Re: [asterisk-users] OpenVPN/SNOM 820: a review.
On Thu, Feb 18, 2010 at 03:05:14PM -0500, Ken D'Ambrosio wrote: Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as Thanks for the review, I was wondering if snom's mass deployment tools they have used in their other phones work with the 820's and openvpn ? [snip] One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken -- You're free. And freedom is beautiful. And, you know, it'll take time to restore chaos and order -- order out of chaos. But we will. - George W. Bush 04/13/2003 Washington, DC 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] Losing local SIP phones when internet goes down?
On Fri, Feb 05, 2010 at 01:21:38PM +, Nikhil Nair wrote: Hi again, OK, I've now installed a local caching nameserver, but don't see any change at all. Just to add to the discussion, my setup I was using a local bind9 server for local/authorative and recursive queries I think from memory it was asking for a sip address to register with and the record had a ttl or 600 (5min) so could expire very easily. can I suggest maybe whilst eth1 (the internet link) is down, stop and restart asterisk with logging and check to see what fails IN detail, what I did: [snip] 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] Losing local SIP phones when internet goes down?
On Thu, Feb 04, 2010 at 09:52:35PM -0600, Warren Selby wrote: On Thu, Feb 4, 2010 at 9:20 PM, Nikhil Nair nn...@pobox.com wrote: No, again, I can cut off the internet altogether with ifdown eth1, and the SIP phones (via eth0) continue to work fine, as does the Zap channel. It's only if eth1 is up but the ADSL router is down (or, indeed, the phone line is down), that the problem arises. N. It sounds as though something is wrong with the RTP routing. If you ifdown the connection, you're deleting the routes. If the ADSL modem goes down, your server still thinks it has a route out to the internet. I've noticed a similar issue on one of my client's installations. They lost their internet connection and the phone system went haywire. As soon as it was restored, everything returned to normal. What happens if you manually execute just ifconfig eth1 down? I haven't the route should dissappear tried this, I'm just curious if it will also delete the route, or will it leave it there? If the route is still there, what happens to your phone connections? Another idea - what happens when you bind your SIP connections to the internal IP address (using bindaddr=10.9.8.1 in the general section of your sip.conf file) , and then unplug the modem? I'm sorry I'm unable to offer an exact solution, but I'm very interested in any results you may come up with. maybe try tcpdump on the inside interface of the debian box limit to just packets with a sip phones address. What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again -- As far as the legal hassling and wrangling and posturing in Florida, I would suggest you talk to our team in Florida led by Jim Baker. - George W. Bush 11/30/2000 Crawford, TX 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] Siemens Gigaset + Asterisk Query?
On Sat, Jan 23, 2010 at 08:08:28AM +0100, Philipp von Klitzing wrote: Hi! I was wondering if you can use the base station as a outbound pots connection for asterisk. I currently have a tdm410 to do fxs/fxo ports and would like to get rid of it, I used to use a spa3102, but it only had 1 fxo (telephone connector). I like the idea of the siemans but I would like to control the pots fallover from asterisk. if not the siemans are there any other bases that would fit the bill ? The AVM Fritz!Box 7270 could do the job, but I am not sure if you can get that with an English language web interface. DECT and SIP registrar (for LAN only) are available with a recent firmware. This box might be a bit oversized for what you are trying to do, though. German wife :), nice (the box) but not exactly what I wanted, I like seperate adsl modem, linux firewall, dlink ap, lets me control things Philipp -- Keybuk Perl 6 scares me doogie you can name your operators anything. the name here is the string '~|_|~' * Lo-lan-2 runs away screaming Keybuk it looks like a diagram of a canal lock :) jaybonci japanese smiley operators? nickr ^_^ -- in #debian-devel 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] Snom vs Polycom
On Fri, Jan 22, 2010 at 05:06:17PM -0300, Andrew Latham wrote: I have worked on many snom phones over the years I have never had a snom phone go bad... I have had about 10 in the last 12-18 months, I had 1 with a fault hand set plug - the reseller replaced it. Other wise they have been great. I have repaired stuck screens and overheated sticky bits but all in all snom are great phones. I recently showed my personal phone to some people including a VoIP engineer that fell in love with my snom 360... It has scratches and dents from use and abuse but the real shocker was when I turned the phone over and showed them the date of manufacture ... 2004... I am working with some Polycom phones right now. They look ok. I don't hear any better speaker that people talk about. They are troublesome to administrate or provision as a single phone. provision the snom is a pain to setup - well you have to do a lot of work, but its work while - lots of documentation on the snom site on mass distribution - makes life simple I would recommend it for anything more than 3 phones (all you need is a deamon provided by snom written in perl and a http server) [snip] 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] Siemens Gigaset + Asterisk Query?
Hi I was wondering if you can use the base station as a outbound pots connection for asterisk. I currently have a tdm410 to do fxs/fxo ports and would like to get rid of it, I used to use a spa3102, but it only had 1 fxo (telephone connector). I like the idea of the siemans but I would like to control the pots fallover from asterisk. if not the siemans are there any other bases that would fit the bill ? Alex On Fri, Jan 22, 2010 at 07:14:56PM -0500, John Hurley wrote: From my experience, unless you have another base station for sets you would want to configure separately it is not possible. I may be wrong, hopefully. Sent from my Android phone -Original Message- From: Alan Lord (News) [alansli...@gmail.com] Received: 1/22/10 7:01 PM To: asterisk-users@lists.digium.com [asterisk-us...@lists.digium.com] Subject: [asterisk-users] Siemens Gigaset + Asterisk Query? When you configure the Siemens gigaset handsets (I have S685IP), there is a single option for all handsets to use either the POTS interface or VOIP as the default outbound destination - you then need to add a dial suffix if you want to use an alternate outbound route. Does anyone have any suggestions as to how to make just *one* of the DECT handsets only use the POTS but others default to their Asterisk SIP subscriptions? The POTS is on the Gigaset base station not on the Asterisk server. TIA. Al -- If you found somebody that had information about an attack on America, you'd want to know as best as we can to find out what the facts are. - George W. Bush 12/12/2005 Philadelphia, PA 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] Looking at Asterisk for 8000sq/ft residential use
On Tue, Dec 29, 2009 at 11:30:21PM -0500, C F wrote: Before I start I am a Panasonic certified dealer AND I have installed over 100 Asterisk systems that are in production. That said for your application use Panasonic, DONT use Asterisk. Use the Panasonic KX-TDA50G. Supports up to around 50 ports. I initial started the email to point out that this is a non commercial project. But after a quick google, this looks like a nice unit for around the US$600 mark (101Phones.com) In addition to Analog and their proprietary Digital phones they support: * SIP * DECT Wireless with their unique Wireless CellSites (so there are no dead spots). * Door boxes with door realease * External relays * CTI software * Built in MOH port * Built in external paging port Dont use the TAW848 as it uses analog proprietary phones which in addition of having just 12 programmable buttons it must have 2 pairs to work. Its options are also limited, as well as it's been discontinued. [snip] 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] TDM 400 hardware(?) issue
Hi I use one of these http://www.soekris.com/net5501.htm fairly cheap to buy and to run, I have a tdm410 in there and it has worked flawlessly I am running debian i386 on the box - it also doubles as my firewall/router/vpn/adsl box. I do have one problem with the box (but I have seen on other boxes as well), when the adsl drops it causes a ip loop (6to4 routing), which hoggs irq's this in turn causes a software bug in the dahdi/zaptel driver which means I have to reload the dahdi/zaptel module in asterisk - easy to capture (do it with ppp-up) Alex On Tue, Dec 22, 2009 at 11:53:02AM -0700, Greg Woods wrote: the machine will lock up because the TDM board or the Dahdi driver goes south. /var/log/messages starts filling up with repeated messages: kernel: TDM PCI Master abort [snip] Thank you to everyone who has taken the time to reply. 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] iphone client app
On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote: IAXDIAL is free on app store works great on WiFi even true NATs but seem blocked for GPRS. ta HB [snip] Well I have a 3gs - will tell you how that goes. installed (non cracked), but I am on wifi now, easy to configure and sounds okay (initial use). My only concern with it - it's not just a voip client, its many other things as well. not sure if I want to be a fring user as well as all the other memberships I have :) decided against siax - have to pay for the base model. 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] iphone client app
On Tue, Dec 15, 2009 at 09:14:16PM +1100, Alex Samad wrote: On Tue, Dec 15, 2009 at 08:33:56AM +0100, hbk wrote: [snip] My only concern with it - it's not just a voip client, its many other things as well. not sure if I want to be a fring user as well as all the other memberships I have :) upon further investigation I am going to use siphon - a free iphone sip client - works well and is only a sip phone :) [snip] 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] iphone client app
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote: Gavin Spurgeon gspurg...@dageek.co.uk writes: iSip (£2.39) http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8 I have been very impressed by the audio quality from iSip, at least from the other end so to speak. It shares the basic flaw of not being able to run in the background with every other iPhone app. They try to can't you use backgrounder ? mitigate that problem with their Push service, if you give their server your passwords and allow them to access your Asterisk... /Benny 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] iphone client app
On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote: Fring, it's free and works perfectly with an Asterisk server.. thanks On 13 Dec 2009, at 10:15, Alex Samad wrote: Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex 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] iphone client app
Well I have a 3gs - will tell you how that goes. decided against siax - have to pay for the base model. installed fringe, but no voip over 3g, have to wait till i get home, but it registered with my asterisk server so .. I am looking for the hacked fring.ipa which allows voip over 3g, just so I can try I am in Aus, and i bought the phone outright so no lock in. A On Mon, Dec 14, 2009 at 07:25:37PM -0500, Mike Bessette wrote: I find that Siphone works great on the iTouch. Tried it with my own asterisk box as well as Callcentric and MagicJack and it was very clear and stable. Haven't played with it since the last firmware update though as the update removed support for 3rd party headsets . On 12/14/09, Alex Balashov abalas...@evaristesys.com wrote: I personally have not had much luck with these softphones because the iPhone 3G seems to be underpowered and just doesn't run them well enough to sustain good voice quality, irrespective of wifi network conditions. I could be mistaken, though. It's not going to happen over ATT's 3G network -- not down here in the southeastern US, as far as I can tell. So I can't speak to whether voice works over 3G. -- Sent from mobile device On Dec 14, 2009, at 6:57 PM, Alex Samad a...@samad.com.au wrote: On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote: Fring, it's free and works perfectly with an Asterisk server.. thanks On 13 Dec 2009, at 10:15, Alex Samad wrote: Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex ___ -- 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 -- That I'm running on my dad's name... I'm proud of my dad... I reconciled my love for my dad a long time ago. - George W. Bush When asked by Larry King 'What do people misunderstand about you most?' 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
[asterisk-users] iphone client app
Hi Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone Alex 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] newbie question
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote: On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, [snip] 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less Or, you can use the script command to capture the output to a file so you can refer to it as needed. I find screen helpful here you can set the scroll back buffer to a large number and you can detach the running screen from the console to reattach some time later. my default scroll back buffer is set to around 1000 usually enough to capture what I need, plus you can cut paste between screens script is also useful to capture the console log to a file when you are trying to debug a call and your console output looks like a broken fire hydrant. -- It's amazing I won. I was running against peace, prosperity, and incumbency. - George W. Bush 06/14/2001 speaking to Swedish Prime Minister Goran Perrson, unaware that a live television camera was still rolling. 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] ip source aware Authentication
On Mon, Nov 16, 2009 at 08:17:27AM -0600, Kevin P. Fleming wrote: Alex Balashov wrote: As far as I know, Asterisk has no way to restrict the content of the domain portion of the Contact URI. However, most commercial SBCs should have a way to filter this, and it is highly recommended that you do so. It does actually; we added it to address this very issue. This also keeps people from registering Contact URIs using RFC-1918 addresses that happen to live inside the ITSP's network, or other nastiness. for use newbies could you give an example - or point to one -- Like you, I have been disgraced about what I've seen on TV that took place in prison. - George W. Bush 05/13/2004 Parkersburg, WV 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] Best Firewall Suggestions?
