Re: [Freeswitch-users] tcp call misses sip message

but FS does use tcp for that call leg -> RX 1167 bytes ... from *tcp* ...: And after all there can be other SIP transports combinations FS should interconnect... ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswit

Re: [Freeswitch-users] Configuring freeswitch with voicepulse

Hi Jason Thanks for your response, I setup the configuration with 2 proxies based on the example of the freeswitch wiki. I looked at freeswitch.log and found the following line. Dialplan: sofia/internal/1...@74.207.249.79 Action set(effective_caller_id_number=1222333) Dialplan: sofia/interna

Re: [Freeswitch-users] suggestions for hardware.

On Thu, Nov 12, Orien Love wrote: > Since I have not had any replies about the atom board I am guessing > that nobody has used one, Could somebody tell me what is a good CPU > speed / Memory / FSB be? > I really do not have a large budget and cannot afford to buy > something that will no

Re: [Freeswitch-users] suggestions for hardware.

Thank you Dana and Michael for your replies, I am getting a spa3000 in the mail soon so I can try it out and see if it will work for my needs, I am going to implement a automatic attendant thanks to the information provided. Since I have not had any replies about the atom board I am gu

Re: [Freeswitch-users] Large number of destinations

Or, of course, there is always mod_xml_curl. Basically, XML dialplan on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a web application server, web application server responds with XML routing response, FreeSWITCH routes the call. On Thu, Nov 12, 2009 at 5:53 PM, Rupa Schomaker

Re: [Freeswitch-users] mod event socket

What exactly are you typing when you connect? Also, which version of FS? -MC On Thu, Nov 12, 2009 at 2:32 PM, srinivasula reddy < srinivas.ksvre...@gmail.com> wrote: > > > > HI all, > > i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 > port, when i am trying to connect free

Re: [Freeswitch-users] CDR for Failed Calls

enable the b leg logging On Thu, Nov 12, 2009 at 3:19 PM, wrote: > I am using xml_cdr to generate CDR results from FreeSWITCH servers, and > I've noticed that failed call attempts are not showing up in the results. > > Whereas the failed attempt is showing up in the Master.csv file. > For exampl

Re: [Freeswitch-users] Large number of destinations

On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij wrote: > On 11/12/09 9:59 PM, Rupa Schomaker wrote: > If I read it right, this is suited for "complete" nrs. So would I have a > system connected with lots of DIDs, I would put them in easyroute. Then > for systems with lots of number ranges, I would us

Re: [Freeswitch-users] tcp call misses sip message

tack on a ;transport=tcp /b On Nov 12, 2009, at 4:27 PM, RobertT wrote: > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.free

Re: [Freeswitch-users] Polycom SoundPoint IP501

I was trying to configure it just on the phone itself, but apparently even though it says Auth. User on the phone setting, it doesn't actually set the auth username according to the web interface. After using the web interface to configure the phone it works now. Thank you for your responses.

Re: [Freeswitch-users] Large number of destinations

On 11/12/09 9:59 PM, Rupa Schomaker wrote: Hi! > Take a look at mod_easyroute. Cool, I remember "quick-reading" about that module and thinking "nah, not needed". Then when the plan changed and I needed the large amount of routes it didn't struck me that easyroute is what I need for what I want t

[Freeswitch-users] mod event socket

HI all, i have connected Freeswtich(mod event socket) through telnet(tcp) 8021 port, when i am trying to connect freeswtich it it taking 20 seconds to get response from FS, can i able to reduce tcp response time? thanks Srinivasula Reddy K ___ FreeSWITC

[Freeswitch-users] tcp call misses sip message

Hello everyone! I'v got strange problem with incomplete call via tcp transport. When I perform bridged call from one ua (no matter what transport udp or tcp) through FS this call's leg b message sequence (over tcp) lacks finishing SIP message what in it's turn cause the call to be disconnected by t

