Re: [Freeswitch-users] Choosing a Codec.

On the other hand, a u-law WAV turned into L16 and then back to u-law to be sent down the line shouldn't suffer any alteration at all - if it does, the there's something wrong with the translation. The quality dropping over time is almost certainly down to something else. Vinuth -can you get a

Re: [Freeswitch-users] Choosing a Codec.

On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Indeed. Storing prompts as 8k, 16-bit WAVs makes a lot of sense. [I am inordinately

Re: [Freeswitch-users] mod_conference scalability

Hi Brian, Have a look at this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop - I took a quick look through the code and couldn't see any reason why you shouldn't have a bunch of eavesdroppers listening to a single caller. I'd be surprised if this didn't perform a lot better for

Re: [Freeswitch-users] SIP Re-invite

I'd be suspicious of: (a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3; (b) the branch on the Via tag changing (c) (not sure about this one) the SDP session ID and version changing for what's the same session. --Dave Anthony, I have pasted the invite sip trace here:

Re: [Freeswitch-users] SIP Re-invite

Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave The trace that I pasted on the pastebin was from our analyzer,Tektronix

Re: [Freeswitch-users] mod_conference scalability

Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple

Re: [Freeswitch-users] FS Rocks!!!!!!!!!

On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: Update to latest Did you type make current yet? Tony hates build skew Brilliant. Michael Collins-san Shrinks all usual advice Into one Haiku. --Dave ___ FreeSWITCH-users mailing list

Re: [Freeswitch-users] Hardware echo cancellation.

On Fri, 2009-11-20 at 09:57 +0800, Steve Underwood wrote: On 11/20/2009 05:15 AM, David Knell wrote: On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: The audio path between kernel and user space is not stable with any current PC based telephony system. At some point

Re: [Freeswitch-users] Hardware echo cancellation.

: can be done in a cross-platform way, can use FPU/MMX/SSE without guilt and voodoo, etc. And there is no reason why the same algorithm would perform differently if implemented in hardware or on the host CPU. And the OP only needed four E1s.. --Dave /b On Nov 18, 2009, at 5:39 PM, David

Re: [Freeswitch-users] Hardware echo cancellation.

On Fri, 2009-11-20 at 00:15 +0800, Steve Underwood wrote: The audio path between kernel and user space is not stable with any current PC based telephony system. At some point in the day the odd chunk of data is lost here and there, whether you use asterisk, callweaver, yate or FS, with

Re: [Freeswitch-users] Hardware echo cancellation.

Hi Tim, Here you go: http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html I am about to build a new machine as a VOIP server. I am going to get either a quad core intel or a six core AMD processor with at least eight gigabytes of RAM in it. Given that much horsepower I am

Re: [Freeswitch-users] Hardware echo cancellation.

Hi Tim, In (very) brief: maybe, no, and depends on the definition of 'lots'. By lots I mean somewhere between 50 to a 100 but it's mostly an IVR application so all it will be doing is either playing prompts or recording messages. Almost no live conversations. For the sort of box you're

Re: [Freeswitch-users] Detecting Forwarding Device Caller ID (Off Topic)

Hi Max, Some operators (e.g. Orange in the UK) will allow you to have two numbers attached to one SIM, which might get you 4/10ths of the way to your goal. I'm struggling to see why purchasing DIDs won't help, unless your FS box has no internet connectivity. A bit more background on the problem

Re: [Freeswitch-users] Gateways, configuration or directory with mod_xml_curl

Hi Eric - The way that we do it is to keep each gateway in its own Sofia profile. Issuing api sofia profile profile start reloadxml does one call to the web server for that profile's XML, which can be pretty compact if it just contains one gateway. --Dave Hello, We are looking at

Re: [Freeswitch-users] More than One DID

://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http

Re: [Freeswitch-users] Automagic Phone Provisioning

@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk

Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users

Re: [Freeswitch-users] Caller Authentication

://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http

Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

Steve's point: our real-world reliability was determined elsewhere. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http

Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

month without a crash? Thank you for your Time and help in advance, and I am more than willing to take all the information gathered here and create a wiki page to help other people with the same questions/problems. best Ray -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E

Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

with that, I'm currently interested in call center + FS scenario. Diego On Fri, Aug 28, 2009 at 9:01 PM, David Knell d...@3c.co.uk wrote: Hi Raimund, One FreeSWITCH box will be quite enough to handle the call volumes that you're talking about

Re: [Freeswitch-users] Odd sonus warning

://messages.finance.yahoo.com/Stocks_(A_to_Z)/Stocks_S/threadview?m=tebn=16942tid=827769mid=-1tof=15rt=1frt=2 for an amusing read ;-) --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users

