Re: [asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party)
I've found an interesting link. It might help you out. http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html Peter Joe Carroll wrote: > Hello... > We're attempting to track down an intermittent echo issue. Our setup is > sipsippri to carriers. We have less than 2 ms latency > on the networks (FTTx), totally SIP w/ G711u. The party hearing the echo is > the subscriber using sip. The PSTN users does not hear the echo. > > We should be note that there is zero echo when calling sip to sip with or > without reinvites enabled. > > We have several different phones; linksys, polycom, & grandstream (both atas > and phones). It's difficult to reproduce the problem regularly so isolation > is an issue. > > > Thanks in advance.. > -Joe > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] odd audio problem
I am having an odd audio problem. See setup diagram below. When a call comes in it get routed through the 1st asterisk box (currently running 1.2) through another asterisk box (running 1.4.11). All audio is good. When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays to the 2nd asterisk box. The AA answers the call and the audio is good. Once the call is forwarded to an agent. The agent hears everything no problem, but the audio returned to the callers is really bad. It sounds like it is missing 75% of the audio. There is no packet loss and 10 ms ping times between the two asterisk boxes. All audio streams are g729 and there is no trans coding anywhere. When I recorded the audio on both asterisk boxes using Mixmonitor, the recorded files sounded good. +-+ | TNT MAX | +-+ | | SIP G729 V ++ | Asterisk Box 1 | ++ | | IAX2 G729 V ++ | Asterisk 1.4 Box 2 | ++ | | IAX2 G729 V Agents Peter ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems chanskype on ubuntu gutsy
Hi all, I'm trying to install chanskype (http://www.chanskype.com) on an Ubuntu gutsy machine with 2.6.22-14-generic kernel. First I got some compile errors, include linux?capability.h in main.c fixed those errors. Now I'll try to load theier module, but got the following error: insmod: error inserting './ivcs.ko': -1 Unknown symbol in module In dmesg I find: [3123904.331682] ivcs: Unknown symbol malloc_sizes My searches on google don't give me any usefull result. Has someone any id or suggestions? I'm using the latest version of chanskype available on their website. there is only a bin foor ubunut 6.x. Thanks in advance, kr -Peter signature.asc Description: Digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] include statements in IVR
Try removing the include statements from the default context and see what happens. Also double check to make sure calls are sent to the default context. Peter On Nov 2, 2009, at 3:40 AM, Thomas Person wrote: > I want to match specific contexts to menus. > If users dial a number (example: 1703444) then start with > context big10-IVR > If users dial a number (example: 1567444) then start with > context cleveland-IVR > It is not working. I have played with the include statements and am > close but no cigar. > > Here is a part of my config. Please send comments. Thank you > > > [default] > ;include => stdexten > include => big10-IVR > include => cleveland-IVR > exten => _1703XXX,1,Goto(big10-IVR,s,1) > exten => _1567XXX,1,Goto(cleveland-IVR,s,1) > > > [big10-IVR] > exten => s,1,Answer() > exten => s,n,Background(dir-welcome) > ;exten => s,n,WaitExten(1) > ;exten => s,n,Background(astcc-please-enter-your) > ;exten => s,n,Background(zip-code) > ;exten => s,n,Wait(7) > exten => s,n,Background(washington-dc) > ;exten => s,n,Authenticate(,a) > ;exten => s,n,Background(pin-number-accepted) > exten => s,n,Playback(queue-thankyou) > exten => s,n,Background(ginger110109) > > [cleveland-IVR] > exten => s,1,Answer() > exten => s,n,Background(dir-welcome) > exten => s,n,WaitExten(1) > exten => s,n,Background(astcc-please-enter-your) > exten => s,n,Background(zip-code) > exten => s,n,Wait(7) > exten => s,n,Background(washington-dc) > exten => s,n,Authenticate(,a) > exten => s,n,Background(pin-number-accepted) > exten => s,n,Playback(queue-thankyou) > exten => s,n,Background(ginger110109) > exten => s,n,Hangup() > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] include statements in IVR
Check the channel driver configuration file, or fire up CLI with max verbosity and monitor its output while calling the dialplan extensions. CLI is like a good friend that tells you whats going on and if there are any errors in you configuration. Peter On Nov 2, 2009, at 4:39 AM, Thomas Perron wrote: > How do I check > > On 11/1/09, Peter wrote: >> Try removing the include statements from the default context and see >> what happens. Also double check to make sure calls are sent to the >> default context. >> >> Peter >> >> On Nov 2, 2009, at 3:40 AM, Thomas Person wrote: >> >>> I want to match specific contexts to menus. >>> If users dial a number (example: 1703444) then start with >>> context big10-IVR >>> If users dial a number (example: 1567444) then start with >>> context cleveland-IVR >>> It is not working. I have played with the include statements and am >>> close but no cigar. >>> >>> Here is a part of my config. Please send comments. Thank you >>> >>> >>> [default] >>> ;include => stdexten >>> include => big10-IVR >>> include => cleveland-IVR >>> exten => _1703XXX,1,Goto(big10-IVR,s,1) >>> exten => _1567XXX,1,Goto(cleveland-IVR,s,1) >>> >>> >>> [big10-IVR] >>> exten => s,1,Answer() >>> exten => s,n,Background(dir-welcome) >>> ;exten => s,n,WaitExten(1) >>> ;exten => s,n,Background(astcc-please-enter-your) >>> ;exten => s,n,Background(zip-code) >>> ;exten => s,n,Wait(7) >>> exten => s,n,Background(washington-dc) >>> ;exten => s,n,Authenticate(,a) >>> ;exten => s,n,Background(pin-number-accepted) >>> exten => s,n,Playback(queue-thankyou) >>> exten => s,n,Background(ginger110109) >>> >>> [cleveland-IVR] >>> exten => s,1,Answer() >>> exten => s,n,Background(dir-welcome) >>> exten => s,n,WaitExten(1) >>> exten => s,n,Background(astcc-please-enter-your) >>> exten => s,n,Background(zip-code) >>> exten => s,n,Wait(7) >>> exten => s,n,Background(washington-dc) >>> exten => s,n,Authenticate(,a) >>> exten => s,n,Background(pin-number-accepted) >>> exten => s,n,Playback(queue-thankyou) >>> exten => s,n,Background(ginger110109) >>> exten => s,n,Hangup() >>> >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com >>> -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to escape characters in Dialplan
Somewhere \n needs to be converted into utf8 new line. Asterisk should
do this for you but it doesnt.
Try opening the dialplan in hex mode and insert hex code for utf8 new
line where the line break should be.
Peter
On 17 jan 2010, at 12.09, Dominik wrote:
>
> Hello,
> I'm using Asterisk 1.6.2.0 and I like to use escape characters with
> SendText,
> because I can just delete the message from my phone (Thomson
> Speedtouch
> ST2030) display by sending a return-char (\n).
> But \n is not escaped: I tried already:
>
> exten => 222, n, SendText(\n)
> exten => 222, n, SendText("\n")
> exten => 222, n, SendText('\n')
> exten => 222, n, SendText(`\n`)
>
>
> So how can I use escape characters in dialplan?