[Freeswitch-users] CDR for Failed Calls

I am using xml_cdr to generate CDR results from FreeSWITCH servers, and I've noticed that failed call attempts are not showing up in the results. Whereas the failed attempt is showing up in the Master.csv file. For example, I've initiated some outbound calls that show up in the Master.csv as "RECO

Re: [Freeswitch-users] Large number of destinations

Take a look at mod_easyroute. On Thu, Nov 12, 2009 at 1:14 PM, Robin Vleij wrote: > Hi all, > > I'm currently building a proof-of-concept box using Freeswitch. Coming > from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. > > The plan is to make some sort of SIP router, some wo

Re: [Freeswitch-users] Polycom SoundPoint IP501

I am using Polycoms (430 and 501) with FreeSwitch. How do you provision them? Via WEB or config files? If you use config files than I can send you some sample files. Regards, __Yehavi: On Nov 12, 2009, at 11:41 AM, Adam Ford wrote: > > > Has anyone used a Polycom Sou

[Freeswitch-users] Large number of destinations

Hi all, I'm currently building a proof-of-concept box using Freeswitch. Coming from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. The plan is to make some sort of SIP router, some would call it an SBC I guess. There will be no PBX stuff, just gateways that talk to each other.

Re: [Freeswitch-users] Calls per second on FreeSWITCH

Matt, Thank you so much! bgapi did the trick. - Tina > Tina, > > How are you originating the calls? from the console? Try bgapi > originate... > > --matt > Voice Broadcasting - http://www.hellohunter.com/voice_blast.php > > On Fri, Nov 13, 2009 at 12:57 AM, wrote: > >> I'm trying to increase

Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK)

Remember you have a plain old regular analog connection between the FXO port of the SPA and your "phone line". The FXO circuit is just an analog switch (open or closed) if no one answers on the IP side and the person on the PSTN side hangs up, then the FXO side should sense a change in the polari

Re: [Freeswitch-users] Calls per second on FreeSWITCH

Tina, How are you originating the calls? from the console? Try bgapi originate... --matt Voice Broadcasting - http://www.hellohunter.com/voice_blast.php On Fri, Nov 13, 2009 at 12:57 AM, wrote: > I'm trying to increase the number of calls per second that I can originate > from FreeSWITCH, but

[Freeswitch-users] Calls per second on FreeSWITCH

I'm trying to increase the number of calls per second that I can originate from FreeSWITCH, but I cannot seem to get more than two-per-second. (I am trying to use FS to initiate thousands of calls quickly) switch.conf.xml I beefed up the max-sessions and sessions-per-second in the switch.conf.xml

Re: [Freeswitch-users] Polycom SoundPoint IP501

Not sure what do you have in your config file for the polycom exactly? btw you hijacked the Cisco Presence thread by clicking reply.. and changing the subject please don't do that in the future. Click new message and input the address for the list. Thanks, Brian On Nov 12, 2009, at 11:41

Re: [Freeswitch-users] Cisco 79x1 & Presence

They do it in their own weird way... if you wanna track it down I know their are examples of it out there. /b On Nov 12, 2009, at 8:25 AM, Peder wrote: > Has anybody every figured out how to get presence working on a Cisco > 79x1 w/ > FreeSWITCH? I spent quite a bit of time 6+ months ago on

[Freeswitch-users] Polycom SoundPoint IP501

Has anyone used a Polycom SoundPoint IP501 or similar hard phone with FreeSWITCH? I configured one to register with my FreeSWITCH server using one of the default sip profiles to test and I get "[DEBUG] sofia_reg.c:1688 SIP username 1001 does not match auth username" in the log file and the phone do

Re: [Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number"

Take a look at the freeswitch debug log, it should tell you exactly why it hung up. Mike On Nov 12, 2009, at 10:01 AM, Lei Tang wrote: > Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal > sip endpoint of FS. > I added two dialplan in public dialplan xml file. as flo

Re: [Freeswitch-users] att_xfer and Loopback

if you provide a console trace of both situations with console loglevel debug and put them on pastebin i can tell you what's happening. On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX wrote: > Thanks Anthony, > > however this rather deteriorated the situation. > Now it works the following > - A ca

Re: [Freeswitch-users] Does OpenZap support CTR21?

