

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

If you installed FreePBX then it would be that softwares job to manage the sofia profiles... wouldn't it? /b On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote: > Actually I did that and it worked fine. I had the problem the SECOND > time I run FS and freepbx. And (@Brian) mod_sofia was lo

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

actually I meant "the SECOND time I *RAN* FS and freepbx." On Mon, Aug 24, 2009 at 20:24, Raffaele P. Guidi wrote: > Actually I did that and it worked fine. I had the problem the SECOND time I > run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were > not > > > On Sun, Aug 1

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

Actually I did that and it worked fine. I had the problem the SECOND time I run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were not On Sun, Aug 16, 2009 at 16:04, Carlos Talbot wrote: > When you configure FreePBX for the first time it wipes out the sip_profiles > directory

Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

Your ACK message must not be valid (dialog matching or something else) so every 1 call will generate 30 retries that are queued up in the sip stack. at 100cps you will be generating this problem 100 times per second and queue up countless unfinished dialogs thus eating up the cpu. On Mon, Au

Re: [Freeswitch-users] topology hiding leaking information in SDP

if you establish an audio path that is using the same codec there is no difference in performance as the codecs do not do anything when both sides are the same. On Mon, Aug 24, 2009 at 11:33 AM, Hound Dog wrote: > Brian, > > I set proxy_media to false ad now getting clean SDP without all the >

[Freeswitch-users] topology hiding leaking information in SDP

Brian, I set proxy_media to false ad now getting clean SDP without all the incoming data thank you this also means that the packets have to go all the way in and out of the codec chain ( instead of being taken in and thrown out with no processing ) , is that the case ? what is the perf

Re: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0

there is a configuration option in the xml file to control which ip it binds to. On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX wrote: > Hello, > > is there any chance to limit the listening ips of the xml-rpc server > (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)? > > Best regard

Re: [Freeswitch-users] topology hiding leaking information in SDP data

You can disable transcoding on the sofia profiles... see defaults. /b On Aug 24, 2009, at 10:28 AM, Vlasis Hatzistavrou (KTI) wrote: > But that would allow tell FS to also do transcoding in some cases, > correct? > > Is there a way to avoid transcoding and still build the required SDP > in >

Re: [Freeswitch-users] topology hiding leaking information in SDP data

But that would allow tell FS to also do transcoding in some cases, correct? Is there a way to avoid transcoding and still build the required SDP in a clean manner? Best regards, Vlasis Hatzistavrou. Brian West wrote: > You're using Proxy Media and the only clean way to do this is not use > pr

Re: [Freeswitch-users] topology hiding leaking information in SDP data

You're using Proxy Media and the only clean way to do this is not use proxy media that way a complete clean SDP will be generated for the B- Leg. /b On Aug 24, 2009, at 9:45 AM, Hound Dog wrote: is there a way to have FS build a clean SDP message ?

[Freeswitch-users] topology hiding leaking information in SDP data

carriers need topology hiding , its an important feature for both security and also to hide you business partners from each other freeSwitch talks about it and also does a good job in hiding the signalling topology there is however a hole in the SDP manipulation that I am trying to plug and

[Freeswitch-users] fifo question

Hi, Does anyone know the purpose of fifo_orbit_announce? When does fifo_orbit_announce get played? Thanks, JB ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCR

Re: [Freeswitch-users] CHANNEL_CREATE and full information

try looking for the channel_originate events which is fired as soon as any outbound call is mature enough to have variables. On Mon, Aug 24, 2009 at 12:57 AM, Mikhail Krivushin wrote: > I put some variable in originate command for identify created channel for > my app using event socket. > > Is

Re: [Freeswitch-users] Couple of questions

I updated testclient.c so you can see how now. #include #include #include int main(void) { esl_handle_t handle = {{0}}; esl_connect(&handle, "localhost", 8021, "ClueCon"); esl_send_recv(&handle, "api status\n\n"); if (handle.last_sr_event && handle.last_sr_event->body) {

Re: [Freeswitch-users] unable to download non-US sound files

No french files exist yet. ;) /b On Aug 24, 2009, at 1:00 AM, Juan Backson wrote: > Any idea what's wrong? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRI

[Freeswitch-users] CHANNEL_CREATE and full information

I put some variable in originate command for identify created channel for my app using event socket. Is any ability to get all channel variables without uuid_show on channel_create? Can I ask FS to take me all information on channel creating? Can this be in trunk, if it cannt be get without source

Re: [Freeswitch-users] Couple of questions

For api responses you have to use body. I should have reminded that as well, sorry... Jmesquita On 8/24/09, Matt Riddell wrote: > On 24/08/09 4:27 PM, João Mesquita wrote: >> >> I need to work a little bit more on that documentation as well.. I saw a >> few conflicts with the core documentation

Re: [Freeswitch-users] zrtp endpoints have different sas through fs1.0.4

Me too :) - Original Message - From: "Brian West" To: Sent: Monday, August 24, 2009 1:53 AM Subject: Re: [Freeswitch-users] zrtp endpoints have different sas through fs1.0.4 > Wish they would send me one for my E63 for testing... only been > working with zfone 3 so far. > > /b > > On

[Freeswitch-users] XML-RPC on different ip than 0.0.0.0

Hello, is there any chance to limit the listening ips of the xml-rpc server (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)? Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch

Re: [Freeswitch-users] Screaming monkeys on ext 5000

On Mon, Aug 24, 2009 at 5:31 AM, Brian West wrote: > It requires internet connectivity.  It calls a remote system to play > which is out of our control. > Yeah, I noted this too, since a couple weeks at least... Maybe let's Todd know it's monkeys are out of voice? -giovanni

Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86

Hi Vladimir, Did you get an update on this at all? Thanks vmorales wrote: > Hi Michal, > > Just checking in to see if you've been able to take a stab at this. > > Thanks, > Vladimir > > > -Original Message- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-u