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

/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing

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

On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote: Just curious - if it seems to be working with Asterisk but not FreeSWITCH then could you do some tcpdumps of working vs. non-working calls and then analyze them with Wireshark? I think Jason Garland's ClueCon presentation(s) might be

Re: [Freeswitch-users] Recommendations for an interface between FreeSwitch and ANSI C code

from others, the event socket is almost certainly the right place to start. We've used it exclusively for our stuff for getting on for two years, and I've no regrets. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk

Re: [Freeswitch-users] VoiceMail transcription

the computer spit out. If you have any insights on this, that would be great. -pete Original Message Subject: Re: [Freeswitch-users] VoiceMail transcription From: David Knell d...@3c.co.uk Date: Mon, August 10, 2009 11:51 am

Re: [Freeswitch-users] Loopback and bypass_media

Just to add my $0.02-worth (if you're feeling generous..) - I don't think that the dialplan is expressive enough to do what's needed here, and that's where the trouble's coming from. It's not enormously tricky to build a generic dial this set of numbers according to these rules service using

Re: [Freeswitch-users] arriving today for ClueCon

-- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk On 3 Aug 2009, at 13:41, Brian West br...@freeswitch.org wrote: Just look for large groups of people with laptops. I'm sure you can't miss us. /b On Aug 3, 2009, at 5:46 AM, Chad Phillips

Re: [Freeswitch-users] FS beats Aculab Prosody S on subjective test on lay users for conference quality

/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing

Re: [Freeswitch-users] Application to Record Calls - Out of Band

/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman

Re: [Freeswitch-users] Barge on on prompts

___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David

Re: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid

/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users

Re: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid

wrong there! Gotta use CDATA because it has and in the data you're setting. And you'll wanna export it I suspect. action application=export![CDATA[sip_h_P-Asserted-Identity=sip: ${caller_id_numb...@1.2.3.4]]/action /b On Jul 16, 2009, at 9:51 AM, David Knell wrote

Re: [Freeswitch-users] Even socket packets/chunks

://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch

that are important to me are answering machine detection as well as detecting SIT intercept tones in the early media stream... any love here? Not sure on these, but I'm *am* sure that someone else will be ;-) Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W

Re: [Freeswitch-users] Baby Update!

Britain's favourite poet) whose This Be The Verse suggests otherwise: http://www.artofeurope.com/larkin/lar2.htm [as a recent father myself, I'm trying to prove him wrong..] --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk

Re: [Freeswitch-users] Language Handling: call for assistance

-- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http

Re: [Freeswitch-users] Is there any license G729?

. For example, my family is geographically quite dispersed, and we use Skype with video a lot - particularly for grandparents to keep up with grandchildren. The usefulness and appropriateness of video calling depends very much on the target market; it's not, of itself, a bad thing. --Dave -- David Knell

Re: [Freeswitch-users] Announcement: Howler-optimised G.729A Solution for FreeSWITCH

it. Expecting them to give something back in addition would be entirely unreasonable. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users

Re: [Freeswitch-users] FS as a Class 5 switch

low-end modern server (Core 2 Duo, etc.) ought to do just fine. Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] Sample rate and recordFile

-- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http

Re: [Freeswitch-users] FreeSWITCH to sell Content, tones, MMS, etc ?

Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk

Re: [Freeswitch-users] Is Freeswitch ready for prime time?

- some wholesale in/out, some IVR, some calling card, some callthrough - with a total value in the millions of dollars; we have no complaints. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk

Re: [Freeswitch-users] funny effect after minimizing xml files

://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ Freeswitch

Re: [Freeswitch-users] Make current fails (build 13537)

At the risk of evisceration (but with the intention of helping avoid future brain dead build vs. idiot admin debates), I'd suggest that, when significant new bits are added to the switch core, they should default to being off and require a configuration option to turn them on. Such config

Re: [Freeswitch-users] g729 support

If you live in patent-free country, you can try this: http://github.com/Deepwalker/fs_itu_g729/tree/master I've a friend who says he knows of someone who's tried it in non- patent-free countries, and it works fine there too. Alternatively, an Asterisk box makes a perfectly good G.729 to

Re: [Freeswitch-users] Unicast isn't working

Hi Artem, Please to see that some of the stuff I wrote is useful to someone..! I've written an FS module which will send the audio over - it's more efficient than using unicast. Let me know if you'd like a copy. Cheers -- Dave - Original Message - From: ? To:

Re: [Freeswitch-users] Double-dtmf detection in IVR when a call isrouted through FreeSWITCH

Hi Drew, When you say that the problem goes away if you don't use start_dtmf, do you mean that you get one tone recognised per tone or none? If the former, then you've got DTMF being signalled out of band as well; in that case, why do you need inband detection? --Dave - Original

Re: [Freeswitch-users] Testimonials

:-) Cheers -- Dave David Knell wrote: Hi Maxim - We've used FreeSWITCH for switching large volumes of wholesale traffic and for a variety of IVR services; we no longer use anything else. See http://www.softivr.com for something which we've built on it. Cheers -- Dave

Re: [Freeswitch-users] no RTP send during Voice Mail recording

Add something like memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE); after char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE]; in switch_ivr_play_say.c (line 395) --Dave Thank you very much, dujinfang, for your help! When I use action application=set data=record_waste_resources=true/

Re: [Freeswitch-users] Compact, fanless appliance?

You might want to take a look at this: http://www.amazon.com/IEEE802-11N-Wireless-Broadband-MZK-W04NU-Designed/dp/B000YDS0YG - twice as much everything as the NSLU2, and is supposed to run OpenWRT just fine. I've one sat in front of me right now, although I've not yet plugged it in - have to

Re: [Freeswitch-users] VoiceXML

. FreeSWITCH needs demand to get vxml and it's not there yet. For now, it looks like the FS community is waiting for demand instead of trying to create it. David Knell wrote: On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same

Re: [Freeswitch-users] VoiceXML

On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or

[Freeswitch-users] Event sockets and IPC

Hi - Consider a situation where I've two call legs up, each talking to a controlling process using an outbound event socket connection from FS. They each know the other's UUID. I'd like to send an arbitrary message from one to the other. I was going to build some sort of IPC thing, but it

Re: [Freeswitch-users] Entire Wiki.FreeSwitch.org on Single PDF ?

If someone has a way to make true mirrors that support read/write this would be interesting. Do it robustly, transparently and in real time and that's the problem of distributed source code revision control mostly sorted as well. Although I'm not sure I'd really want to use the kernel anyone

Re: [Freeswitch-users] FreeSwitch Complex IVR System

Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever platform you choose. The documentation starts here:

Re: [Freeswitch-users] Re-2: FreeSwitch Complex IVR System

-Apr-2009 17:35) From:David Knell d...@3c.co.uk To: g...@exram.de Hi Guido, My preferred way is to talk to FS through its event socket interface. This allows you fully to control FS, whilst giving you the power to write the code in whatever language and on whatever

Re: [Freeswitch-users] Call bridge in free switch

Take out the brackets - originate sofia/profile/1001... (and you might want to replace profile with the name of the profile to use) There's documentation here which might help: http://wiki.freeswitch.org/wiki/Mod_commands#originate --Dave Hi Mike, I tried the following command per ur

Re: [Freeswitch-users] mod_vmd--can it do more than simply find VoiceMail Beeping?

On Tue, 2009-04-07 at 17:17 +1000, Jason White wrote: Matthew Fong mattdf...@gmail.com wrote: My question is, is there a way to use mod_vmd to detect if an answering machine or human has picked up within the first 1-2 seconds after being answered? Probably not. If you have an algorithm

Re: [Freeswitch-users] How to call multi gateways for failover with early media?

On Mon, 2009-04-06 at 00:08 -0400, Kristian Kielhofner wrote: Actually using 180 w/o SDP provides for enhanced call handing functionality while only requiring (in many cases) one additional test scenario. Consider the current example (all 180s are actually 180s w/o SDP and 183 is 183 w/

Re: [Freeswitch-users] How to call multi gateways for failover with early media?

On Sat, 2009-04-04 at 03:32 -0400, Kristian Kielhofner wrote: 180/183 with SDP is a bit ambiguous. I always get frustrated when various people /insist/ on using 183 w/ SDP just for ringback. Have they never heard of 180 w/o SDP? Let me generate my own local ringback and/or handle the

Re: [Freeswitch-users] Another FreeSWITCH First!