Interesting on my asterisk box I have installed Virtualbox and I run my firewall/router in a vm, stripped down linux box with iptables, I have snapshoted the image to a working image. it only does ip forwading/vpn/iptable stuff ends up being a low foot print, 256M + 8G / Alex On Tue, Oct 13, 2009 at 09:53:54PM +0200, Grygoriy Dobrovolskyy wrote: Allmost your solutions require second server or some hardware, why do you use shorewall ? Its a iptables rule generator with a friendly config files. Mine was up and running in 30 min or reading some docs. And you can trace all traffic live. Good day. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A prisoner of war is a man who tries to kill you and fails, and then asks you not to kill him. -- Sir Winston Churchill, 1952 signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk dialplan to share fax line
Hi I look after a site which is using asterisk and a vsp for its primary telco needs, so I am on holiday for a week and of course some jack arse has decided to reboot the server and something has gone wrong with the remote access. Now they don't have any internet and i can't fix it remotely. Bad for them because they don't have any inbound call capabilities, they do have outbound via the fallback pstn line connected to the asterisk box, which is also a fax line - its primary useage. So what i would like to to when i get back is to setup the pstn line to be a backup inbound line as well. shared fax and normal phone line. How can I work the dial plan to 1) pickup the call 2) if its a fax hold the line till the fax machine takes over or will now that asterisk has picked up the line I have a tdm410 card with fxo + fxs ports. do I attach the fax machine to the asterisk server and just treat it as a extension, but letting all its outbound calls go through the pstn line on the tdm card - will the fax work that way ? the question is still how to I deal with inbound calls/faxes ? Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] system cmd + fax line
Hi I have a site that has asterisk install with a tdm410 one port is connected to a pstn that is used as a backup outbound line when/if the internet/voip is unavailable. Currently my dial plan for this line is to ignore it, I just basically do a s,1,noop s,n,wait (60) s,n,hangup what I would like to do is if I call from a certain number be given an option to run some commands or for the beginning just run a simple command via system something like s/041100,1,goto(specialcontext,s,1) s,1,noop s,n,wait(60) s,n,hangup [specialcontext] s,1,system(do a linux command) is that about right. Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium transcoding card
On Thu, Sep 24, 2009 at 05:32:24PM -0500, Michael Graves wrote: On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote: Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent generic board out there that I could be investigating? It would be such a shame to waste a PCI slot that could have a voice-card in it. Thanks, Steve Looking ahead, but not that far, I'd like to see that card extended to transcode between wideband codecs (G.722, G.722.1, G.722.1C, AMR-WB SILK) in addition to G.729a and G.723.1. I would like to see something to plug into a tdm410 it could take up one of the ports of the 4 port card (not sure if it exists already) - that would be cool Michael -- I want to thank you for the importance that you've shown for education and literacy. - George W. Bush 04/13/2005 Washington, DC signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/WiFi handsets?
On Wed, Sep 23, 2009 at 09:39:09AM -0700, mgra...@mstvp.com wrote: I had a good experience with that Polycom/Spectralink phone. Very rugged as you say. The experience did highlight the weaknesses in consumer Wifi AP, which reinforced my commitment to continue using DECT around my office. I concur, I settle on the snom M3, but i did not have any requirements to leave the office with the device. DECT seems to drain the battery a lot less then Wifi Michael Original Message Subject: Re: [asterisk-users] SIP/WiFi handsets? From: Jason Baker jba...@glastender.com Date: Wed, September 23, 2009 10:02 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Ken, I did lots of research on this for my VoIP deployment here where I work. We have a huge manufacturing floor and all the supervisors have wifi phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged little phone with great sound quality and some good features. We use a managed switch to create seamless wifi coverage over all of our AP's. Provisioning the phone is pretty easy, but no web browser if you were planning on using the phone to travel with, some hotels require login for internet access. I also tried a clamshell wifi SIP phone by D-Link. This phone actually works really well, but we had some minor issues with it so we went with all Spectralink phones. But the D-Link phone would be good choice if you plan to take your wifi phone on the road. I also tested the Linksys WIP330 which I thought was a terrible phone. Very difficult to use. Good luck. http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html http://www.dlink.com/products/?pid=485 http://www.voipsupply.com/linksys-wip330-na Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Kenhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- And, most importantly, Alma Powell, secretary of Colin Powell, is with us. - George W. Bush 01/30/2003 Washington, DC signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with dialplan - gotoif ?
Hi
This is the output from show dialplan dial-sipmnf-sippt-pstn
[ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ]
's' =1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config]
2. Dial(SIP/${ar...@${sipmnf},${ARG2},${OUTBDIAL})
[pbx_config]
3. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]
4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok) [pbx_config]
[pt] 5. Dial(SIP/${ar...@${sippt},${ARG2},${OUTBDIAL})
[pbx_config]
6. Set(GLOBAL(FOUNDME)=${DIALSTATUS}) [pbx_config]
7. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pstn:ok)
[pbx_config]
[pstn] 8. Dial(${PSTN}/w${ARG1},${ARG2},${OUTBDIAL}) [pbx_config]
[ok] 9. Goto(dialer-exit,s,1(${ARG3}) [pbx_config]
-- Executing [0296332...@in-uniden:1] Verbose(DAHDI/1-1, 1,Dialing
0296332828 normal) in new stack
Dialing 0296332828 normal
-- Executing [0296332...@in-uniden:2] Log(DAHDI/1-1, Notice,Dialing
0296332828 normal) in new stack
[Sep 22 18:20:42] NOTICE[18347]: Ext. 0296332828:2 @ in-uniden: Dialing
0296332828 normal
-- Executing [0296332...@in-uniden:3] Gosub(DAHDI/1-1,
dial-sipmnf-sippt-pstn,s,1(0296332828,70)) in new stack
-- Executing [...@dial-sipmnf-sippt-pstn:1] Verbose(DAHDI/1-1, 1,Dialing
0296332828 on mnf pt pstn) in new stack
Dialing 0296332828 on mnf pt pstn
-- Executing [...@dial-sipmnf-sippt-pstn:2] Dial(DAHDI/1-1,
SIP/0296332...@mynetfone-09105023,70,WKT) in new stack
== Using SIP RTP CoS mark 5
-- Called 0296332...@mynetfone-09105023
-- SIP/MyNetFone-09105023-097809e0 is making progress passing it to
DAHDI/1-1
-- Got SIP response 486 Busy Here back from 125.213.160.81
-- SIP/MyNetFone-09105023-097809e0 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@dial-sipmnf-sippt-pstn:3] Set(DAHDI/1-1,
GLOBAL(FOUNDME)=BUSY) in new stack
== Setting global variable 'FOUNDME' to 'BUSY'
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY =
CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
-- Executing [...@dial-sipmnf-sippt-pstn:5] Dial(DAHDI/1-1,
SIP/0296332...@pennytel-8889186044,70,WKT) in new stack
i believe i have captured the relevant logging from the console. my problem is
with Gotoif statement
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY =
CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
from my understanding BUSY != CHANUNAVAIL, therefor it should have jumped to ok
which is s,9.
What have I missed !
thanks
Alex
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Re: [asterisk-users] Problem with dialplan - gotoif ?
On Tue, Sep 22, 2009 at 07:57:56AM -0400, Leif Madsen wrote:
Alex Samad wrote:
4. GotoIf([${DIALSTATUS} = CHANUNAVAIL]?pt:ok)
[pbx_config]
i believe i have captured the relevant logging from the console. my problem
is with Gotoif statement
-- Executing [...@dial-sipmnf-sippt-pstn:4] GotoIf(DAHDI/1-1, [BUSY
= CHANUNAVAIL]?pt:ok) in new stack
-- Goto (dial-sipmnf-sippt-pstn,s,5)
from my understanding BUSY != CHANUNAVAIL, therefor it should have jumped
to ok which is s,9.
What have I missed !
You have missed the leading $ in front of the $[...] portion.
Add the $, and things should start working as you expect.
thanks the bloody obvious :)
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] call-limit on dahdi channel
On Thu, Sep 17, 2009 at 12:02:16PM +0300, Tzafrir Cohen wrote: On Thu, Sep 17, 2009 at 08:18:13AM +1000, Alex Samad wrote: Hi how do i set the call-limit on a dahi line - its connected to the pstn network - shared fax line. How do i tell asterisk not to send more than 1 call there ! Asterisk will not send out more than one call on that line. You want to avoid calling if someone else is calling on a fax machine connected to the same line? yes signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure a coverage pathfor anextension???
On Wed, Sep 16, 2009 at 12:24:22PM -0700, Steve Edwards wrote:
On Wed, 16 Sep 2009, Danny Nicholas wrote:
I'd try this:
- exten = 4000,1,Dial(SIP/4000,20,ikKtT)
- exten = s-NOANSWER,1,Dial(SIP/4001,20,ikKtT)
- exten = s-NOANSWER,2,Voicemail(4000)
- exten = s-BUSY,1,Dial(SIP/4001,20,iKkTt)
- exten = s-BUSY,2,Voicemail(4000)
- exten = h,1,hangup
Don't you need a goto(s-${DIALSTATUS},1) in there somewhere?
I am curios as well, what tell it to do the jump
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[asterisk-users] call-limit on dahdi channel
Hi how do i set the call-limit on a dahi line - its connected to the pstn network - shared fax line. How do i tell asterisk not to send more than 1 call there ! Alex -- Drug therapies are replacing a lot of medicines as we used to know it. - George W. Bush 10/18/2000 St. Louis, MO during the third presidential debate signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question of resiliance
Hi
I am in the process of move a company from pstn to an asterisk setup.
They had 2 pstn lines - only really needed a max of 2 previously.
Now I have installed a tdm410 to handle the cross over from pabx to voip
handset. this has been done, the tdm is now just used to provide a
backup pstn line - only used as a last resort for outgoing calls - as
its shared with a fax line.
I use 2 voip providers one primarily for local and std and the other for
mobile calls, all though they can be backups for each other. and then
the pstn line
originally i had (and still till i get around to changing them) macros
to dial each interface and macro's that handle trying each voip and then
pstn in order, this is based around dialstatus (hackled from the
website).