>
>
> TIA,
> Dominik
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] GSM Gateway
Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend -> trunk or usb ? Or there are other possibilities ? Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway
Hi, I can not find pricing and shipping information for these. I tried to contact their sales for these. We will see, but most likely we will go with portech. Peter On 08.2.2010 15:15, Peter den Hartog wrote: > http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html > > We use this one, and it works great.. easy to setup and it works with a > normal network connection :) > > On Mon, Feb 8, 2010 at 1:52 PM, Peter <mailto:peterp...@aboutsupport.com>> wrote: > > Hello, > > I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS > analogue connection. > > I searched the email archives and found messages from 2008 but not sure > how accurate these are. > > What do you use and how well it works ? The only sensible one I found > is one made by portech and one that is made by Eurodesign. > > The one from portech is like a trunk while the one from eurodesign > relies on USB and project GSMOPEN. > > what would you recommend -> trunk or usb ? Or there are other > possibilities ? > > Thanks, > > Peter > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Groet // Kind regards, > Peter den Hartog > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
On 10.2.2010 18:06, Steve Howes wrote: > On 10 Feb 2010, at 15:50, Peder wrote: >> check out the Cisco SPA504G. They are the newer versions of the >> SPA922, support multiple lines and are fairly cheap too. > > I second that. They're rock solid, good audio quality and easy to > provision. > > S > SPA504G - 1 more vote for it. It is worth having 4 lines even if you need 1 initially. SPA504G supports G722 and sound is awesome even if you do not not use teh HD sound. If you do not care that mcuh about HD sound and do not need PoE SPA941 is a excellent choice - you get really a lot for the price Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone recommendation
SPA504G is LINKSYS with newer look and HD :-) Expect all you had in Linksys SPA9XX + more. I personaly have both phones - differences are not a lot :) Peter On 10.2.2010 20:31, Jeffrey Ollie wrote: > On Wed, Feb 10, 2010 at 12:23 PM, Brent Torrenga wrote: >> >> Coming from someone who uses 7940's and 60's: has Cisco/Linksys embraced >> SIP compatibility with asterisk more completely with the SPA504G's than they >> have the 7940 series? Lack of features on the 7940's is frustrating, and >> makes me hesitant to try other Cisco phones, even if the SPA504G is newer. > > SIP support in newer generations of the 79XX series is much better. I > believe that their goal is to have 100% feature parity between the SIP > and the SCCP images, they are probably 90% now. Whether Asterisk > supports all of those features is another matter though. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
>> On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: >> > >> > Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? >> > >> I was suspecting something with either rtptimeout or sip registration >> timeout, but I'm not sure what. Hi, I have had similar issue. I have downgraded from 1.6 to 1.4 and issue got solved. Never managed to find what is going on. It was happening only if all were true: - linksys phone or pap - asterisk 1.6 - use certain VOIP provider. Solution: moved to 1.4 I hope thsi helps. Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXW4024
Hello, I consider buying three GrandStream GXW4024 and connect 72 analogue phones to asterisk Do you have any feedback how well it works with Asterisk ? I am on a budget, do you have other recommendation for similar setup that get into same budget - connect around 70 analogue phones to asterisk. Thanks in advance. Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Re: SpiderMux?
Hi, I have one in stock - got it from a client who wanted to get rid of all his old IT equipment. Looks strange, did not have enough time to play with it Tried it once, looked hard to configure. It stays unused in the storage room. Peter On 29.4.2010 10:20, Tim Nelson wrote: > Greetings all- > > I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks > rather interesting. Has anyone used one? Where did you purchase it? Pricing? > Operational issues? > > http://spidermux.com/ > > Tim Nelson > Systems/Network Support > Rockbochs Inc. > (218)727-4332 x105 > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 license
Hi all. Dopes someone know how I can move a key license of the g729 codec from one to another machine? Find nothing usefull @ the wiki. Thnx 4 help in advance. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 license
Hi. I'ved registered it for 2 times, so I've got to contact digium. Thnx 4 info. On Mon, May 02, 2005 at 10:26:35AM -0400, Pedro wrote: > Actually called Digium with this exact question last week. They said > that you can register the new license on the new server provided that > you ony registered it once before. They said there is no "unregister" > script to unregister the license from the old server, however. If you > have already used up your 2 registrations, you will need to contact > Digium for assistance on this. I also asked if leaving the keys on my > dev. box would cause a conflict (also was pretty clear that I wanted > to be in compliance with their license agreement) and the lady said > there was no problem and leaving the old keys on the dev. box would > not cause a conflict. > > On 5/2/05, Peter <[EMAIL PROTECTED]> wrote: > > Hi all. > > > > Dopes someone know how I can move a key license of the g729 > > codec from one to another machine? > > Find nothing usefull @ the wiki. > > > > Thnx 4 help in advance. > > > > Regards. > > > > -Peter > > > > -- > > Please no HTML, I'm not a browser > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] large scalable voip setup
Hi all. Is there anyone who have a big experience with large scalable voip setup and want to share some experience, knowlegde? I need to handle a lot concurrent calls, to pstn and to sip gateways' The current setup can't handle the load anymore. I've some solutions in mind, but don't know if it fits well. If someone is willing to communicate, it would be every appriciated. Regards. -Peter -- Please no HTML, I'm not a browser ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK, Disconnect supervision
Thanks very much! I'll definitely sook at these resources as well if other problems come up. I also found some more info on sipura setup un the UK, see http://lists.digium.com/pipermail/asterisk-users/2005-December/140037.html Thank you very much again for your comments. Good to hear that you had no issues with firmware upgrades. I feel a bit more encouraged about it. God bless, Peter Peter, I'm using the firmware 3.1.5(GWb) and was wondering if your suggestions would be of any benefit to me. Incidentally, I've never had an issue upgrading or downgrading the firmware in 2 spa-3000s, I just had to make sure the unit had only just been powered up when initiating the upgrade. (YMMV) Anyway, if you're wanting somewhere else to read & ask questions have a look at http://voxilla.com/forum-viewforum-f-14.html for Sipura/Linksys adapters or http://voxilla.com/PNphpBB2-viewforum-f-17.html for asterisk. Fantastic resources with helpful & knowledgeable respondents. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interest in E1 channel banks?
Brendan, Can we catch up tommorrow before 3pm or around middle of the day on Tuesday next? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHP Gui for Asterisk (AGI questions)
Hi, XML may be the latest but it also adds latency to the whole process - for what benefit? It looks better, we are using the latest technology? If a wheel barrow will do the job why get a D9 Tractor? No flame wars pls, just my 2cents worth. Peter At 19:00 17/03/2003 -0500, you wrote: >I hate to say do it Microsoft's way; but they FINALLY came around with >Win2003 to storing the web server config in XML; and after revisions of >registry storage (basically param=value format), then metabase with >inheritance issues (custom format, no tools to edit) and now they went XML. > >I've always liked the apache layout (although I make a living on IIS) - This >new XML one, although I haven't played with it much yet, looks like the way >*ALL* configs should be. Not that IIS config is the way - but XML. > >As was said, other editors can do it, there's components (windows and *nix >based) to parse xml readily available, etc. > >I've said for a long time xml is NOT the be all and end all like people >profess, and it's ended up doing things that there's no reason to do - >however for config files it looks like a great answer. > >Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers >BitShop, Inc. - http://www.bitshop.com - $149/month colo special > > >-Original Message- >From: Chris Albertson [mailto:[EMAIL PROTECTED] >Sent: Monday, March 17, 2003 5:23 PM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] PHP Gui for Asterisk (AGI questions) > > >I think the way to go with conf. file for Asterisk is XML. > >When I first saw the Asterisk conf files I wondered if Eric >Allman had found a new job working on Asterisk. (That's >a joke for those of you who have had to maintain a sendmail >installation. sendmail.cf is the definition of cryptic) > >Some advantages of XML: > >1) Parsers and file editors already exist for XML. Users could > edit files with ready made GUI tools, programmers can use > XML with XML libraries. There are even web-based tools for > maintaining XML data. > >2) Parsers and file editors can perform file validation. Making > it not-possible to save an invalid file. > >3) (some) Database systems can gobble up XML and spit it back > out. Yes, I think the DBMS idea was resonable for a large > installation. Overkill if less then say a few hundred > extensions. Large sites like to manage phone extension and, > extension to physical location maping and other stuff in a DBMS. > >4) XML (with addition of a style sheet) can be directly displayed > in a web browser > >5) Without a GUI and/or wrb front end the system will remain > only "geek usable". (Your average "phone guy" doesn't know > how to use vi.) > >6) XML readers can ignor parts of the XML file they don't understand. > This allows one file to carry information for multiple readers > ad for new additions too the file not to break older readers. > >--- Steven Critchfield <[EMAIL PROTECTED]> wrote: >> On Mon, 2003-03-17 at 11:36, Stefano Finetti wrote: >> > I was wondering about a little php-based GUI to manage Asterisk >> Extensions. >> > >> > Many way to obtain this, but i think that implementing in a php >> script the >> > AGI Commands should obtain the best results (more, the best result >> would >> > come with AGI+Mysql instead of a text file like extensions.conf >> but...). >> >> Text files would be better than a database since you could comment on >> what you are trying to do with a text file. Also a text file can be >> munged easier than a database when a change in argument format comes >> out >> such as the function style of calling apps in asterisk. Maybe if you >> need webbased configuration you could make a script that held your >> working copy either in a flat file or text file , then generated a >> new >> extensions.conf file as you commit changes. Once commited, you make a >> call to asterisk to reload via the manager port. >> >> > The problem is that I've tried to understand *where* and *how* >> apply AGI >> > commands, without, of course, any good result. >> > >> > In which way AGI commands are passed to asterisk? >> > Into the console? >> > Executing applications via extensions.conf? >> >> AGI commands come from a script invoked by asterisk itself, and >> communicate via STDIN/STDOUT with asterisk. >> >> >> -- >> Steven Critchfield <[EMAIL PROTECTED]> >> >> ___ >> Asterisk-Users mailing
RE: [Asterisk-Users] Suggestion for IP phone devices.