Here's a sample SIP/SMTP INVITE (responses omitted for clarity) MAIL FROM: d...@3c.co.uk RCPT TO: mar...@3c.co.uk DATA Call me . --Dave Sent from my iPhone On 1 Apr 2009, at 09:15, Brian West br...@freeswitch.org wrote: You know you could write a transport plugin for Sofia that would do SIP

Re: [Freeswitch-users] freeswitch as a session border controller

Steve Underwood wrote: Anthony Minessale wrote: I'm really starting to feel like we're playing musical threads here. Just avoid playing them through low bit rate codecs. :-) I think we need an echo canceller ;-) --Dave ___

Re: [Freeswitch-users] echo cancellation on PRI cards

Raymond Chandler wrote: What's interesting to me is everyone on this thread except you has said that in real-world scenarios, they need the EC for reliability. One of which, does signal processing programming professionally. It seems to me that if you build a better mouse trap you must

Re: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102

Sorry - my bad - dtmf-type looks like it just controls what's sent, not what's received. Brian's advice is sound, or you can probably work around things right now by editing src/mod/endpoints/mod_sofia/sofia.c - at around line 3838 you'll find: if (dtmf.digit) { /* queue it up */

Re: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102

Hi Brian, It's more sane to have the phone to NOT send them both in the first place because it is WRONG to send both info and 2833 and NOT totally expect the far end to make heads or tails of it. How about actually have the phone manufacture fix their broken phone? In an ideal world,

Re: [Freeswitch-users] Problem about always Double accept DTMFs

Hi - It looks like you're getting digits both in the RTP stream and as SIP INFO. Try adding param name=dtmf-type value=rfc2833 / to the SIP profile you're using for inbound calls. --Dave I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS to PSTN with DID numbers. For

Re: [Freeswitch-users] Cepstral and RSS feeds

In the meantime, you can work around this by using the swift executable to turn text in to WAV files, and then just play them back. Works fine for short(ish) texts - there might be a bit of a delay if you wanted the thing to read back War and Peace. --Dave

Re: [Freeswitch-users] Cepstral and RSS feeds

, at 8:53 PM, David Knell wrote: War and Peace. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: When there is Echo being generated from the far end, usually in a bridged call. If you application is just an IVR, with no far end connectivity, then you shouldn't need an echo can. If you are bridging calls, then at some point you may need it, depending on what else is

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: David Knell wrote: Steve Underwood wrote: When there is Echo being generated from the far end, usually in a bridged call. If you application is just an IVR, with no far end connectivity, then you shouldn't need an echo can. If you are bridging calls

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: [whopping big snip] The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. I'm sorry if you find disagreement offensive; you might not wish to read beyond this point if so. and, in the case of the decade-old

Re: [Freeswitch-users] echo cancellation on PRI cards

Steve Underwood wrote: David Knell wrote: Steve Underwood wrote: [whopping big snip] The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. I'm sorry if you find disagreement offensive; you might

Re: [Freeswitch-users] Rewriting Remote Party ID

Hi Rod, You can set it directly: action application=set![CDATA[sip_h_Remote-Party-Identity=${caller_id_number} sip:{$caller_id_numb...@1.2.3.4;screen=yes;privacy=off]]/action action application=bridge data=sofia/gateway/wherever/+$1/ --Dave using these functions like this did nothing on

Re: [Freeswitch-users] FS SIP audio quality?

louder as sounding better on the same source material - even if the additional volume isn't detectable as such. Which, I guess, explains my 25 years of going to Motorhead gigs. --Dave -- David Knell, Director, 3C Limited T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 http://www.3c.co.uk

Re: [Freeswitch-users] Thread hijacking and BT interop

Oops - I did it again ;-) --Dave ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

Re: [Freeswitch-users] Struggling with Originate

Hi Nik, How do you need to modify it? Cheers -- Dave Hi Guys, I’m placing calls ok by using the event socket. However, I need to modify the To: Sip header prior to the call going out for routing purposes. Is it possible to do this in the Originate action? If not, can someone explain

Re: [Freeswitch-users] Generating calls from external source

Hi Nik, Here's a snipped in Perl that launches an outbound call: if (my $sock = IO::Socket::INET-new(Proto ='tcp', PeerAddr = '127.0.0.1', PeerPort = 8021)) { print $sock auth XXX\n\n; print $sock api originate

Re: [Freeswitch-users] How to bridge without Answer?