Russell.Mosemann wrote: > Yes, it should just work. I'd recommend Dahdi (complete), because Zaptel > is not being developed anymore. Thanks for the links. Turns out this card seems incompatible with the motherboard I have, so I'll concentrate on the Linksys 3102 instead. -- View this message i

[Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text="Unallocated (unassigned) number"

Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. I added two dialplan in public dialplan xml file. as flow: Every thing is ok when call to number 8. but when I call the second number "*114", fs

[Freeswitch-users] Cisco 79x1 & Presence

Has anybody every figured out how to get presence working on a Cisco 79x1 w/ FreeSWITCH? I spent quite a bit of time 6+ months ago on it and could never get it to work. Peder ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http

Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk?

This is just basic freeswitch dialplan concepts. It has nothing to do specifically with gtalk. Seems like you need to step back and do some more reading on the dialplan. ;) /b On Nov 12, 2009, at 7:02 AM, David Schwartz wrote: What I am looking for is hard coding a number (e.g. 1010) tha

Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk?

Thanks I overlooked that :) D. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Milena Sent: Thursday, November 12, 2009 3:36 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freesw

Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK)

Thanks for the pointers - I'll head off there now.. regards Dave - Original Message - From: "Fred-145" To: Sent: Thursday, November 12, 2009 12:59 PM Subject: Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK) > > > Dave Stevenson-4 wrote: >> Has anyone had similar pr

Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk?

Hello, Obviously it is possible, next time try to search better, the answer is on the same blog Mark pointed you too: http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/

Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk?

Thanks Mark I read this and didn't find a dialplan (do I need one?) to make calls into gtalk? I mean how would I even dial in? via URI (e.g. some...@gmail.com)? Wouldn't that just send the call to gmail? What I am looking for is hard coding a number (e.g. 1010) that would enable me to call it

Re: [Freeswitch-users] SPA3102 Won't drop the PSTN line (UK)

Dave Stevenson-4 wrote: > Has anyone had similar problems with the SPA3102 or has any ideas where I > can look to get to the bottom of the problem. (I have just upgraded the > SPA3102 to the latest 5.1.0 firmware) Before investigating further, you might want to ask in those forums to check that

Re: [Freeswitch-users] cd-sounds vs. sounds?

mercutioviz wrote: > I believe that French and Spanish sounds are in the works by the > community. > The only other sounds I'm aware of are the Russian ones. Thanks for the tip. -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26318115.html Sent from the

Re: [Freeswitch-users] Displaying caller ID on LED?

Mitch Capper wrote: > I did something like this recently. Thanks for the feedback. I'll see how Linux can be made to send stuff to a USB display. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26318100.html Sent from the Freeswitch-users mailing

Re: [Freeswitch-users] How to pick up someone's phone remotely.

Thank You for such an elegant and simple solution that I have not thought about. With an exception that I'm using FS 1.0.4 right now and it appears that something changed in time and following line should use hash instead of db (when using default 1.0.4 FS config): . After a few hours of experi

Re: [Freeswitch-users] Can I use mod_dingaling to call INTO gtalk?

Check this page out... maybe the info should be put on the wiki... http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ On Thu, Nov 12, 2009 at 9:03 PM, David Schwartz wrote: > All of the example I see allow me to call FROM gtalk. > > > > Help? > > > > Thanks, > > > > David > > _

[Freeswitch-users] Can I use mod_dingaling to call INTO gtalk?

All of the example I see allow me to call FROM gtalk. Help? Thanks, David ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mail

Re: [Freeswitch-users] att_xfer and Loopback

Thanks Anthony, however this rather deteriorated the situation. Now it works the following - A calls B - B enters *4 gets an announcement and enters digits for C (A get MOH) - C is called - As soon as C picks up the call, A and C both have no voice (and B is dropped) - When A hangs up, C hangs up