No, supports only Cisco Skinny Protocol, no SIP or H.323 Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roderick Montgomery Sent: Thursday, 20 March 2003 5:26 a.m. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Suggestion for IP phone devices. According to Stefano Finetti: > Snom Phones, if you want SIP support > > Otherwise, any CISCO ip Phone that supports SIP or the Cisco 7905, but > in this last case you'll have to install also a H323 gatekeeper since > the 7905 works with h323 and not SIP. Anyone know whether the older Cisco 12SP+ works with asterisk? Thanks, rm - Roderick Montgomery [EMAIL PROTECTED] http://thecomplex.com/> the fool stands only to fall, but the wise trip on grace... [Sarah Masen] - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 cards
You need to get the ETSI or Euro version of PRA from Telstra and then it will work, they offer it as well as their quaint version of PRA access. Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark McKibbin Sent: Friday, 13 June 2003 1:01 p.m. To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E1 cards We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria3820 Australia www.dcsi.net.au [EMAIL PROTECTED] Ph. 1300 665575 Fx. 1300 556595 -Original Message- From: Simon J Mudd [mailto:[EMAIL PROTECTED] Sent: Friday, 13 June 2003 7:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk logging questions [EMAIL PROTECTED] (Michael Manousos) writes: > > While logger.conf indicates that you can log to a file I can't see > > _where_ the logging is sent to (which directory). Looking at the > > source it seems > > that the directory used is specified in ast_config_AST_LOG_DIR but I can't > > see where this variable is defined. Help? > > In /etc/asterisk/asterisk.conf, section [directories]: astlogdir => > <.> Ok. Thanks. Having installed all the sample conf files I'm slightly overwhelmed looking for things. Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Module app_perl
Anthony, Don't take the comments about your work too seriously. We like to do things the easiest way, and its their way of helping you . I liked your work. Could you show us a database example of your work. Peter At 14:19 17/06/2003 -0700, you wrote: >ok forget it I was telling a story of what inspired me to write >this module that embeds a perl interpreter into the asterisk process and >to see if anyone was interested in it (IT'S AN EXPERIMENT) What I get are >several replies telling me to RTFM and I would no longer need to solve the >problem that inspired my original idea. problem could be resolved >or I would never have gotten the Idea since the end-result is barely >related to that problem. Testing expressions in variables is not even the >reason to use the module the module allows you to run perl code from within >and perform(RTFM) The example I put in the README just happens to show >how to perform an eval on an expression using the new perl functionality >the module implementsI could just as easily show an example how to >connect to a database on startup and create extensions on the fly or how >to make a web server that runs inside asterisk any of these already are >possible sorry again for being inventive or for wanting to do it in perl >whichever you want to hang me for.) SheeshPlease don't bother to >further discern what I did and did not read I don't totally know how >to work asterisk cos I already know I'm simply playing with it. I only >have 2 weeks total experience with asterisk so I can tellI just thought > findings so far since I was under the impression this was a forum. >sorry.. Do you Yahoo!? > SBC Yahoo! DSL - Now only $29.95 per month! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and FWD
I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => [EMAIL PROTECTED]/1 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=1 fromuser=1 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to mailbox: exten => 71,1,Dial(SIP/[EMAIL PROTECTED]) are bounced back with 403 Forbidden What I'm doing wrong there? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - first impressions
I'm still a newbie in Asterisk, just yesterday installed it for home use (so
I can call home while travelling). Using AVM A1 (BT Speedway) ISDN card.
Anyway, I find it very hard to locate supporting information for Asterisk.
User's Handbook is still a draft, this mailing list is not easily
searchable, and any info out on Usenet is scarce. I spent some 30mins just
trying to find out naming convention for ISDN4Linux channels!
My home Asterisk is now functioning, but I still got some minor quirks to
solve:
1. SIP softphone provides "Ringing" indication when I call another SIP
softphone, but not when the call goes out via ISDN. All I get is "Trying"
until called party picks up.
2. The |r option to Dial() always hungs up after half-a-second. This happens
with all types of channels, tried with Modem, SIP and IAX. I thought this
option was meant to provide "Ringing" indication to caller?
3. Couldn't find out how to perform an action when extension goes off-hook,
before any digits are dialed. I'd like a specific extension to dial
predefined number as soon as it goes off-hook (I'll have MGCP hardphone for
which I'll want to use this).
4. Tried calling US toll-free number using
TRUNK=IAX2/gnophoneuser:[EMAIL PROTECTED]
exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:[EMAIL PROTECTED])
exten => _91800NXX,2,Congestion
The call is connected but no audio comes thru - is the gateway still in
operation btw?
Any help greatly appreciated!