/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http

[Freeswitch-users] FreeSWITCH, MRCP and Perl

-- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] SBC configuration

Hi Ash, It's the former. Here's a snippet from a dialplan of ours - this takes calls with a specific prefix from a specific IP address and forwards them to a particular carrier: extension name=sample condition field=network_addr expression=^10\.1\.1\.12$ / condition

Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl

Hi Brian, With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the

Re: [Freeswitch-users] voicemail disk quota poll

://www.freeswitch.org -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo

Re: [Freeswitch-users] Bridging from Event Socket API

/[EMAIL PROTECTED] [BREAK] Any idea what i'm missing? Thanks, Klaus. -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users

Re: [Freeswitch-users] Problem with Freeswitch capturing DTMF

/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

Re: [Freeswitch-users] mod_cepstral

there was a problem with one the apr lib. So ensure you are on the current sources. Best regards Peter David Knell schrieb: Hi - Is anyone using mod_cepstral with Cepstral 5.x? I installed demo Cepstral and mod_cepstral this morning; the module loads fine, but then complains about a couple

[Freeswitch-users] mod_cepstral

Hi - Is anyone using mod_cepstral with Cepstral 5.x? I installed demo Cepstral and mod_cepstral this morning; the module loads fine, but then complains about a couple of symbols not being provided in the libs when asked to speak. --Dave ___

Re: [Freeswitch-users] Supress Unregister at external gateway

://www.freeswitch.org -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo

[Freeswitch-users] mod_openmrcp

Hi - Is anyone out there using mod_openmrcp? Be interested in finding someone further down the learning curve than I am at the moment.. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

Re: [Freeswitch-users] How many calls can FreeSwitch bridge simultaeously

it to do. Mike On Nov 16, 2008, at 5:39 AM, David Knell wrote: Hi George, We've run a few (up to four, I guess) hundred concurrent calls without trouble routing media as you describe on a Core 2 Duo box. Unless you've got gigabit Ethernet, you'll run out of bandwidth not far beyond this point

Re: [Freeswitch-users] Recommendations on ATA's for WinXP and motherboard.

for testing with no problems at all. I'd expect any ATA to work fine, +/- some possible tweaking if there's NAT happening somewhere between the ATA and FreeSWITCH. --Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

. (the persing is the same for both situations) David Knell wrote: [EMAIL PROTECTED] wrote: Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't

Re: [Freeswitch-users] mod_cdr revival (or new module maybe)

for a couple of seconds Two error cases: can't parse or can't find data which should be there: move the file in to another directory to be examined by real eyes; DB insert fails: break out of inner loop and it'll be retried after a short pause. --Dave -- David Knell, Director, 3C Limited T: 020 8114

Re: [Freeswitch-users] playback: Easy ways to handle different situations?

On Oct 21, 2008, at 12:11 PM, Dennis wrote: 2008/10/21 Brian West [EMAIL PROTECTED]: If you add event-lock:true to your message you send then it will do what you want. It will play the files in order as you expect. Thanks, this works quite nice for playing a list of soundfiles. BUT,

Re: [Freeswitch-users] Socket inbound or outbound with PHP?

Hi Dennis I think I have problems to understand what is happening in detail. Just a short explaination, of what we are doing. We use socket outbound and have a PHP script, which is listening to myevents. If a new call is coming in, we fork the PHP script in a new process to handle this

Re: [Freeswitch-users] Newbie: Avaya SES Freeswitch 407 Proxy Authentication error

On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote: Record-Route: sip:10.0.2.154:5060;lr Record-Route: sip:10.0.2.151:5061;lr;transport=tls what's the 'lr' next to the port number? short for 'loose routing' - see here for a bit of an explanation:

Re: [Freeswitch-users] OT: Phoenix Artwork?

Hi Mike, http://images.google.co.uk/images?hl=enq=phoenixbtnG=Search+Imagesgbv=2 might help, and congratulations on the new middle name! How's it pronounced? Cheers -- Dave On Oct 12, 2008, at 6:58 AM, Michael S Iokytwjnmiwqasxz Collins wrote: If anyone knows of cool Phoenix artwork

[Freeswitch-users] SoftIVR.com

Hi - I'm after some alpha testers of a very alpha product. SoftIVR is something we've been putting together for a little while - it's a hosted IVR, using FreeSWITCH for its back end. A very quick rundown of what it does and how it does it: - each user can have one or more virtual IVRs; -

Re: [Freeswitch-users] Bridge 2 calls, catchup hangup and connect to 3rd party

Hi Brigit, Set continue_on_fail=true and hangup_after_bridge=false should let you just continue with your script after the bridge. Cheers -- Dave Hi, I'm running FreeSWITCH 498:8901 on x86_64 GNU/Linux. I'm trying to get person 1 to speak to person 2, then after person 2 hangs up,

Re: [Freeswitch-users] Load test - performance not even matching Asterisk

Going back a step, to where Jon was seeing more packets than there should have been, I've just encountered a similar issue having upgraded to the latest, from what was probably a fairly old release - months old, rather than weeks. I've got two FS boxes (let's call them FS1 and FS2), each of

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