My problem is when I have had long adsl problems (pain in the back side,
back to base alarm system) eventually asterisk seems to not want to talk
to the voip phone - nor does it allow any calls to be placed.
my guess is this
srvlookup=yes
which I had until recently - decided I don't really need it, my guess it
asterisk has a problem with name resolution and get stuck, but stuck
such that call processing can't happen.
snippet of my macro
[macro-dial-sipmnf-sippt-pstn]
;
; Enter with these
; ARG1 = number to dial
; ARG2 = timeout value
; ARG3 = flag determines if hangup or return on no answer
; HR = hangup and return (default)
; RT = return without hangup (must set)
;
; Returns with FOUNDME = DIALSTATUS
;
;
exten = s,1,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,2,Dial(SIP/${ARG1}${SIPMNF},${ARG2})
exten = s,3,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,4,GotoIf([${DIALSTATUS} = CHANUNAVAIL]?5:12)
;
exten = s,5,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,6,Dial(SIP/${ARG1}${SIPPT},${ARG2})
exten = s,7,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,8,GotoIf([${DIALSTATUS} = CHANUNAVAIL]?9:12)
;
exten = s,9,Set(GLOBAL(FOUNDME)=ANSWER)
exten = s,10,Dial(${PSTN}/${ARG1},${ARG2})
exten = s,11,Set(GLOBAL(FOUNDME)=${DIALSTATUS})
exten = s,12,Goto(s-${DIALSTATUS},1)
;
Alex
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Re: [asterisk-users] Question of resiliance
On Sun, Aug 30, 2009 at 06:49:06PM -0700, Kyle Kienapfel wrote: It's been my experience that when asterisk does a dns lookup, for externhost or to do a SIP register, it blocks the whole server. Not sure if 1.6 has that problem or just 1.4 though as my internet has been stable while im awake these days just for clarity I am running on 1.6 [snip] Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Breaking news, but what happened? 11.000 channels on one server
On Tue, Aug 25, 2009 at 07:30:08PM +0200, Olle E. Johansson wrote: 25 aug 2009 kl. 18.50 skrev John A. Sullivan III: On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote: 25 aug 2009 kl. 16.20 skrev Olivier: [snip] mode in Linux on any old switch and it works reasonably well other than for some excessive ARP traffic. However, as we found out the hard way when building our Nexenta SAN, bonding works very well with many-to-many traffic but does very little to boost one-to-one network flows. They will all collapse to the same pair of NICs in most scenarios and, in the one mode where they do not, packet sequencing issues will reduce the bandwidth to much less than the sum of the connections. Take care - John That is very good feedback - thanks, John! Which means that my plan B needs to be put in action. Well, I did create a new branch for it yesterday... ;-) any thoughts of different media like 10G ethernet or infiniband ? /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I mean, these good folks are revolutionizing how businesses conduct their business. And, like them, I am very optimistic about our position in the world and about its influence on the United States. We're concerned about the short-term economic news, but long-term I'm optimistic. And so, I hope investors, you know -secondly, I hope investors hold investments for periods of time -that I've always found the best investments are those that you salt away based on economics. - George W. Bush 01/04/2001 Austin, TX signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stutter playback
Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again . Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stutter playback
On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote: On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote: Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again . Alex I would check cpu, diskpace, memory, I/O, network wasn't that, I have a alarm system on the backup pstn line, seems like there is something wrong there, cause when I remove the alarm system from the equation everything seems okay, so I am guessing it was causing some problem on my tdm410 card. strange thing is i did not see any spikes on io , cpu, network... Alex -- Think of it! With VLSI we can pack 100 ENIACs in 1 sq. cm.! signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote: Kevin P. Fleming wrote: Jeff LaCoursiere wrote: On Tue, 18 Aug 2009, Kevin P. Fleming wrote: [snip] [snip] Here's my $0.02. If you don't want an echo canceller, specify echocanceller=none,x-y and have dahdi_cfg print a warning (at any verbosity level) when an echo canceller is not specified for a channel. Personally, I would also like to see an option that says Use the hardware canceller, like echocanceller=hw,x-y. This would have the Yep, thumbs up on the hw = hardware echo can.. added benefit of being able to display an error/warning when the hardware canceller is specified but no hw canceller is present. It goes against my grain to not specify a canceller to mean use a harware one if it happens to exist. -Dave -- They said, 'You know, this issue doesn't seem to resignate with the people.' And I said, you know something? Whether it resignates or not doesn't matter to me, because I stand for doing what's the right thing, and what the right thing is hearing the voices of people who work. - George W. Bush 10/31/2000 Portland, OR signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]
On Sat, Aug 15, 2009 at 10:58:07PM +1000, Lee, John (Sydney) wrote: I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no problems since Dec last year. We are using Digium TE412P to connect to [snip] Pid: 0, comm: swapper EIP: 0060:[,C0417911.] CPU: 1 EIP is at smp_call_function+0x99/0xc3 EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1) EAX: 0002 EBX: ECX: 0001 EDX: 00fb ESI: 0003 EDI: EBP: c0417ae0 DS: 007B ES: 007b CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0 [c0417ae0] stop_this_cpu+0x0/0x33 [c041794e] smp_send_stop+0x13/0x1c [c0425bcf] panic+0x4c/0x16d [c040da17] intel_machine_check+0xf9/0x146 Have you installed mce - the machine check error logger to see what this is, you might get more relevant information from there. I have also seen this cpu stuck when my adsl goes down and i get a very short lifed ipv6 routing looping - sucks up cpu and doesn't let it intr which starves the other cards. Alex [c040d91e] intel_machine_check+0x0/0x146 [c0403ccf] error_code+0x39/0x40 [c0403ccf] mwait_idel+0x25/0x38 [c0522200] acpi_processor_idle+0x154/0x3b4 [c0403c90] cpu_idle+0x9f/0xb9 === Q1. A strange thing is I could not find this error message in /var/log/messages or dmesg. The soft lockup error message can only be found on the machine itself. Q2. Could it be kernel incompatibility problem? However, we did not ever change anything since it was installed. Q3. From the error message, how do I know it is a software (kernel?) or hardware problem? I would appreciate if someone could give me any suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Quick, you must come with me, you're in great danger! Why? Because I'll kill you if you dont. (Sourcery) signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan tips
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote: Here's how I think your dialplan should look: exten = 101,1,Ringing exten = 101,2,Answer() exten = 101,3,Dial(SIP/quentin,10) exten = 101,n,VoiceMail(1...@default,u) exten = 101,n,Playback(vm-goodbye) exten = 101,n,Hangup() exten = 101-BUSY,1,Playback(busy) exten = 101-BUSY,n,Wait(3) exten = 101-BUSY,n,VoiceMail(1...@default,b) exten = 101-BUSY,n,Playback(vm-goodbye) exten = 101-BUSY,n,Hangup() Hi I have a question about this dialplan, why does the dial do a jump to 101-DIALSTATUS, is there a goto 101-DIALSTATUS missing ? Alex [snip] signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TDM400P in Soekris net5501-70?
On Mon, Jul 20, 2009 at 05:58:51PM -0400, Brian McEntire wrote: Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! yep thats right, I run the local ssd drive and the tdm from there. Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -END PGP 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 -- Scully: I've heard the truth. Now what I want are the answers. The X-Files: Paper Clip 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] Digium TDM400P in Soekris net5501-70?
On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. 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
[asterisk-users] how to enable dial to a...@asterisk.blurb.com
Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War 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] transfer option and pressing #
On Mon, Jul 13, 2009 at 11:50:00AM -0500, Brent Davidson wrote:
Alex Samad wrote:
Hi
I have setup forwarding - xfering - where you press # and then the
extension. I add t to the dial cmd.
My problem is that when you call something like internet banking they
want #, but when # is pressed asterisk gets it instead. is there a way
around this ?
I haven't been able to get asterisk to listen to flash either
Alex
The easiest solution would probably be to look in features.conf and
change the option for forwarding to require two consecutive # presses.
actually when into features and change all the options to *digit
instead of #
The other option would be to put an explicit dial rule for the numbers
that need the # bypass and have them omit T and from the dial command.
You could also set up a dat abase with a simple web front end for your
users to enter numbers that need to have the transfer function bypassed
this is a home system for now (also testbed)
and do something like this (I use AEL so this is in AEL Format)
macro specialDial (ext) {
if (${DB_EXISTS(bypass/${ext})}) {
Dial (${TRUNK}/${ext});// Dial without transfer
} else {
Dial (${TRUNK}/${ext},,T); // Dial With Transfer
}
}
This is assuming you create a table called Bypass in your Asterisk
Database and add the number to the database.