TIA,
Peter
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[Asterisk-Users] Detecting off-hook state on extension
I'll have MGCP hardphone that needs to dial pre-defined number as soon as it goes off-hook. So far I'm lost as to how (if at all) this can be implemented in Asterisk. Any pointers? TIA, Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E400P E1 Pin Layout
>From Archives courtesy of Don: Email Dated: Fri, 4 Apr 2003 17:51:29 -0600 Normal RJ48 jack (some call it an RJ45 which is the same physically) out of equipment on pins 1 & 2 (Tip & Ring) into equipment on pins 4 & 5. (Tip & Ring) Normally tip is the lower number so I would *guess* that you need. pin 1 --> pin 22 pin 2 --> pin 47 pin 4 <-- pin 23 pin 5 <-- pin 48 Don Pobanz At 23:39 30/06/2003 +0200, you wrote: >What is the pin layout of the E1 sockets on the E400Ps? What pins are for >the TX and RX pair? > >Regards >Fredrik >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP with Cisco doesn't work
I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/[EMAIL PROTECTED] MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/[EMAIL PROTECTED] MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb: 'NTFY', Identifier: '2', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 0.1' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 2 OK to 192.168.154.99:2427 Posting Request: RQNT 1 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 62ea241a R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.154.99:2427 MGCP read: 510 1 Protocol Error or Version Unsupported from 192.168.154.99:2427MGCP read: 510 1 Protocol Error or Version Unsupported from 192.168.154.99:2427Verb: '510', Identifier: '1', Endpoint: 'Protocol', Version: 'Error or' 1 headers, 0 lines Posting Request: RQNT 2 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 62ea241a R: hu(N), hf(N), D/[0-9#*](N) S: ro to 192.168.154.99:2427 MGCP read: 510 2 Protocol Error or Version Unsupported from 192.168.154.99:2427MGCP read: 510 2 Protocol Error or Version Unsupported from 192.168.154.99:2427Verb: '510', Identifier: '2', Endpoint: 'Protocol', Version: 'Error or' 1 headers, 0 lines ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 woes
I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled, and chan_h323 module does not load with "undefined symbol _ZTI19H323AudioCapability". What could be the problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
> Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making > "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i get > an "Undefined symbol" error when I try to start Asterisk. As far as I can RTFM. Use specified versions of pwlib & openh323 instead of latest/CVS ones, and you should be OK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP show channels display
Why wouldn't "SIP show channels" display lag & jitter, it's always 0ms? Is there a "deeper reason" for this or this is just something not implemented yet? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
> Is it possible to use * as a gateway in the following setup: > >LAN (with Windows NT/Linux PCs) > | > Ethernet (IP) > | > Linux PC with * and AVM Fritz! ISDN Adapter > | >ISDN > | >Someone with a analog/digital phone (POTS) Works like a charm with standard I4L drivers. CAPI may work better, I4L does not recognize "ringing" condition for example. However installing chan_capi + CAPI driver is not as straightforward as I'd like it to be, and haven't got around to it yet Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
> What problems do you have with the chan_capi install? > > I am not a hardcore linux guru but it wasn't too hard to setup chan_capi.. Missing capi.h etc. I guess these are installed by CAPI driver, but I had problems compiling these... for some reason I couldn't just compile a kernel module because RH9's stock kernel is compiled with gcc2 while system is provided with gcc3 (or was it the other way around?) Then I got into kernel compilation issues, there are numerous dependency bugs in RH's kernel source, and didn't have much time to play around with the driver anymore... Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
> What problems do you have with the chan_capi install? > > I am not a hardcore linux guru but it wasn't too hard to setup chan_capi.. Missing capi.h etc. I guess these are installed by CAPI driver, but I had problems compiling these... for some reason I couldn't just compile a kernel module because RH9's stock kernel is compiled with gcc2 while system is provided with gcc3 (or was it the other way around?) Then I got into kernel compilation issues, there are numerous dependency bugs in RH's kernel source, and didn't have much time to play around with the driver anymore... Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and poor voice quality
> Calling * via SIP produces very good sound. Calling * via the chan_capi > produces horrible sound. However, if I dial 500 in the demo menu to > connect to the IAX at digium the sound is good again. ie: > > ISDNCall->AVM-B1-Card->Asterisk = All prompts sound horrible > SIP->Asterisk = Prompts are good Stupid question... why don't you use I4L instead of chan_capi? I've wanted to use chan_capi myself but due to lack of time haven't been able to get it running yet. However I4L produces good audio quality, although I miss extended ISDN features... I'm using AVM Fritz PCI card Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn4linux
> My Eicon ISDN card turned up today so - plugged it in and went through > the modem.conf. It reports unable to open /dev/ttyI0 > > The problem is I have never used ISDN with Linux - let alone a telephony > app - and I have no idea even where to start. Some pointers would be > appreciated. Check out I4L FAQ http://www.isdn4linux.de/faq/ Also, not all Eicon cards can be made to work under Linux. I have Eicon Diva Pro that wouldn't work... ever. Basically only Hisax chipset is supported Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Fritz & RedHat 8.0
Title: Message All you really should need is: modprobe hisax type=27 protocol=2 id=isdn0 and in modem.conf: driver=aopendriver=i4ltype=i4l; ISDN example;group=1msn=xxxdevice => /dev/ttyI0device => /dev/ttyI1 Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installing the drivers and which versions to use. Thanks, Stuart
[Asterisk-Users] Transitioning from existing PBX
Hello - Warning: newbie questions at 12 o'clock! I am looking at transitioning my house from a Panasonic KXT1232 to an asterisk-based PBX (*PBX). We currently have 5 CO lines and 16 extensions plus ADSL with a static IP. I have looked at the various docs, but find I still have some questions. 1. Interface cards a. Should I buy a T100P + channel bank? (BTW: Why isn't this what is in the developer's kit, instead of the T400P?) b. If I am going to simply replace the Panasonic and supply 2-wire FXS (is that correct terminology?) to the existing jacks, can I use the T100P with (Z-Plex 10? model A or B?) and have 8 FXO and 16 FXS? I think so, but the developer's kit only references FXS c. Or must I buy more equipment, e.g. 4 X100P's + 4 TDM400Ps?. d. Are there other combinations worth considering? e. Others to be avoided? 2. To prevent disruption to the rest of the household, I would like start out with the *PBX behind the KXT -- i.e. the *PBX's CO lines are actually extensions on the KXT. In this situation, is the *PBX (or Z-Plex) seeing FXS or FXO on its "CO" lines? I would guess FXS, but I'm confused. 3. If the *PBX is receiving an in-bound IP call, can it present it to the KXT as another CO line? Is this trivial or postdoctoral stuff? 4. Any suggestions about the best way to convert 2-wire analog phones to IP-based? Cisco? Grandstream? Xnet? Others? 5. We have 8 Panasonic 4-wire phones with Lots O' Buttons (tm). What should I consider for replacements? ADSI? IP-phones? Soft IP phones? Any opinions about features vs. price vs. voice quality vs. reliability? (I'm sure there are! :-) ) Thanks in advance! Peter (When I spell-checked this message, Netscape suggested "epiphanies" as a replacement for IP-phones. :-) ) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why are FXO so expensive?