Good luck,
Brent
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[asterisk-users] transfer option and pressing #
Hi I have setup forwarding - xfering - where you press # and then the extension. I add t to the dial cmd. My problem is that when you call something like internet banking they want #, but when # is pressed asterisk gets it instead. is there a way around this ? I haven't been able to get asterisk to listen to flash either Alex -- Why is it there are so many more horses' asses than there are horses? 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] Asterisk module trouble
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote: Hello, i have just installed asterisk 1.6.0.10 on debian 5.0 like: ./configure;make menuselect; make;make install any reason to not use the deb files ? There are no erorrs, but folder /usr/lib/asterisk/modules is empty. What am i doing wrong? Where are modules? p.s. Doing the same on Slackware, i ve got all selected modules at /usr/lib/asterisk/modules. -- But the true strength of America is found in the hearts and souls of people like Travis, people who are willing to love their neighbor, just like they would like to love themselves. - George W. Bush 02/09/2004 Springfield, MO 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] snom mass deploy help
On Mon, Jun 29, 2009 at 10:22:54AM +0200, Loic Didelot wrote: Sorry for dropping in so late, but maybe our solution do configure snom phones can help you We have written a small script that scans the network for snom phones. This is done by doing broadcast pings and using the arp-scan command and reading the arp cache. Then we filter the results on the mac addresses that start with 00:04:13. Of course there are other solutions to get the list of IP addresses of your snom phones. Once we have a list of IP addresses we push our configuration url to the snom phones by calling a url on the phones. http://$IP/dummy.htm?settings=savesetting_server=$massdeploymenturl DHCP was not an option for us because we did not want to change or interfere with the customer existing network structure. I have found this to be the simplest way to do it https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder Best regards, Loïc Didelot. On Thu, 2009-06-18 at 21:25 +1000, Alex Samad wrote: Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex ___ -- 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 -- What I'm suggesting to you is, if you can't name the foreign minister of Mexico, therefore, you know, you're not capable of what you do. But the truth of the matter is you are, whether you can or not. - George W. Bush 11/06/1999 as quoted in the Seattle Post-Intelligencer 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
[asterisk-users] Sangoma A200
Hi I was wondering if any one has used these cards, I am looking at this as a replacement for the tdm410, I have some issues with installing the tdm410 in a small case because of the power plug being at the end of the board. I am in australia seems like we have a different setup for out fxs voltage, any one in oz using this card ? Thanks Alex 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] Sangoma A200
On Sun, Jun 28, 2009 at 10:10:48PM +1000, Rob Hillis wrote: Yes, although not for connecting to the PSTN - I've used one for connecting to a legacy NEC PABX. Thats good Voltage isn't the issue - the difference is in the impedance. Australia I get this in my dmesg when I load up the rdm410 modules [1083334.103487] Freed a Wildcard [1083336.171371] ALAW override parameter detected. Device will be operating in ALAW [1083338.040522] Boosting ringer on slot 1 (89V peak) [1083338.040542] Port 1: Installed -- AUTO FXS/DPO [1083340.340472] Boosting ringer on slot 2 (89V peak) [1083340.340492] Port 2: Installed -- AUTO FXS/DPO uses complex impedance (220+820Ohm resistors with a 120nF capacitor) whereas the US uses a straight resistor. Did yo buy from the us or local ? Alex Alex Samad wrote: Hi I was wondering if any one has used these cards, I am looking at this as a replacement for the tdm410, I have some issues with installing the tdm410 in a small case because of the power plug being at the end of the board. I am in australia seems like we have a different setup for out fxs voltage, any one in oz using this card ? Thanks Alex ___ -- 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 -- Some men are born mediocre, some men achieve mediocrity, and some men have mediocrity thrust upon them. -- Joseph Heller, Catch-22 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] How to force TDM410 to alaw
On Tue, Jun 23, 2009 at 11:32:08AM -0500, Shaun Ruffell wrote: Alex Samad wrote: I am having some problem forcing my tdm410 to alaw over ulaw... You will want to set the alawoverride module parameter to 1. i.e. 'modprobe wctdm24xxp alawoverride=1' or alternatively, edit your /etc/modprobe.d/dahdi file and add a line options wctdm24xxp alawoverride=1 you know I checked modinfo a couple of times and missed it, wondering if this could be added to the opermode=AUSTRALIA to set alaw as default Alex Cheers, -- You're probably wondering why somebody who has been in politics is talking about Social Security. After all, it's been called the third rail of American politics. You grab a hold of it, and you get electrified. - George W. Bush 03/04/2005 South Bend, IN 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
[asterisk-users] 1000Hz kernel
Hi I was reading this article on installing asterisk 1.6 + debian http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian and I noticed they suggested to recompile to 1000Hz enable kernel, I currently have a 250Hz stock standard kernel. I am running on a soekris board - amd geode cpu. Is recompiling the kernel to the 1000Hz going to be beneficial to me, the box is primarily used for firewall router / voip (Asterisk) Alex ___ -- 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] 1000Hz kernel
On Wed, Jun 24, 2009 at 01:02:08AM +0300, Tzafrir Cohen wrote: On Wed, Jun 24, 2009 at 07:10:15AM +1000, Alex Samad wrote: Hi I was reading this article on installing asterisk 1.6 + debian http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian and I noticed they suggested to recompile to 1000Hz enable kernel, I currently have a 250Hz stock standard kernel. I am running on a soekris board - amd geode cpu. Is recompiling the kernel to the 1000Hz going to be beneficial to me, the box is primarily used for firewall router / voip (Asterisk) If you use Asterisk 1.6, just upgrade DAHDI to 2.2 and you won't need a 1000Hz kernel. out of curiosity why is that the case ? Is there a demand for a backport of this to Zaptel? -- The bottom line is simple: If Congress passes a law that does not clarify the rules, if they do not do that, the program is not going forward. - George W. Bush 09/15/2006 Washington, DC speaking with typical clarity at a White House Press Conference 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
[asterisk-users] How to force TDM410 to alaw
Hi I am having some problem forcing my tdm410 to alaw over ulaw, I have 1.6.1.0 asterisk (debian i486) dahdi1:2.2.0 built with the hardware echo canceller firmware /etc/asterisk/chan_dahdi.conf alaw=1-4 but I have this in the general section, before any channel definition dahdi show channel 1 Default law: ulaw even when i have a call going it is still ulaw Thanks Alex 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
[asterisk-users] help setting tone zone
Hi I am seeing this in my syslog [235900.797660] dahdi: Registered tone zone 0 (United States / North America) I am in Australia so I would want to set them to AUS zone I have got this though options wctdm24xxp opermode=AUSTRALIA thanks -- See, we love -- we love freedom. That's what they didn't understand. They hate things - we love things. - George W. Bush 08/29/2002 Oklahoma City, OK 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] help setting tone zone
On Fri, Jun 19, 2009 at 11:08:49AM +0300, Tzafrir Cohen wrote: On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote: Hi I am seeing this in my syslog [235900.797660] dahdi: Registered tone zone 0 (United States / North America) I am in Australia so I would want to set them to AUS zone I have got this though options wctdm24xxp opermode=AUSTRALIA Set loadzone and defaultzone in /etc/dahdi/system.conf . I have and when i trolled through my syslog, I notice sometimes it sets it to 0 and some times not! -- Well, that's going to be up to the pundits and the people to make up their mind. I'll tell you what is a president for him, for example, talking about my record in the state of Texas. I mean, he's willing to say anything in order to convince people that I haven't had a good record in Texas. - George W. Bush 09/20/2000 on MSNBC 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
[asterisk-users] snom mass deploy help
Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex 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] snom mass deploy help
On Thu, Jun 18, 2009 at 07:34:38AM -0400, Alex Balashov wrote: I thought TFTP (and therefore, DHCP option 66) is the only autoprovisioning method Asterisk supports? seems like the documentation from snom for V7, includes the pnp method as well. it sends a subscribe to a multicast address (224.0.1.75) and the listener is meant to respond with a notify which has the url which is normally sent my dhcp, i was hoping to use that this links talks a bit about multicast http://www.voip-info.org/wiki/view/SIP+registrar+server I think I would prefer this method, but I can't find where to set asterisk to listen to the multicast address nor where to program the notify reply Alex 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] help setting up transfering
On Thu, Jun 18, 2009 at 08:06:24AM -0500, Danny Nicholas wrote: Have you tried #1103 or *2103? The # would do a blind transfer, the * would initiate an attended transfer. I tried combination of flash extention flash *67 exten extention *67 exten and re iterated with # instead of *. The doco seemed to suggest after I press flash I should heard a dial tone ! which i don't Alex -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad Sent: Wednesday, June 17, 2009 9:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] help setting up transfering Hi I am trying to get transferring of calls working, I place a call from ext 101 = 103 and then from 101 I try and transfer the call to 102 (such that it will be 102=103), I have tried flash and *2 and nothing seems to work. I have allowed transfers in sip.conf, I am expecting a dial tone when i hit flash 101 - dahdi/1 a uniden pots phone Thanks Alex ___ -- 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 -- I always jest to people, the Oval Office is the kind of place where people stand outside, they're getting ready to come in and tell me what for, and they walk in and get overwhelmed in the atmosphere, and they say, man, you're looking pretty. - George W. Bush 11/04/2004 Washington, DC 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] snom mass deploy help
On Thu, Jun 18, 2009 at 11:57:20PM +0200, Philipp Kempgen wrote: Alex Samad schrieb: seems like the documentation from snom for V7, includes the pnp method as well. it sends a subscribe to a multicast address (224.0.1.75) and the listener is meant to respond with a notify which has the url which is normally sent my dhcp, i was hoping to use that I think I would prefer this method, but I can't find where to set asterisk to listen to the multicast address nor where to program the notify reply I have already told you that Asterisk is not involved in the process of configuring the phone. sorry go the replies after I had sent the email In order to use Snom's PnP configuration method you have to write a daemon which opens a socket on 224.0.1.75 (sip.mcast.net), join the multicast group, read packets and send appropriate ua-profile notification events. This is not what I had hopped for, I had hoped it was something asterisk could handle Have a look at the code I mentioned to get the idea. I will have a look Philipp Kempgen -- I know what I believe. I will continue to articulate what I believe and what I believe I believe what I believe is right. - George W. Bush 07/22/2001 Rome, Italy 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] snom mass deploy help
On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote: On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote: I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options Alex Balashov schrieb: I thought TFTP (and therefore, DHCP option 66) is the only autoprovisioning method Asterisk supports? Asterisk is not involved here at all. Snom supports what they call PnP config. Technically: ---cut--- # SIP Event Notification: # http://tools.ietf.org/html/rfc3265 # SIP UA Profile Event Package: # http://tools.ietf.org/html/draft-ietf-sipping-config-framework-15 # http://tools.ietf.org/html/draft-channabasappa-sipping-app-profile-type-03 # # Snom 3xx: # http://wiki.snom.com/SIP_Traces#PnP_Config # other drafts: # http://tools.ietf.org/html/draft-petrie-sip-config-framework-01 # http://www.cs.columbia.edu/sip/drafts/sip/draft-schulzrinne-sip-config-events-00.txt ---cut--- Gemeinschaft (Asterisk-based open-source PBX) comes with a SIP UA config responder. http://www.amooma.de/gemeinschaft/ https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder I saw links to this on the voip-info site, but my german is non existant. but on first glance through this seems to be what I want. GNU GPL. Philipp Kempgen -- I always jest to people, the Oval Office is the kind of place where people stand outside, they're getting ready to come in and tell me what for, and they walk in and get overwhelmed in the atmosphere, and they say, man, you're looking pretty. - George W. Bush 11/04/2004 Washington, DC 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
[asterisk-users] Analogue card recommendation
Hi I have 2 digium cards (tdm410) with combination of fxs + multiple fxo ports. I have had a quick look at sangoma B series cards. I was wondering if there is a card out there with hardware echo canceller say max 4 ports (mix of fxs/fxo) g729 encoding onboard Alex -- More and more of our imports are coming from overseas. - George W. Bush 09/26/2005 On NPR's Morning Edition 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] Recompiling dahdi-linux after kernel update - To minimize downtime
On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote: After a kernel update (but before rebooting) Is there a way to recompile Zap/Dahdi against the new kernel? My objective is to eliminate the additional downtime that occurs while recompiling/installing zap/dahdi after booting into the new kernel. Please correct me if I'm wrong: My understanding is that until you reboot (after a kernel update), recompiling zap/dahdi still compiles against the OLD kernel, and that's why zap/dahdi doesn't start after rebooting into the new kernel (even if you recompiled it just before rebooting). So my question is: Is there a method to recompile dahdi/zap against the new kernel such that the only downtime is the actual server bounce itself? OR is the current best practice just to simply to reboot, recompile, restart? without trying to start a distro war under debian you can do m-a -t build -l kernel version dahdi aslong as you have the headers installed it will build a module against it. I think if you use the tgz tar ball you can actually specify KDIR to point to the directory with the headers in it. Thanks in advance. -Karl ___ -- 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 -- I also understand how tender the free enterprise system can be. - George W. Bush 07/08/2002 White House press conference, Washington, D.C. 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] Scaling
On Wed, Jun 17, 2009 at 09:34:55AM -0400, Matt Florell wrote: On 6/17/09, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 17 Jun 2009, Steve Totaro wrote: Hi, [snip] Gordon The TC400B is up to 120 channels of G729a now: http://www.digium.com/en/products/voice/tc400b.php wow if you could only get it as a module for a tdm410, that would be cool MATT--- ___ -- 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 -- Ninety percent of everything is crap. -- Theodore Sturgeon 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
[asterisk-users] help setting up transfering
Hi I am trying to get transferring of calls working, I place a call from ext 101 = 103 and then from 101 I try and transfer the call to 102 (such that it will be 102=103), I have tried flash and *2 and nothing seems to work. I have allowed transfers in sip.conf, I am expecting a dial tone when i hit flash 101 - dahdi/1 a uniden pots phone Thanks Alex 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] Help building dahdi for debian
On Sun, Jun 14, 2009 at 03:10:03PM +1000, Alex Samad wrote: On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware okay just for completeness in the thread, I have built the modules and it seems to be working, now to just re config asterisk -- I don't have people coming in the rope line saying, 'I'd like a new bridge, or how about some more highway money.' They're coming to say, 'I'm coming to tell you, Mr. President, I'm praying for you.' - George W. Bush 09/12/2006 Washington, DC said to journalists in the Oval Office (as reported by the National Review) 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] Help building dahdi for debian
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] Although I think I did see it download the firmware The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware Hi I have downloaded http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fw-vpmadt032-1.07.tar.gz and placed it in /usr/share/dadhi (un tarred ) when I reload the wctdm24xxp module I see [ 3350.588717] Found a Wildcard TDM: Wildcard TDM410P (4 modules) [ 3375.529165] Freed a Wildcard [ 3428.201978] Boosting ringer on slot 1 (89V peak) [ 3428.202059] Port 1: Installed -- AUTO FXS/DPO [ 3433.358455] Boosting ringer on slot 2 (89V peak) [ 3433.358474] Port 2: Installed -- AUTO FXS/DPO [ 3433.597478] Port 3: Not installed [ 3434.139532] Port 4: Installed -- AUTO FXO (AUSTRALIA mode) [ 3434.160522] VPM100: Not Present [ 3442.191181] Booting VPMADT032 [ 3446.075609] VPMADT032: Present and operational (Firmware version 117) but when I run dahdi_genconf it use the opensource echo canceller ! also can seem to find cahn_dadhi.so -- He was a state sponsor of terror. In other words, the government had declared, 'you are a state sponsor of terror.' - George W. Bush 01/23/2006 Manhattan, KS On Saddam Hussein 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
[asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
Hi it seems like chan_dahdi.so is missing in debian asterisk 1.4.21 so I have upgraded to 1.6 and no I can load chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref [Jun 16 21:22:30] WARNING[4360]: loader.c:653 load_resource: Module 'chan_dahdi.so' could not be loaded. for a simple change over, its turning out to be not so simple -- A dictatorship would be a heck of a lot easier, there's no question about it. - George W. Bush 07/27/2001 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] missing chan_dahdi.o in debian asterisk 1.4.x
On Tue, Jun 16, 2009 at 07:03:06AM -0500, Kevin P. Fleming wrote: Alex Samad wrote: it seems like chan_dahdi.so is missing in debian asterisk 1.4.21 so I have upgraded to 1.6 and no I can load chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref Moving from Asterisk 1.4 to 1.6 is not a 'simple changeover' :-) It's a major upgrade. In spite of that, for the time being chan_dahdi in Asterisk 1.6 requires that res_smdi be loaded first. If you are manually loading modules (instead of using 'autoload=yes' in /etc/asterisk/modules.conf'), you need to ensure that you load res_smdi before chan_dahdi. There are other modules that have dependencies like this as well, you can see what they are by running 'make menuselect' and looking at the dependency list for each module you want to load. ta, well I have banged my head against the wall for a bit and all seems to be working. for some reason i had no_load res_smdi which caused the problem. some question I have now is when i do a dahdi show channel 1 i get these interesting results Echo Cancellation: 128 taps currently OFF I have a hardware echo can and I have asked for it to be turned on ! Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take my last bug bear (maybe bug) is core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - 56003 12001 1200128001 12001 12000 48002 164010 300018 - 28001 20001 40002 gsm 312020 - 16002 1600232002 16002 16001 52003 168011 304019 - 32002 24002 44003 ulaw 296020 44004 - 116002 2 1 36003 152011 288019 - 16002 8002 28003 alaw 296020 44004 1 -16002 2 1 36003 152011 288019 - 16002 8002 28003 g726aal2 312020 60004 16002 16002- 16002 16001 52003 168011 304019 - 1 24002 44003 adpcm 296020 44004 2 216002 - 1 36003 152011 288019 - 16002 8002 28003 slin 296019 44003 1 116001 1 - 36002 152010 288018 - 16001 8001 28002 lpc10 320022 68006 24004 2400440004 24004 24003 - 176013 312021 - 40004 32004 52005 g729 336021 84005 40003 4000356003 40003 40002 76004 - 328020 - 56003 48003 68004 speex 336022 84006 40004 4000456004 40004 40003 76005 192013 - - 56004 48004 68005 ilbc - - - -- - - - - - - - - - g726 312020 60004 16002 160021 16002 16001 52003 168011 304019 - - 24002 44003 g722 312020 60004 16002 1600232002 16002 16001 52003 168011 304019 - 32002 - 20001 slin16 332021 80005 36003 3600352003 36003 36002 72004 188012 324020 - 52003 20001 - the numbers seem to be way off, it seems to be able to do g729 - ulaw, my vsp only sends g729. but compared to 1.4 the numbers came back as single or maybe 2 digit - the g729 was 3 digits - I am on a soekris board, a amd geode machine. Something is I think going askew Thanks Alex -- I am a person who recognizes the fallacy of humans. - George W. Bush 09/19/2000 Oprah from Bush courts women in cozy 'Oprah' visit by William Goldshclag printed in the New York City edition of the Daily News, Sept. 20, 2000, page 5 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] Help building dahdi for debian
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote: On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote: [snip] dahdi_genconf generates configuration. It is a tool intended to help you and not a required step. It defaults to using mg2[1]. You can tell it to use a different echo canceller in the configuration it generates by setting 'echo_can hpec' in /etc/dahdi/genconf_parameters http://docs.tzafrir.org.il/dahdi-tools/#_sample_genconf_parameters [1] Actually, in the Debian package I patched it to default to use OSLEC, as it's available there and works better than mg2. The patch is so how do I use the hardware echo can and how can I tell it is working ??? trivial: http://patch-tracking.debian.net/patch/series/view/dahdi-tools/1:2.2.0~rc3-1/echocan_oslec also can seem to find cahn_dadhi.so That is part of asterisk . not the 1.4.21 deb - but its in 1.6.1 -- That's just the nature of democracy. Sometimes pure politics enters into the rhetoric. - George W. Bush 08/08/2003 Crawford, TX 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] Help building dahdi for debian
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote: Tzafrir Cohen wrote: Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) It's automatic; nothing needs to be specified in /etc/dahdi/system.conf at all. If chan_dahdi is configured to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. got told by support to check cat /sys/module/wctdm24xxp/parameters/vpmsupport the thing that is interesting is that dahdi_cfg -vv shows me mg2 (I have mg2 in the /etc/dahdi/system.cfg). Should I just leave echocanceller out fo system.conf ? and dadhi show channel 1 still shows echo cancellation off ? alex -- After all, Europe is America's closest ally. - George W. Bush 02/23/2005 Mainz, Germany 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] missing chan_dahdi.o in debian asterisk 1.4.x
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote: Alex Samad wrote: some question I have now is when i do a dahdi show channel 1 i get these interesting results Echo Cancellation: 128 taps currently OFF I have a hardware echo can and I have asked for it to be turned on ! that makes sense, I think the confusion here is that the documentation has lots of stuff on the software ec, but not much on the hardware ones for example to use the hardware echo can do I leave echocanceller blank in /etc/dahdi/system.conf ? Why doesn't dahd_cfg -vv show up the hardware ec This is nothing new; the echo canceller on a channel is not enabled unless the channel is in an active call and the call hasn't caused it to be disabled (via tones). If you are looking at a channel that is inactive, this is exactly what you will see. Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take That is not valid syntax for /etc/dahdi/system.conf, correct syntax would be 'alaw=1-4'. typo in the email, I actually had alaw=1-4 my last bug bear (maybe bug) is core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) Note the phrase (in microseconds) here. This behavior has changed from Asterisk 1.4, and is documented in the documentation that came with Asterisk 1.6. If you haven't read the CHANGES and UPGRADE files thoroughly, you are spending time trying to understand things (assuming they are problems) that you don't need to spend. very true Something is I think going askew It sounds like you are trying to do a major upgrade without actually taking the time to learn what has changed and what that will require you to do. That's really very important, which is why we spend time writing that documentation in the first place :-) yes that is true, but its a home system, I will do work after falling over the pitfalls at home. -- Just when you think Life's a Bitch, it has puppies. 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] missing chan_dahdi.o in debian asterisk 1.4.x
On Wed, Jun 17, 2009 at 07:16:53AM +1000, Alex Samad wrote: On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote: Alex Samad wrote: [snip] Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take That is not valid syntax for /etc/dahdi/system.conf, correct syntax would be 'alaw=1-4'. typo in the email, I actually had alaw=1-4 just to follow up I have the above set placed a call to voicemail and did a dahdi show channel 1 the relevant bits Default law: ulaw Echo Cancellation: 256 taps currently ON Alex 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] Help building dahdi for debian
On Wed, Jun 17, 2009 at 01:23:19AM +0300, Tzafrir Cohen wrote: On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote: On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote: Tzafrir Cohen wrote: Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) It's automatic; nothing needs to be specified in /etc/dahdi/system.conf at all. If chan_dahdi is configured to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. got told by support to check cat /sys/module/wctdm24xxp/parameters/vpmsupport the thing that is interesting is that dahdi_cfg -vv shows me mg2 (I have mg2 in the /etc/dahdi/system.cfg). Should I just leave echocanceller out fo system.conf ? That's the software EC. It doesn't matter. ok and dadhi show channel 1 still shows echo cancellation off ? What is the exact value there? Is it at the time of an active call? kevin advised on this, it shows the status at the time of a call. I tried whilst making a call and it was on -- This case has had full analyzation and has been looked at a lot. I understand the emotionality of death penalty cases. - George W. Bush 06/23/2000 Seattle Post-Intelligencer 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
[asterisk-users] setting codecs on the fly