> For smaller systems, you'd have to go a NetJet ISDN BRI card ($150? for two lines) Try BT Speedway BRI ISDN, ~20$ on ebay Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to detect process 2 frames
What does this error message mean? WARNING[262160]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 2 frames I've been getting these a lot lately, sound quality seems to have suffered. I'm using I4L driver with Fritz PCI ISDN card. However, even the regular echo test sounds a bit choppy over LAN connection. Happens both on CVS and 0.4.0 versions TIA, Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Question / ISDN
Hello, I am looking for a pbx solution which is not too expensive but flexible :-) (a customer is in need of a call center and crm solution). The customer favors linux on the call center server and the crm clients. On a search for solutions i found Asterisk but it looks as supports analog phones and H.323 only. My question, is a Asterisk based PBX suitable for use in a (small) call center environment ? Does it scale ? Does it work together with GNU Bayonne ? Is there a possibility to use digital (ISDN) phones ? Thanks for help Peter -- dadi-linux www.dadi-linux.de Peter Eckhardt Fon: +49 6071 951256 Weberstr. 36BFax: +49 6071 951257 64846 Groß-Zimmern [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question / ISDN
I just found the draft of the handbook. The software is amazing Does anyone use Asterisk in Germany on a BRI (S2M) interface ? Peter Brian West wrote: Peter, Did you read the website? Not only does it support h323. Inter-Asterisk Exchange (IAX) H.323 Session Initiation Protocol (SIP) Media Gateway Control Protocol (MGCP) http://www.asteriskpbx.com/index.php?menu=features bkw On Thu, 21 Aug 2003, Peter Eckhardt wrote: Hello, I am looking for a pbx solution which is not too expensive but flexible :-) (a customer is in need of a call center and crm solution). The customer favors linux on the call center server and the crm clients. On a search for solutions i found Asterisk but it looks as supports analog phones and H.323 only. My question, is a Asterisk based PBX suitable for use in a (small) call center environment ? Does it scale ? Does it work together with GNU Bayonne ? Is there a possibility to use digital (ISDN) phones ? Thanks for help Peter -- dadi-linux www.dadi-linux.de Peter Eckhardt Fon: +49 6071 951256 Weberstr. 36BFax: +49 6071 951257 64846 Groß-Zimmern [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- dadi-linux www.dadi-linux.de Peter Eckhardt Fon: +49 6071 951256 Weberstr. 36BFax: +49 6071 951257 64846 Groß-Zimmern [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Crash using alsa
Hi, i have compiled astersik using todays cvs. Worked like a charm. Asterisk runs when the oss module is enabled but crashes badly with alsa. [res_musiconhold.so] => (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [chan_alsa.so] => (ALSA Console Channel Driver) asterisk: pcm.c:5488: snd_pcm_sw_params_set_silence_threshold: Assertion `val < pcm->buffer_size' failed. Aborted (core dumped) I use alsa 0.9.6 on SuSE 8.2 (Linux 2.4.20). Just for info Peter -- dadi-linux www.dadi-linux.de Peter Eckhardt Fon: +49 6071 951256 Weberstr. 36BFax: +49 6071 951257 64846 Groß-Zimmern [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slowly get it ... Hardware
Hello, i just got asterisk up and working (without alsa). Looks really promising. Now i have some questions regarding hardware I have the following setup PSTN --- serveral PBX (Asterisk) --- digital phones PRI (S2M) Ports analog phones VoIP On the PSTN side i could use Eicon/DIVA PRI 30M cards. Would a Digium T100P or E100P also work in germany ? For conferencing it looks as I would need one Zaptel card. What is recommended in such a setup (if I can't use Digium equipment on the PSTN side) ? What is needed to connect analog and digital devices ? For analog devices it looks as I need T1 interfaces and channel banks. Is that ok ? What hardware is recommended ? And how can I connect ISDN (digital) equipment ? Is there a way to do that ? Thanks Peter -- dadi-linux www.dadi-linux.de Peter Eckhardt Fon: +49 6071 951256 Weberstr. 36BFax: +49 6071 951257 64846 Groß-Zimmern [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Still no audio on SIP phone
I have been using X-Lite on FWD without any troubles and recently became interested in trying asterisk. I am able to register from X-Lite and dial a number - I've tried dialing some of the sample numbers in the sample extentions.conf file, like 500 and 1234, they appear to dial correctly from X-lite but nothing else happens - no audio is heard. My understanding is that I should hear some sort of message. I already found one problem - on my debian system - /usr/bin/mpg123 was a symbolic link pointing to mpg321. I've corrected that and installed mpg123/unstable and made sure it was the real deal (deleted the symbolic link, etc). I am still not getting any audio. My setup: Debian (from Knoppix - a mix of unstable, testing, stable), no hardware phone cards, one software SIP phone (X-lite). Everything is on a LAN (no firewall involved). Where should I begin to find this problem? I've tried starting asterisk with lots of verbose flags, but I don't see anything suspicious. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
> > correctly from X-lite but nothing else happens - no audio is
> > heard. My understanding is that I should hear some sort of
>
> I am using x-lite with the asterisk demo no problem. All I modified was
> sip.conf
>
> Is the asterisk server and your x-lite client on the same LAN segment?
>
> Is all iptables and firewall code turned off on the asterisk server?
>
> Gavin Hollinger
>
Here is the message I am getting from Asterisk:
*CLI> -- Executing VoiceMail("SIP/2000-3296", "u1234") in new stack
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Playing 'vm/1234/unavail'
-- Playing 'vm-intro'
-- Playing 'beep'
-- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0001
WARNING[229391]: File app_voicemail.c, Line 673 (leave_voicemail): No audio available
on SIP/2000-3296??
-- User hung up
It shows it is playing the files, but nothing is heard on the Xlite SIP software side.
When Asterisk starts up, it complains about OSS and ALSA problems -
sound capabilities on the console are irrelavent in this case aren't
they?
I've tried deactivating several different codecs in X-lite - it doesn't help.
Peter.
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Re: [Asterisk-Users] Still no audio on SIP phone
adding nat=yes to the sip definition made no difference. Does Asterisk use the DSP in your sound card to do the audio processing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem solved - sort of..
I started to suspect the X-lite client in my problem (I was getting no audio when calling into asterisk) because after I would make test calls to asterisk, setting X-lite back to my FWD account - I would get no audio with FWD either, even though the sound card was working and I got dial-tone, etc. I tried the SJPHONE client - and low and behold - asterisk works fine. So it's definately not an asterisk problem. Something flaky with X-lite. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote: > > correctly from X-lite but nothing else happens - no audio is > > heard. My understanding is that I should hear some sort of > > I am using x-lite with the asterisk demo no problem. All I modified was > sip.conf > > Is the asterisk server and your x-lite client on the same LAN segment? > > Is all iptables and firewall code turned off on the asterisk server? > > Gavin Hollinger Whould you mind sharing the important bits of your X-lite config with me (network, sip settings)? Did you download the generic X-lite or the one for FWD? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
For the benefit of others having this problem - I installed the latested CVS build and the problem went away - I can hear audio now from X-lite. I was using the debian unstable package. Here's what I did: cd /usr/src mkdir asterisk export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login (supply the anoncvs password here when prompted) cvs checkout asterisk cd asterisk make make install make samples make sounds (I think that's right - memory's getting fuzzy from age) added my extention back into sip.conf works like a champ. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modems and Tivos? Oh my!
Does the Digium FXS card support modems (and Tivo devices)? If so, to what speed have they been tested? Also, on a somewhat unrelated question: How does the FXS card generate ringing voltages if the PC only supplies 12 volts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modems and Tivos? Oh my!
On Thu, Sep 04, 2003 at 06:26:03PM -0500, James Sharp wrote: > > On Thu, 2003-09-04 at 17:22, Peter Pauly wrote: > >> Does the Digium FXS card support modems (and Tivo devices)? > >> If so, to what speed have they been tested? > > Assuming that you can do native zaptel bridging (Going from an FXS port to > an FXO port in the same machine), you should be able to get up to 33.6. > No 56k, unfortunately, because of the multiple D/A & A/D conversions. > > If you're codecing the audio and passing it over IP, you should be able to > get 33.6 if you use ulaw (non-compressed) encoding. Any of the > compression based codecs will most likely make your modem link up at 9600 > or a flaky 14.4. > > > > Why would one use dialup for a TiVo. My TiVo has never touched a > > telephone line ever. When I bought it I hacked it to work over my cable > > modem link using PPP to my workstation and have never risked loosing the > > internal hardware. > > Or spend a few bucks and get yourself one of the ethernet card kits. Then > you don't have to worry about ppp connections and you can drag the video > off the unit as well. I can't use the networking kits because it is a DirecTivo HDVR2. Supplying this thing with dial-tone is a consideration for me before I dump my POTS line. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Phone -> Asterisk -> SIP LD Provider question
If Asterisk registers with a SIP long distance provider and I make a call from an IP phone through Asterisk to that LD provider, does the RTP (audio) traffic flow between the two end points directly (normally the IP phone and the LD provider) or does it flow through Asterisk? I'm asking because I have Asterisk running behind a NAT firewall along with an IP Phone (software) and I'm trying to get it working with Iconnecthere (ICH). I am able to register, connect , but no audio. I have ports opened up on the firewall, but they point to the Asterisk machine and not the IP phone machine. In this scenario, any audio traffic would have to go through the asterisk box to reach the IP phone. Is that how it works? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mixed FXO and FXS on one Adtran, T1 card?