Hi I would like the option to set the codec used on a call by call basis. I have a tdm410 2fxs + 1fxo. when I make calls to my vsp, they go through as ulaw, I am guessing because I have allowed if for the vsp (g729, alaw and ulaw). I would prefer to use g729 from the fxs to the vsp but I would like the option to make alaw calls on demand and maybe ulaw. Or even to set the default on the digium card to alaw - that seems to be the default in OZ Thanks 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
[asterisk-users] tdm loosing interrupts and latency
Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here (http://forums.digium.com/viewtopic.php?p=126997highlight=sid=9de59f41f1a93ee8701b28fdd0cf6073) Seems like the driver (and this is in zaptel dadhi code), increases latency by +1 until 30. and then the card seems to not work. In my case I have seen latency increase from 8m (I have this as a starting point in the module load) up to 17ms usually around here the fxs and fxo ports stop working . I have to unload and then reload the module. bummer. I can think of a couple of solutions 1) build some intelligence to bring down the number when things are okay 2) build logic to say if a number is provided on module load to fix it to that 3) add a sysfs (/proc) interface to allow changing this value on the fly I could also try and solve my problem with the dead loop detection cat /proc/interrupts CPU0 0: 23809265XT-PIC-XTtimer 1: 0XT-PIC-XTi8042 2: 0XT-PIC-XTcascade 4:255XT-PIC-XTserial 5: 459544XT-PIC-XTeth1 8: 0XT-PIC-XTrtc0 10: 95177163XT-PIC-XTwctdm24xxp0 11: 28938443XT-PIC-XTeth0 12: 28938632XT-PIC-XTeth3 14:3624228XT-PIC-XTide0 15: 1XT-PIC-XTehci_hcd:usb1, ohci_hcd:usb2 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts TRM: 0 Thermal event interrupts SPU: 0 Spurious interrupts ERR: 0 MIS: 0 as you can see with the interrupts the wctdm24xxp0 is above eth0 (local lan) and eth3 (my adsl) eth1 is wireless and not heavily used So any one had this problems, any other possible solution to this ? How to engage digium to providing a fix for this ? Alex 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] tdm loosing interrupts and latency
On Mon, Jun 15, 2009 at 08:19:33PM -0500, Lyle Giese wrote: Alex Samad wrote: Hi [snip] as you can see with the interrupts the wctdm24xxp0 is above eth0 (local lan) and eth3 (my adsl) eth1 is wireless and not heavily used So any one had this problems, any other possible solution to this ? How to engage digium to providing a fix for this ? Alex If your ppp is dropping, that means you have lost Internet connectivity, correct? If that is the case, then that is your problem as Asterisk does not tolerate the lose of DNS resolution very well. I think you have missed the point of the question. But I use and internal dns server, I understand if I lose my adsl my voip calls will be lost, but I also route some calls out pstn, they should stay. But the problem is with the digium driver not with asterisk (which make s me think this might not be the right mailing list !) Lyle Giese LCR Computer Services, Inc. -- The suicide bombings have increased. There's too many of them. - George W. Bush 08/15/2001 Albuquerque, NM 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] Help building dahdi for debian
On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote: To get this to work can i simply apt-get source dahdi-linux modify debian/patches/series to comment out no_firmware_download then dpkg-buildpackage -rfakeroot -us -uc -b should that work ? Yes gave it a try, didn't seem to down load any thing, going to go back to zaptel to get it installed and working and I will investigate some more with the dadhi later Alex -- I think it's very important for world leaders to understand that when a new administration comes in, the new administration will be running the foreign policy. - George W. Bush 01/12/2001 interview with USA Today 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] Help building dahdi for debian
On Sat, Jun 13, 2009 at 01:10:33PM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 07:51:54PM +1000, Alex Samad wrote: On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote: To get this to work can i simply apt-get source dahdi-linux modify debian/patches/series to comment out no_firmware_download then dpkg-buildpackage -rfakeroot -us -uc -b should that work ? Yes gave it a try, didn't seem to down load any thing, Could you please be more specific? sorry, got to remember people can't read my mind I have download the source, made the change to series files and then did the dpkg-buildpackage -us -uc -b, whilst watching the process it did not seem to download any files. I compared (not closely - ie line by line) the similar process by download the tgz from digim and doing a make then a make -n install with the later i saw wgets in the former nothing, when i checked the .deb files for firmware files they weren't there! alex Have you built a modules package from the generated dahdi-source package (using m-a)? -- I'm not going to talk about what I did as a child. What I am going to talk about -- and I am going to say this consistently -- [is that] it is irrelevant what I did 20 to 30 years ago. What's relevant is that I have learned from any mistakes I made. I do not want to send signals to anybody that what Gov. Bush did 30 years ago is cool to try. - George W. Bush in an interview with WMUR-TV in New Hampshire, when asked if he had used drugs, marijuana, cocaine 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] Help building dahdi for debian
On Sat, Jun 13, 2009 at 05:10:34PM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 09:46:23PM +1000, Alex Samad wrote: On Sat, Jun 13, 2009 at 01:10:33PM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 07:51:54PM +1000, Alex Samad wrote: On Sat, Jun 13, 2009 at 11:58:40AM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 09:51:24AM +1000, Alex Samad wrote: To get this to work can i simply apt-get source dahdi-linux modify debian/patches/series to comment out no_firmware_download then dpkg-buildpackage -rfakeroot -us -uc -b should that work ? Yes gave it a try, didn't seem to down load any thing, Could you please be more specific? sorry, got to remember people can't read my mind I have download the source, made the change to series files and then did the dpkg-buildpackage -us -uc -b, whilst watching the process it did not seem to download any files. It merely packages (most of the) the source tarball in the dahdi-source binary package, which is later built with dahdi-linux. that makes sense, I had a time constraint, will look at it next weekend, when I have some times. because I want to move to 2.6.29 (You can technically extract the modules tarball from dahdi-source and use m-a directly with it, but it's a bit complicated) -- Professor: Okay, Amoeba Boys. Thank you for your cooperation. You're free to go. Bossman: Go?! I thought we were criminals! You locked us up and everything! Professor: Well, everything's all right now. Out you go. 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] Help building dahdi for debian
On Sat, Jun 13, 2009 at 11:04:05PM +0200, Erik de Wild: Tripple-o wrote: perhap this is not an answer to your question but following this procedure will result in a working Asterisk system. I use this list when setting up an Asterisk system after installing a basic Debian etch network install : [snip] Usually I just modprobe dahdi_dummy so MeetMe() works fine. Why making a problem of not having the debian packages available if building a working system from source is so simple? Hope this is useful for someone Thanks, but I try and do things the debian way - use deb's not tar balls easier to maintain. My only issue has been that because of debian rules the firmware for the hw echo cancellor isn't provided Alex \erik Date: Sat, 13 Jun 2009 09:51:24 +1000 From: Alex Samad a...@samad.com.au Subject: Re: [asterisk-users] Help building dahdi for debian To: asterisk-users@lists.digium.com Message-ID: 20090612235124.gb17...@samad.com.au Content-Type: text/plain; charset=us-ascii ___ -- 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 -- But minister, it isn't like this film is the first troublesome thing to come out of Canada. Let us not forget Bryan Adams. No, no. The Canadian government has apologized for Bryan Adams on several occasions. 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] Help building dahdi for debian
On Sun, Jun 14, 2009 at 08:20:18AM +1000, Alex Samad wrote: [snip] It merely packages (most of the) the source tarball in the dahdi-source binary package, which is later built with dahdi-linux. that makes sense, I had a time constraint, will look at it next weekend, when I have some times. because I want to move to 2.6.29 okay started up my vm machine to build this, apt-source fix series dpkg-buildpackage -us -uc -b install the source deb m-a build dahdi dpkg -c dahdi-modules-2.6.29-2-486_2.2.0~dfsg~rc5-1.1-alex01+2.6.29-5_i386.deb and I still can't see any firmware !! only kernel modules. Although I think I did see it download the firmware (You can technically extract the modules tarball from dahdi-source and use m-a directly with it, but it's a bit complicated) -- we: The single most important word in the world. 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] Help building dahdi for debian
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: On Sun, Jun 14, 2009 at 12:23:41PM +1000, Alex Samad wrote: On Sun, Jun 14, 2009 at 08:20:18AM +1000, Alex Samad wrote: [snip] It merely packages (most of the) the source tarball in the dahdi-source binary package, which is later built with dahdi-linux. that makes sense, I had a time constraint, will look at it next weekend, when I have some times. because I want to move to 2.6.29 okay started up my vm machine to build this, apt-source fix series dpkg-buildpackage -us -uc -b install the source deb m-a build dahdi dpkg -c dahdi-modules-2.6.29-2-486_2.2.0~dfsg~rc5-1.1-alex01+2.6.29-5_i386.deb and I still can't see any firmware !! only kernel modules. Although I think I did see it download the firmware I think I solved my problem. the kernel modules downloads and compiles int the vpmadt032_x86_32.o_shipped blob. I was expecting and looking for the firmware! silly me The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware okay -- When I was a kid I remember that they used to put out there in the Old West a wanted poster. It said, Wanted: Dead or Alive. - George W. Bush 09/18/2001 Washington, DC 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
[asterisk-users] Help building dahdi for debian
Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi what is the best way forward to recompile with hardware echo canceller support. Thanks Alex 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] Help building dahdi for debian
On Fri, Jun 12, 2009 at 05:40:16PM +0300, Tzafrir Cohen wrote: On Fri, Jun 12, 2009 at 11:58:51PM +1000, Alex Samad wrote: Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi What do you mean? I suspect that the following patch: http://patch-tracking.debian.net/patch/series/view/dahdi-linux/1:2.2.0~dfsg~rc5-1/no_firmware_download yeah I downloaded the source and found this patch, BTW: legal purity aside, downloading an external source at build time is generally a big no-no for a build server on Debian. Thus this downloading breaks my intention to get the modules distributed in the distribution as part of linux-modules-2.6 package (which includes all sorts of external modules). I understand, but it was feasible in zaptel to build a package were all you had to do was download the firmware - I haven't looked at the dep / source requirements so I am not sure if the is feasible on the debian build servers If you're not using those packages (and build at a place with internet connectivity) you should have no problem. Which is the case for you? building my own package from your source just removing the above offending patch :) any chance of getting digium to host a digium debian repo (sort of how virtulbox doit), that way they could have a fully build package ? alex -- Neither in French nor in English nor in Mexican. - George W. Bush 04/21/2001 declining to take reporters' questions during a photo op with Canadian Prime Minister Jean Chretien 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] Help building dahdi for debian
On Sat, Jun 13, 2009 at 01:40:48AM +0300, Tzafrir Cohen wrote: On Sat, Jun 13, 2009 at 06:57:11AM +1000, Alex Samad wrote: any chance of getting digium to host a digium debian repo (sort of how virtulbox doit), that way they could have a fully build package ? Or resolve the issues that made this patch necessary in the first place. To get this to work can i simply apt-get source dahdi-linux modify debian/patches/series to comment out no_firmware_download then dpkg-buildpackage -rfakeroot -us -uc -b should that work ? -- I was not prepared to shoot my eardrum out with a shotgun in order to get a deferment. Nor was I willing to go to Canada. So I chose to better myself by learning how to fly airplanes. - George W. Bush 02/25/1990 Dallas Morning News 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] Query about tdm410 cards
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote: On Wed, 10 Jun 2009, Alex Samad wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I know this mighs sound odd, but do you really need the +12V connection? You only need it if you have analogue phones plugged in and not exchange lines.. I have 2 fxs + 1fxo so I know - this is obvious and you probably do have analogue phones plugged in, but I'm just checking!!! we all miss the obvious at some time Gordon ___ -- 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 -- If we are going to save a generation of young people, our children must know they will face bad consequences for criminal behavior. Sadly, too many youths are not getting that message. Our juvenile justice system must say to our children: We love you, but we are going to hold you accountable for your actions. - George W. Bush 01/01/2000 2000 Bush campaign literature 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] Query about tdm410 cards
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote: At 02:01 PM 6/10/2009, you wrote: http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me I'd buy a standard one, pull the pins, cut off the wire end of the plug, put it back in bend the pins over and insulate it with a bit of hot melt or heatshrink. Probably as good as anything you'll buy. I have soldered to the back of the board, the molex pins go all the way through the pcb Ira ___ -- 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 -- I'm not available for comment.. 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
[asterisk-users] Query about tdm410 cards
Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the pwr2400b - ie how it connects to the cards. The web site is a bit devoid of the information and all the photo's are not clear. this would make my life rather simple, I have 12V + GND to supply the card - seems like people have done this with a TDM400, unfortunately the 410 is longer Alex 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] Query about tdm410 cards
On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote: On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. You can probably find the extension cable or connector you need here http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me thanks -- Why is it that all of the instruments seeking intelligent life in the universe are pointed away from Earth? 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] Query about tdm410 cards
On Wed, Jun 10, 2009 at 05:49:22PM -0500, Kevin P. Fleming wrote: Alex Samad wrote: I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the pwr2400b - ie how it connects to the cards. The web site is a bit devoid of the information and all the photo's are not clear. No, the PWR2400B includes a PCI bracket with cables that connect to the Molex connectors on the cards. oh well I have soldering iron, seems like a few people have soldered to the connectors underneath Alex -- My plan reduces the national debt, and fast. So fast, in fact, that economists worry that we're going to run out of debt to retire. - George W. Bush 02/24/2001 radio address 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
[asterisk-users] Help with inbound dialplan
Hi I am trying to setup asterisk at home, I have 1 in bound VSP (I have a register cmd setup for that in asterisk). At home I have a cordless phone with 2 line capability - I currently have 2 spa3102's in place to handle the 2 lines ( I am in the process of buying tdm410 to handle to handle this and the backup pstn line). I also have 2 laptops setup with soft sip phones. What I would like to see happen is when an inbound call comes, I would like to ring 1 line on the cordless phones - because it sounds really weird when both the lines ring at the same time. and I would like to see both the laptops ring if they are connected. this is what I am looking at using exten = s,1,Dial(SIP/SPA3102bSIP/laptopSIP/tlaptop,20,j) exten = s,n,VoiceMail(v...@spa3102,u) exten = s,n,Hangup exten = s,102,Dial(SIP/SPA3102aSIP/laptopSIP/tlaptop,20,j) exten = s,n,VoiceMail(v...@spa3102,u) exten = s,n,Hangup exten = s,203,VoiceMail(v...@spa3102,b) SPA3102b is line 2, SPA3102b is line 1 on the cordless and laptop tlaptop are the laptop SIP definitions. one problem that happens is that when somebody is making a call on line2 (SPA3102b) outbound, asterisk will still send an inbound call to the SPA3102b because of call waiting. Is there some way of avoiding this - seems silly to me Is this the way to do ? or is there some way to create a gobalvar which is a dynamic dial string, but I can't figure out how to modify a gobalvar on registration ? Thanks Alex 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] IP phone recommendation
On Wed, Jun 03, 2009 at 08:23:13PM +1000, Rob Hillis wrote: Christian Stredicke wrote: Check out the snom 300 or the snom 820... Good lord... talk about two extremes... :) The Snom 300 is pretty good, but the 320 is much better and costs around a *third* of what the Snom 820 does. I have been looking at a snom 300, which seems okay. the display goes a bit haywire occasionally - not sure why yet. Are the 320 worth the extra money ? Alex Stick with the older model snoms. So far I've seen nothing about the 820 to justify the significant extra expense. ___ -- 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 -- We ended the rule of one of history's worst tyrants, and in so doing, we not only freed the American people, we made our own people more secure. - George W. Bush 05/03/2003 Crawford, TX 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
[asterisk-users] zaptel to dahdi
Hi i have just recently installed asterisk 1.4 server with a digium card 410, i used the zaptel packages in debian. now I have notice the move to dahdi which seems to be a rename and some changes as well. is it a easy change from zaptel to dahdi ? any sort of gotchas to watch out for ? Alex 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
[asterisk-users] question about reinvite
Hi My setup is Internet - firewall - asteriskbox - spa3102a - spa3102b the spa's can talk to the firewall directly. The firewall does NAT. The current asterisk flow for outgoing calls is phone = spa3102 = asterisk = vsp and vis versa for inbound calls. can I use re invite for outbound calls such that the spa3102 ends up talking directly to the vsp, i believe the firewall will handle it properly. But I am guessing i can reinvite on the inbound because all my 5060 traffic goes back to the asterisk server. So the question is on the sip.conf [context for spa3102a], can I put reinvite here and if not where can i put it. Thanks Alex 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] stucked calls in asterisk 1.4
On Thu, May 28, 2009 at 10:49:38AM +0200, Stefan Schmidt wrote: David Backeberg schrieb: On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote: all server are in one rack in our datacenter and are connected to an HP Procurve 2650 switch, which has been setup around 3 months ago, cause of the old switch died silent in the night. all server had two interfaces and i have allready tried to route the traffic between the pbx and the routing server over the second interface, where database requests normally run. But this didnt solved the problem too. i will try to increase the UDP buffer size in the linux kernel, maybe this will take some affect. I will say that asterisk-1.6 is supposed to have a better SIP stack than 1.4. Perhaps the difference in performance will help you. Specifically, check out: http://svn.digium.com/svn/asterisk/branches/1.6.0/CHANGES http://svn.digium.com/svn/asterisk/branches/1.6.1/CHANGES I'd recommend you go to at least 1.6.1.* series i will have a look at 1.6.1 version but the problem is a network/kernel problem. i´ve tried to set the udp rcv buffer to 1mb but that didnt change the problem. Hi I am new to asterisk so my suggestions might be a bit silly. Why not setup a iax2 connection bettween the asterisk servers, because its a lower overhear and more efficient. what i can see is that the asterisk itself sometimes lag so no call can be started, but running calls wont be broken. in netstat -su i see a big amount of udp packet receive errors when it lags. Like 600 to 800 pakets per second. if you are seeing packet loss, have you check the stats on the cards and the ports on the switch any sort of loss might indicate a faulty card/port you could also try some network testing with iperf by default it tests udp. the other thought is keep going the way youwere add another nic card to server a such that you have eth0 for your sip clients eth1 for router server b eth2 for router server c it would just take a bit of routing magic. The thing to watch then is the bus these card are on if they share the same pci bus. But you are not really moving around much data. Try linux-kernel mailing list or do a google for udp buffer tunning from memory there is the kernel option which is the abs max, but the program must also select a buffer amount which is usually the default value which is not the max value. another thing to look for (you haven't mentioned you nic type or kernel version), but I had a realtek 8168b which would sort of work with the in kernel driver (8111c) on any kernel 2.6.28. After 2.6.28 the in kernel driver worked really well. system would be under load packets would be come corrupt for example ethtool -S eth0 NIC statistics: tx_packets: 75371387 rx_packets: 970026891 tx_errors: 0 rx_errors: 0 rx_missed: 0 align_errors: 0 tx_single_collisions: 0 tx_multi_collisions: 0 unicast: 969467810 broadcast: 201337 multicast: 357744 tx_aborted: 0 tx_underrun: 0 in syslog i cant see anything that would cause asterisk to react like this. anyone an idea what can cause this problem? best regards steve ___ -- 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 -- Documentation: Instructions translated from Swedish by Japanese for English speaking persons. 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] stucked calls in asterisk 1.4
On Thu, May 28, 2009 at 02:15:08PM +0200, Stefan Schmidt wrote: Alex Samad schrieb: Hi Hi Alex, I am new to asterisk so my suggestions might be a bit silly. Why not setup a iax2 connection bettween the asterisk servers, because its a lower overhear and more efficient. We had changed from iax connections to sip connections cause we had timing problems with iax. Also in meetme and when a client send a fax over the connection. This problems has gone after changing to sip connects. okay [snip] its not a network packet loss its in the kernel or asterisk itself. okay [snip] the other thought is keep going the way youwere add another nic card to server a such that you have eth0 for your sip clients eth1 for router server b eth2 for router server c i´ve allready tried this, and the problem still occurs. i thought you had only add 1 extra nic. The thought was that 2 servers could push 1 nic over the limit, where as a 1 nic per server possibly not. But if you are sure its not a networking packet loss issue then its mute [snip] Try linux-kernel mailing list or do a google for udp buffer tunning from memory there is the kernel option which is the abs max, but the program must also select a buffer amount which is usually the default value which is not the max value. increase the max udp buffer size was just something i´ve found to try but i´ve set the default value to 512kB by now, maybe this will help. At work there was a thread similiar to this, but different application problem with udp. Some who new said (if I can remember it rightly). there is a max/default/min for the system and an application can ask for a bigger buff, but by default it gets default, so changing the max isn't going to help by itself. another thing to look for (you haven't mentioned you nic type or kernel version), but I had a realtek 8168b which would sort of work with the in kernel driver (8111c) on any kernel 2.6.28. After 2.6.28 the in kernel driver worked really well. system would be under load packets would be come corrupt system info: HP Proliant DL380 G5 2x Intel Xeon 2,3 GHZ quad core 6GB Ram Debian Lenny ver. 5.0.1 kernel: 2.6.26-1-amd64 I´ve recompiled with dont optimize and debug threads and when the asterisk hangs or lag i see in core show lock the following: [snip] maybe this is a bug? I am not a asterisk person so can't comment on that. But another thought came to mind, maybe run 2 instances of asterisk - it seems like you have plenty of horse power here it looks like it should be able to cope (I am presume no translating ?) The other thing is HP have a PSP pack for there servers - they have just started to release debian based ones as well - current alien'ed rpm packages but its a start :) alex best regards steve -- There is an old custom among my people. When a woman saves a man's life, he is grateful. -- Nona, the Kanuto witch woman, A Private Little War, stardate 4211.8. 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] visp multiaccount + firewall configuration problem
On Mon, May 25, 2009 at 09:29:54AM +1000, Paul Hales wrote: Alex Samad wrote: Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it did not seem to have any problem. now I have placed another box to act as a firewall in front of the asterisk box and I can't seem to register both lines. the sip account details are the same except for the username + id. so same destination ip. I would guess what I would really like to do is set a bindport for a particular account. port = 5061 I believe that sets the remote port no the asterisk port and the visp only listens on 5060. PaulH ___ -- 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 -- I was raised in the West. The West of Texas. It's pretty close to California. In more ways than Washington, D.C., is close to California. - George W. Bush 04/08/2000 Los Angeles, CA in Los Angeles as quoted by the Los Angeles Times 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
[asterisk-users] visp multiaccount + firewall configuration problem
Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it did not seem to have any problem. now I have placed another box to act as a firewall in front of the asterisk box and I can't seem to register both lines. the sip account details are the same except for the username + id. so same destination ip. I would guess what I would really like to do is set a bindport for a particular account. The other solution I can think of is setting up a asterisk server on the firewall, but I am unsure what that worked and it didn't work through the nat firewall. So I have come to mailing list to see if there are any other solutions Alex 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] Dialplan Priorities and Sort Order...