Can I configure an Adtran channel bank with a mixture of FXS and FXO cards and have them come into a single T100P T1 card? It seems like this would be a cheaper solution than trying to load a bunch of PCI cards into a PC. Also, when shopping for an Adtran (on ebay) - what do I need to watch out for as far as interfaces or capabilities to make sure that it works with a Digium card? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xlite = no sound
On Tue, Sep 09, 2003 at 11:41:17AM +0100, Skuse, Phil wrote: > Yes. They are on the same subnet. > I solved my sound problem with X-lite by using the latest CVS version and compiling that. I had been using the stable and unstable versions out of Debian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xlite = no sound
On Tue, Sep 09, 2003 at 11:04:34AM +, WipeOut . wrote: > Where did you get access to X-Ten.com's CVS server? > > I didn't know they had the source code for x-lite available.. > Sorry, I should have been more clear - I used the latest version of Asterisk via CVS. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote: > That would be reinvite= and canreinvite= in the user entry for each SIP > endpoint. Asterisk will allow the endpoints to talk directly to each > other if both those settings are = yes (the default, I think) AND both > endpoints use the same protocol (SIP) AND the same codec. > This is the single most useful bit of info I have seen on the mailing list since I have joined. Thanks Mr. Wieling. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie - sip, pxb, ata, nat
hi all,
I don't know how to setup asterix to work as PBX.
If I want just basic configuration with 2 SIP phones (snom, ata), what
all I have to write in the configuration files, or respectively in the
configuration of ata and snom ?
If there is any good documention available, send me URL too.
All (ata, snom) are behind firewall (nat) and astrix is on the public
IP, but I can move for testing end point to the public IP.
best regards
Peter Hudec
--
mail: [EMAIL PROTECTED] www: [http://www.postel.sk]
phone: [+421 2 50203163] icq: [99518783]
gpg: [http://hudecof.net/data/hudecof.gpg]
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[Asterisk-Users] Start of all recordings cut off
I'm using a Cisco 7960 with asterisk and any recording on the machine, be it voicemail prompts, time of day, echo test message, etc, is cut off for the first 1/4 to 1/2 second. I've tried setting the phone to gsm but it still happens. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
On Thu, Sep 11, 2003 at 08:42:18PM -0600, Dave Packham wrote: > hmm works for me... its the exact same code that is installed on the sample server > listed below and I dont get the problem there. lemme know more info and ill look > into it > > Dave > Well, there is no such domain as "phpconfig". It's probably pulling this file off of your machine. However, you've suceeded in teasing us - it looks very cool so far. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Start of all recordings cut off
On Thu, Sep 11, 2003 at 09:30:35PM -0700, John Todd wrote: > > Before running any application that has sound playback (Playback, > Background, VoiceMailMain2, etc.) it would be wise to execute an > Answer first, then a Wait(2) to allow for VoIP channels to fully > establish and settle. > Adding Answer had no effect. Adding Wait(1) solved the problem. Maybe it's because Asterisk runs on a slow machine (750Mhz P3). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
On Thu, Sep 11, 2003 at 10:12:50PM -0600, Dave Packham wrote: > nope > > when I click on something on the left I get a FQDN not just the pne you had > > Hmmm. > Further info: it works with Microsoft Internet Explorer. It does not work with Mozilla 1.4 under Linux. It also does work with Mozilla Firebird under Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
On Thu, Sep 11, 2003 at 07:57:58PM -0600, Dave Packham wrote: > I have put my phpconfig stuff out into the Digium CVS tree. > > Project name is > > phpconfig. > > see it at > > http://rads.netcom.utah.edu/phpconfig/phpconfig.php > > Looks cool, but the links don't work on the left. It wants http://phpconfig/phpconfig.php? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Source for 50-pin amphenol cables?
I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Vendors for Asterisk other than "DIGIUM"
Tarun, The Digium site shows other hardware that is compatible with Asterisk. I would strongly urge you to support Digium by buying their products. Peter At 19:49 15/09/2003 -, you wrote: >Hello all, > >Could you please suggest me other hardware vendors whose FXS, FXO >interfaces are compatible with Asterisk. I am aware that Digium >products work well but I would be interested in knowing about >other options available in the market. > >Thanks, >Tarun >___ >Catering meets Camera; Tanya Weds Sahil. >Rediff Matchmaker strikes another interesting match >Visit http://matchmaker.rediff.com?1 > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nufone 800 numbers working?
Is anyone else having trouble dialing 800 numbers through Nufone? I'm getting the SIT tones no matter what number I dial. Normal long distance works fine. I don't think it's my dial plan (it was working previously). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse offering IAX2 services
I don't know if this has been mentioned yet: Voicepulse is now offering wholesale pricing and IAX2 connectivity for Asterisk users. No fees, pay as you go. They also offer incoming calls for $7.99 per month. See wholesale.voicepulse.com. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone 800 numbers working?
On Thu, Sep 18, 2003 at 07:02:42AM -0700, TC wrote: > >well, i have same problem... > > > >it sounds like nufone is not allowing calling of #800. > >anyone from nufone care to comment? > I have seen nufone die, if the callerid is not > a cid from us 48 try setting your sic to "" > I added SetCallerID and SetCIDName steps before the dial and it works now. Funny, it worked before without these steps. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radio for Music on Hold?
On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote: > Come on people! Fork out $50 for a discman and another few bucks for some > royalty free library music and have that on hold instead.. You're in > control, you know what your callers are listening to, and you're also legal Why go to all this trouble and expense? - skip the Discman and just rip the royaltyfree CD and save the mp3's on the hard drive. (Check the license to make sure you are allowed to do this). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
> Does this thread help? > > http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html > Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x & Asterisk 0.5.0, wonder if it's a softphone's problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware?
I'm currently running firmware version 3.2 on my Cisco 7960. I've seen on the list that several people are running the 5.x latest versions. I've avoided going to higher firmware versions because I'm worried about potential problems or issues with the encryption mechanism used in the later firmware versions. (Once you go to an encrypted firmware version, you can't go back, right?) For those of you who have gone to the newer firmware, what features or benefits have you seen by going with the newer firmware? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Cisco "Color" Phone
I thought you guys would be interested to know: eWeek has a short article about Cisco bringing out a new IP phone: 7970G. It has a high resolution color touch-screen display with support for XML and can act as a mini-browser to allow the development of vertical applications. But get this: the price will be $995. I don't think I'll be getting one any time soon. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Starting Development Perl or Python
Hi guys, >From the drift of the mailing list most people seem to use perl for their AGI scripts. I personally have more experience with Python. Could you please advise why or if Perl is more suitable? Is it faster? Better supported? More documentation? Or any other comments. Our environment is multiple asterisks in multiple cities, we plan to be a PSTN gateway, and also handle IVR work. I could be convinced to use Perl but just am looking for some pointers. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] initial review of Grandstream HT-286 ATA device
The PDF on the website says that this thing supports a downloadable ring-tone. This makes me somewhat suspicious - does this thing generate ringing voltages and actually ring the attached analog phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P
Congratulations to the team at Digium! That is great progress. Peter At 08:37 26/09/2003 -0500, you wrote: >I just got back from Boston where we completed testing of the TE410P for >FCC, Euro, and Australian approvals, and I'm happy to say we passed all >our approvals (including Q.921 and Q.931 layers, i.e. libpri as well as >surges) for both telco and leased line applications. Hopefully we'll have >the official documents soon, but I know there are a lot of you out there >that are happy to hear that. > >Mark > >p.s. We were the *first* independent PRI implementation to come through >that lab! Of all the units they've tested, we're the first to choose the >"build" path on "build vs. buy". > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Vocaltec
Hi all, I've got my dirty hands on (free!) Vocaltec 4-port FXO/FXS gateway. It is used unit, I managed to configure correct IP settings in it but am somewhat at loss how to integrate it into my existing Asterisk network. I have no H323 gatekeeper, no Vocaltec Network Manager software, and am not familiar with Vocaltec products at all. While H323 support is compiling on my Asterisk box I was wondering if anyone has experience with such a setup - potential pitfalls, Asterisk dialstrings etc. I intend to access the box directly via H323 (bypassing gatekeeper) as it's on static & public IP. TIA, Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk friendly IAX/SIP wholesalers in Australia
Bryan, IP Telephonics is developing a VoIP gateway service in Australia. It is not yet operational. If you want to discuss anything please email me offlist. Peter Brown At 23:23 2/10/2003 +1000, you wrote: > its a fair question: does anyone know any? Bryan Nolen Lead >Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and DSL Bandwidth queries.