On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote:
Alex Samad wrote:
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
[snip]
I left the busy after dial because this is what the original poster
had. In this case, if the channel does not get hungup then the next
execution will be a busy, letting the caller know the call was not
completed. In the dialplan macro I normally use it checks for the call
status and acts accordingly as seen below. If you are new to Asterisk
syntax this is probably confusing. If you are following it, I don't
exit on a BUSY because I frequently get a BUSY when there is actually a
congestion or channel problem. Anyhow, how often is a line actually
busy these days?
exten = s,n,Set(DIALS1=IAX2/x...@carrier1-out/${ARG1},90,T)
exten = s,n,Set(DIALS2=IAX2/x...@carrier2-out/${ARG1},90,T)
exten = s,n,Set(DIALS3=SIP/${ar...@carrier3-out,90,T)
exten = s,n,Set(DialNum=3)
exten = s,n,Set(DialCount=0)
exten = s,n(dial),Set(DialCount=$[1 + ${DialCount}])
exten = s,n,GotoIf($[${DialCount} ${DialNum}]?h,1)
exten = s,n,Dial(${DIALS${DialCount}})
exten = s,n,Goto(dial)
exten = s-CONGESTION,1,Congestion(5)
exten = s-CONGESTION,n,Macro(rhangup)
exten = s-BUSY,1,Playtones(busy)
exten = s-BUSY,n,Busy(5)
exten = s-BUSY,n,Macro(rhangup)
exten = h,1,GotoIf($[${DIALSTATUS} = BUSY]?s-BUSY,1)
exten = h,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?s-CONGESTION,1)
exten = h,n,GotoIf($[${DIALSTATUS} = CONGESTION]?s-CONGESTION,1)
exten = h,n,Macro(rhangup)
exten = t,1,Macro(rhangup)
Wow thats a neet way to dial multiple providers, can you make it into a
macro and passin an array of numbers ? and maybe another param to
specify how many elements in the array ?
[snip]
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Re: [asterisk-users] Open source SIP client
On Tue, May 19, 2009 at 10:38:24AM +1000, Paul Hales wrote: Not true. I am always wrong. (wait...is that a paradox?) only on the 42nd time PaulH [snip] ContactTel Business wrote: signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Priorities and Sort Order...
On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
What you have here should work just fine except:
exten = _1866NXX,1,Dial(ZAP/g1/${EXTEN}) -- note the change from n to 1.
I also don't understand why you have an Answer after your Dial statements.
I would do this:
Hi
I am new to asterisk and still trying to get head around dialplan. can
I clarify what is happening in this plan
in context general-outbound, you are including pri_outbound (all its
rules are placed at the bottom because its an include this its rules are
sorted bellow the rules in the context, similarly the order of include
is important)
Why do you use busy after the dial ?
; Outbound via POTS
[general-outbound]
include = pri_outbound
exten = _1800NXX,1,Dial(ZAP/g1/${EXTEN})
exten = _1800NXX,n,Busy
exten = _1800NXX,n,Hangup
exten = _1866NXX,1,Dial(ZAP/g1/${EXTEN})
exten = _1866NXX,n,Busy
exten = _1866NXX,n,Hangup
exten = h,1,Hangup
; Outbound via PRI
[pri_outbound]
exten = _X.,1,Dial(ZAP/g0/${EXTEN})
exten = _X.,n,Busy
exten = _X.,n,Hangup
exten = h,1,Hangup
Tim Nelson wrote:
[snip]
--
The best you get is an even break.
-- Franklin Adams
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Re: [asterisk-users] how to ignore a ring on a line
On Fri, May 15, 2009 at 12:12:23PM +0300, Tzafrir Cohen wrote: On Fri, May 15, 2009 at 02:47:30PM +1000, Alex Samad wrote: Hi I have a fxs (tdm410 ) connected to a pstn that is primarily used for faxing, it is meant to be a just in case line. How do I tell asterisk to ignore the line completely - ie don;t pick up when it rings ? Simply don't Answer() in the dialplan . seems logical, I have this exten = s,1,noop exten = s,n,wait(60) before I didn't have anything, when I am at a console I get messages of asterisk trying to pick up the call should I just have an empty context ? -- One of the major difficulties Trillian experienced in her relationship with Zaphod was learning to distinguish between him pretending to be stupid just to get people off their guard, pretending to be stupid because he couldn't be bothered to think and wanted someone else to do it for him, pretending to be so outrageously stupid to hide the fact that he actually didn't understand hat was going on, and really being genuinely stupid. He was reknowned for being quite clever and quite clearly was so -- but not all the time, which obviously worried him, hence the act. He preferred people to be puzzled rather than contemptuous. This above all appeared to Trillian to be genuinely stupid, but she could no longer be bothered to argue about. -- Douglas Adams, _The Hitchhiker's Guide to the Galaxy_ 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] Problem with Asterisk + TDM410 FXO
On Thu, May 14, 2009 at 07:46:26AM +0200, Marco Sambo wrote: FXO channels shuld have FXS signalling, and FXS channels shuld have FXO signalling, so: # FXO channels are 1,2,3 fxsks=1,2,3 # FXS channel is 4 fxoks=4 yep turned it around and tested it out, worked, had to fxs tune to get the fxs channel working. sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is that a attached fxs presents internally as a fxo I have a pstn line attached to the FXO and I have my pabx attached to 2 FXS ports, which signal as fxo into asterisk (I could be wrong about that). # cat /etc/zaptel.conf fxsks=4 fxoks=1,2,3 ___ -- 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 -- The misnaming of fields of study is so common as to lead to what might be general systems laws. For example, Frank Harary once suggested the law that any field that had the word science in its name was guaranteed thereby not to be a science. He would cite as examples Military Science, Library Science, Political Science, Homemaking Science, Social Science, and Computer Science. Discuss the generality of this law, and possible reasons for its predictive power. -- Gerald Weinberg, An Introduction to General Systems Thinking 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
[asterisk-users] how to ignore a ring on a line
Hi I have a fxs (tdm410 ) connected to a pstn that is primarily used for faxing, it is meant to be a just in case line. How do I tell asterisk to ignore the line completely - ie don;t pick up when it rings ? Alex -- I will have a foreign-handed foreign policy. - George W. Bush 09/27/2000 Redwood, CA 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
[asterisk-users] Problem with Asterisk + TDM410 FXO
Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to a fax line and used just in case the internet connection is down. I have tested the pstn line connection with a soft phone and it seems to be working fine. I need some help on how to tell asterisk to ignore the line for incoming ! when I connect the PABX to the FXO ports I ran into a problem. It seems to register okay, I pick up the handset on the pabx and select line 1 and i can hear a dial tone (same with line2) - this is the same what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in use. But I can't hear anything from the pabx - no dtmf tones and thus can't dial! when I try dialing in from the internet to asterisk then to ZAP/g1 the pabx can see the ring and I can pick up the phone I can hear the other end, but they can't hear me. I don't believe its a firewall issue as I can't dial from the pabx okay some print outs # zaptel_hardware pci::05:02.0 wctdm24xxp+ d161:8005 Wildcard TDM410P # ztcfg -vv Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. # cat /etc/zaptel.conf fxsks=4 fxoks=1,2,3 loadzone=au defaultzone=au /etc/asterisk/zapata.conf # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$' [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 Group=1 signalling=fxo_ks context=in-pbx channel=1-2 Group=2 echocancel=yes signalling=fxs_ks context=in-pstn channel=4 Group=3 signalling=fxo_ks context=in-spare channel=3 the thing that has me beet is that it work with the spa9000 I would expect it to just sort of work with the digium card. the os is debian amd64 2.6.26 #dpkg -l asteri* | grep ^ii ii asterisk1:1.4.21.2~dfsg-3 Open Source Private Branch Exchange (PBX) ii asterisk-barbarast.com 0.0.0-1 asterisk setup for hme1.samad.com.au ii asterisk-doc1:1.4.21.2~dfsg-3 Source code documentation for Asterisk ii asterisk-sounds-extra 1.4.7-1 Additional sound files for the Asterisk PBX ii asterisk-sounds-main1:1.4.21.2~dfsg-3 Core Sound files for Asterisk (English) #dpkg -l zapt* | grep ^ii ii zaptel 1:1.4.11~dfsg-3 zapata telephony utilities ii zaptel-modules-2.6.22-2-amd64 1:1.4.11~dfsg-3+2.6.22-4 zaptel modules for Linux (kernel 2.6.22-2-am ii zaptel-modules-2.6.26-2-amd64 1:1.4.11~dfsg-3+2.6.26-15 zaptel modules for Linux (kernel 2.6.26-2-am ii zaptel-source thanks Alex 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] Problem with Asterisk + TDM410 FXO
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote: I think you have your line types mixed up - FXS is for phones, FXO is for lines. sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is that a attached fxs presents internally as a fxo I have a pstn line attached to the FXO and I have my pabx attached to 2 FXS ports, which signal as fxo into asterisk (I could be wrong about that). An analogue passthorugh setup _is_ doable, just not overly recommended. PaulH Alex Samad wrote: Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to a fax line and used just in case the internet connection is down. I have tested the pstn line connection with a soft phone and it seems to be working fine. I need some help on how to tell asterisk to ignore the line for incoming ! when I connect the PABX to the FXO ports I ran into a problem. It seems to register okay, I pick up the handset on the pabx and select line 1 and i can hear a dial tone (same with line2) - this is the same what I get on the spa9000. Asterisk tells me ZAP/1-1 and ZAP/2-1 are in use. But I can't hear anything from the pabx - no dtmf tones and thus can't dial! when I try dialing in from the internet to asterisk then to ZAP/g1 the pabx can see the ring and I can pick up the phone I can hear the other end, but they can't hear me. I don't believe its a firewall issue as I can't dial from the pabx okay some print outs # zaptel_hardware pci::05:02.0 wctdm24xxp+ d161:8005 Wildcard TDM410P # ztcfg -vv Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels to configure. # cat /etc/zaptel.conf fxsks=4 fxoks=1,2,3 loadzone=au defaultzone=au /etc/asterisk/zapata.conf # grep -v '^ *;' /etc/asterisk/zapata.conf | grep -v '^$' [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 Group=1 signalling=fxo_ks context=in-pbx channel=1-2 Group=2 echocancel=yes signalling=fxs_ks context=in-pstn channel=4 Group=3 signalling=fxo_ks context=in-spare channel=3 the thing that has me beet is that it work with the spa9000 I would expect it to just sort of work with the digium card. the os is debian amd64 2.6.26 #dpkg -l asteri* | grep ^ii ii asterisk1:1.4.21.2~dfsg-3 Open Source Private Branch Exchange (PBX) ii asterisk-barbarast.com 0.0.0-1 asterisk setup for hme1.samad.com.au ii asterisk-doc1:1.4.21.2~dfsg-3 Source code documentation for Asterisk ii asterisk-sounds-extra 1.4.7-1 Additional sound files for the Asterisk PBX ii asterisk-sounds-main1:1.4.21.2~dfsg-3 Core Sound files for Asterisk (English) #dpkg -l zapt* | grep ^ii ii zaptel 1:1.4.11~dfsg-3 zapata telephony utilities ii zaptel-modules-2.6.22-2-amd64 1:1.4.11~dfsg-3+2.6.22-4 zaptel modules for Linux (kernel 2.6.22-2-am ii zaptel-modules-2.6.26-2-amd64 1:1.4.11~dfsg-3+2.6.26-15 zaptel modules for Linux (kernel 2.6.26-2-am ii zaptel-source thanks Alex ___ -- 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