Hi guys, Don't want to ruffle feathers, but did I see Ratnakar's email address as being @cisco.com. Is Cisco thinking of using Asterisk? Just a thought. Welcome Ratnakar Peter From: [EMAIL PROTECTED] At 14:50 2/10/2003 -0700, you wrote: >Here is my setup > >7960(A)--Firewall/PAT--dsl-Internetdsl--Firewall/NAT---7960 (B) > | | > | | >7960(C)--NAT--cable- -dsl -- Asterisk > >(A) can communicate with (C) only when C is configured with canreinvite=no. The >call gets dropped in few seconds if canreinvite is set to yes for C. >(A) and (B) can communicate fine when both sides have canreinvite=yes. > >Since (C) is not working with canreinvite, traffic goes thru Asterisk server. >This brings the Dsl connection to asterisk to a crawl. It is so bad that even a >idle ssh connection gets disconnected. > >Is it possible to configure C so that reinvite works. If not what kind of a >bandwidth should I have for Asterisk server. Currently it has a upload of 128K. > >The codec currently getting used is ULAW. Even if I configure 7960's to use >g729, show sip channel reports as using ULAW. > >Thanks, >==ratnakar > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialer
Chris are you willing to post the code? Peter At 19:09 2/10/2003 -0400, you wrote: >Hi James-- > >I got a dialer working without too many hiccups about two >months ago. It relies on changes to chan_agent, app_queue, >a PostgreSQL backend, a Tcl-* manager interface, a bunch of >Tcl glue, and some cron jobs. The results for each call are >logged in right through the phone key pad, and the algorithm >for prediction looks at number of agents logged in, average >length of calls, and a magic number the boss man can set to >speed it up or slow it down, plus a couple others I forget. > >Although it relies on some bastardization of the Caller-ID >(who doesn't), it is in compliance with all the latest FCC >rules. A key to making it stable was the recent placement >of extra locks in the queue and agent code. It still gets >some frozen lines, but I blame it on the Zhone, and they >seem to thaw out when you power cycle the POS channel bank. > >I know there was a separate list setup for discussions about >a predictive dialer, and I would like to contribute my code >there but don't remember who made the list or if it has ever >seen any traffic. Not to make a meta-comment on this >thread, but whenever the discussion of a predictive dialer >does arise, it seems to get spit on by those who aren't fans >of the technology. I think that's a real shame as it >represents a huge market for *. I had some moral qualms >about it, too, but they pale in comparison to those I would >have if, say, I was hacking on voicemail for the Pentagon or >rolling out a PBX at Fox News. > >--Chris > > >On Thu, 2 Oct 2003, James Coberly waxed: > >> Hi, >> >> Some time ago there were posts about Predictive dialing. Has anyone >> seen or made any forward progress on this ability? I would be very >> interested in any further info regarding the ability. >> >> Thanks, >> James- > > >-- > >Chris Maj >0xC0051F6A >5EBB 2035 F07B 3B09 5A31 7C09 196F 4126 C005 1F6A >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] runing asterisk and apache
DAve, JUst wondering whether you can disclose the number of users you have on your system and what CPU memory and disks you have. I'm looking to do multiple functions on a single boxen too. Peter At 16:25 6/10/2003 +0200, you wrote: >On Mon, 2003-10-06 at 15:57, listas iPfone wrote: >> Hi All, >> >> I´m thinking in install apache in my asterisk machine to host a litle site. > >Asterisk + Openvpn + Apache + MySql + Postfix + Amavisd + Sophie + local >mirror of Mandrake Cooker. > >This is Linux at work not M$ at play. > >-- >Dave Cotton <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LINEJACK -- OUTGOING CALLS
Hi, I am successful in getting incoming calls for my linejack But, I am getting an error cannot create channel of type Phone Does linejack do outgoing calls in asterisk ? Thanks,PTA
[Asterisk-Users] Calls out of the PBX
hello, I have jsu configured my first Asterisk PBX and it works well. In our company we have alose one Cisco AS5300. How can I mmake Asterisk to forward calls, which have first digit "0" to that Cisco AS5300. Our gateway is allready configured to handla that calls. best regards hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP & IAX behind NAT
I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no audio. Am I missing something in */router setup or such thing cannot be done? I suspect that I need to override local IP address being sent out in SIP/IAX packets, but I see no such option in either iax.conf or sip.conf TIA, Peter
[Asterisk-Users] BOTH UAs behind same FW/NAT
hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BOTH UAs behind same FW/NAT
WipeOut wrote: Peter Hudec wrote: hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards hudecof You will probably have to use "canreinvite=no" in the UA definitions in the SIP.conf for those two phones.. I have this so Also make sure you have enough badwidth between the UA's and the Asterisk server to sustain 2 calls.. 100Mbit/s full duplex Later.. Still the same ;( hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP & IAX behind NAT
OK, I got IAX to work. Which SIP bone I should break to force it into working? ;) Peter > IAX should work well behind NAT without further configuration than just > a single port forward. RTP based protocols such as MGCP, SIP and H.323 > require helper agents on the NAT box to work, although SIP can be forced > to work by slightly breaking it. > > roy > > On Mon, 2003-10-27 at 10:00, Peter Zeltins wrote: > > I'm trying to set up * server behind NAT. The box is set up as DMZ in > > my DSL router, i.e. all incoming connections without explicit port > > mapping are forwarded to *. So far I'm unable to get this setup to > > work for either IAX or SIP (tried IAXComm & XLite softphones on public > > IP address). Data seems to come in fine (IAX/SIP debug shows message > > interaction taking place), but there is no audio. > > > > Am I missing something in */router setup or such thing cannot be done? > > I suspect that I need to override local IP address being sent out in > > SIP/IAX packets, but I see no such option in either iax.conf or > > sip.conf > > > > TIA, > > Peter > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI & Asterisk Redundancy/Fail-Over
Do you think they would cost much to make them? I'd be keen to get them for E1s. I think they could be sold with an E1PRI card. Especially if VoIP PABX's take off . Peter At 20:12 27/10/03 -0500, you wrote: look for a T1 failover switch. (cheap as dirt on ebay, mine was $7.50, - yes really, I got the decimal in the right spot, hard to find an empty rackmount box that is cheaper.) Basically it looks like a Y in the T1. It contains a csu/dsu on each interface. It decodes the t1 signals, and then re-encodes them using its reframer. It looks at the signal enough to know if one of the branches of the Y is in alarm, lost completely, etc. You can specify the criteria when to switch and when to switch back on restoration of original circuit. These things are actually designed to take dual t1's coming in and select the right one to feed to CPE, but like you I use mine the opposite way around to connect a working set of CPE to the inbound T1. I use for a data t1, but voice/pri is all the same as far as this thing is concerned, it only looks at the clocking, and alarm, etc., not the actual data contained in the serial stream. At 08:02 PM 10/27/2003, you wrote: Hi, I'm trying to put together an * system with extremely high up time. The system as it stands now is 1 dual p4 with raid, redundant power, etc.. and a T100P card. I would like to get a second similar or identical box with another T100P card. I have 1 PRI, how do I get the second box to take over communication with the PRI should the first box fail? Of course I'd like this to be done without manually unplugging the PRI from box 1 and plugging it into box 2. Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BOTH UAs behind same FW/NAT
thanks for explanation. It does not solves this problem, but another one :) best regards hudecof Olle E. Johansson wrote: Philipp von Klitzing wrote: You will probably have to use "canreinvite=no" in the UA definitions in the SIP.conf for those two phones.. In your case you want the opposite: canreinvite=yes A try to sort out these kind of opposite messages: When asterisk connects two SIP phones, it tries to be in the middle of the media path, to have the RTP stream go through Asterisk. This way, Asterisk may send early media and error messages over audio. When the call is connected, asterisk can send SIP re-invites and change the path of the media stream, so that media flows directly between the two phones instead of going through Asterisk. This is canreinvite=yes In your situation, for calling between the phones, you propably don't want the media stream to go SIP UAC -> NAT -> Asterisk -> NAT -> SIP UAS (canreinvite=no) Instead SIP UAC -> SIP UAS (canreinvite=yes) However, I'm unsure if you can have a canreinvite=yes, since you may want asterisk to be in the media path when calling outbound... Also note that some devices does not support SIP re-invites (according to the Asterisk handbook) I'm a bit on thin ice here, so if I'm wrong - please, list, correct me so we can sort this out. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Consultants/Companies in Indianapolis?
Are there any companies/consultants in the Indy area that are Asterisk experts? Please contact me via email. THanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfering, conferencing
hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
> That's for pointing out Walter Snel "hack". > Adding his two additional features would not be > hard. http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html Any idea when these "hacks" will appear in CVS? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfering, conferencing
http://www.asterisk.org/index.php?menu=features - Call features - Call Transfer WipeOut wrote: Peter Hudec wrote: hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof You won't find in in any Asterisk manual becasue these are not features of Asterisk, they are features on the phone.. The phone needs to support transfer and if you want conferencing without using "meetme" then you need a phone that supports conferencing.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfering, conferencing
thanks, you didn't make me happy :( hudecof WipeOut wrote: Peter Hudec wrote: http://www.asterisk.org/index.php?menu=features - Call features - Call Transfer Yes, provided your phone supports "transfer" or you use the "t" or "T" options on your dial string and then use the # key to transfer.. CLI> show application dial ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP behind NAT problem
Hello, my next problem is with SIP device behind NAT. First few seconds of the call are OK. Astrisk is sending the packets to the public IP address of the FW/NAT (62.152.224.3). But this change in 10 second and packets are send to the my public addres.(192.168.1.163). in the sip.conf for the phone(X-Lite) is [998] type=friend username=998 secret= canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=998 nat=1 callerid=0650199802 Can anybody explain me, why the dest IP will change from public one to the private one ? best regards hudecof - CUT - DEBUG[540687]: File rtp.c, Line 388 (ast_rtp_read): RTP NAT: Using address 62.152.224.3:8000 DEBUG[540687]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 4160, ms is 540 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 DEBUG[540687]: File rtp.c, Line 388 (ast_rtp_read): RTP NAT: Using address 62.152.224.3:8000 DEBUG[540687]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 6576, ms is 842 DEBUG[540687]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 7848, ms is 1001 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 DEBUG[540687]: File rtp.c, Line 343 (ast_rtcp_read): RTP NAT: Using address 192.168.1.163:8001 - CUT - -- mail: [EMAIL PROTECTED] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channelbanks for use in europe (Sweden)
At 21:39 29/10/03 +0100, you wrote: Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would be intresting to know how the channelbank sends the callerid (DTMF or FSK) if it sends any callerid? Best regards, Lars Fredriksson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Just remember part of the design of the TE410P is that you can use T1 channel banks (you only get 24 ports) , if you buy these in America they are significantly cheaper than E1 channel banks. Just assign one of the incoming ports on the TE410P to be T1 not E1. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
> http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html > > > > Any idea when these "hacks" will appear in CVS? > > We should all hope "never". That's why you call it a "hack" > because it works for only one very specific case and would break > SIP under Astrisk for most people. It even breaks calls > between Asterisk and local SIP phones. > > Now the trick is to write some code that desides if the trick is > to be used or not for each call by comparing the IP address of > Asterisk and the called SIP phone. > > You migh want to experiment with it and report results. Well, I happen to be one of those very specific cases... ;) and looks like will have experiment with it myself. Although I'd hate to re-invent the wheel. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
> > Well, I happen to be one of those very specific cases... ;) and looks > > like > > will have experiment with it myself. Although I'd hate to re-invent > > the > > wheel. > > Checking e-mail this morning it looks like we have two independent > "fixes" that both do what has been suggested in this thread. > > No need for a third except posibly a merge of the two. Would you care to elaborate? I don't see anything in asterisk-users, and no mention of SIP-behind-NAT in CVS changelog... maybe I should start subscribing to asterisk-dev Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Dan, Looks great. Are you planning to release this with GPL? Peter At 22:21 2/11/03 +0200, you wrote: Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device; - GSM codec support; - advanced phonebook(search/add/replace/delete); - 12 memories with one click access (just click one of the 12 buttons to directly dial the number); - you can memorize IAX type addresses then call them with a click of a button; - 99 memories with two keys access (Mxx); - unlimited number of memory locations (just limited by your HDD capacity), accessible through the phonebook interface; - can dial directly from the phonebook; - can use separate audio device than the default one (you can play MP3's through your soundcard/speaker and use an USB headset fort phone purpose); - digital VU-meter (you can enable/disable it); - digital volume control (Vol UP / Vol Down); - redial/callwaiting callerID functionalities; - can switch between two calls; - out/in/missed/rejected/all calls list; - missed calls indicator; There is no help file available for the moment. I hope to finish it in a couple of days. Please send me your feedback. Thank you, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX hardphones? anyone?
Markster, Do you know where the product referred to by Steven is now? Can it be purchased? If so from whom? Peter At 07:42 3/11/03 -0600, you wrote: On Mon, 2003-11-03 at 07:07, Roy Sigurd Karlsbakk wrote: > hi all > > anyone that've heard of any working IAX hardphones yet? During Phreaknic, Mark showed the IAXY(sp?) a small maybe 2"x3" board that was a single analog FXS port to IAX adapter. It was impressive that he was able to come into the con room, plug into the ethernet and without reconfiguring anything, he started making phone calls out of his office system. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod...
Dan, Hence forth you will be called Dan the man! Peter At 23:20 3/11/03 +0200, you wrote: Hi Dave, - Original Message - From: "David J Carter" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, November 03, 2003 10:28 PM Subject: RE: [Asterisk-Users] DIAX Soft phone v0.9.1 is available for downlaod... > Hi Dan, > > Just downloaded 0.9.1. Works fine on test set up internally. > Good to hear that..;-) > I get my WAN IP dynamically and have used DynDNS.org for updating a URL for > the home network. Could the registration look for this rather than a fixed > IP address? I think that this will not be possible with the actual LIBIAX library... > > Regards, and keep up the good work for us non techies to use. > I'll do it! Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
