Re: [asterisk-users] Royalty for On Hold Music ?
On Tue, 31 Jul 2007, Steve Kennedy wrote: What if the radio is on in the background when I make a call ? is that rebroadcasting ? kind of gets blurry on the definitions there. That's not as you're listening to it and not trying to rebroadcast. I've not been following this thread closely, so apologies if this has already been covered. I had a summer job many years ago (early '90s) for the organisation responsible for collecting royalties in Ireland (IMRO). My recollection is probably a bit off, but the situation was that: - if you played copyrighted music on your phone system you needed a license which was scaled on the number of external channels on your phone system - if you had copyrighted music playing in the background in your office/shop/workplace then you needed a license which was scaled on the number of people working in your office/shop/workplace The reasoning behind both was that the employer was making (or allowing) the music available to third parties which was classed as a performance in a public place, which incurs a royalty fee (public == anything that's not domestic). It didn't matter whether the music came from TV, radio or a recording (and royalties were also levied on the TV, radio and recording companies). IIRC The licenses were typically an annual fee on the order of (back then) about IEP 100-200 (now EUR 127-254). AFAIR the situation was similar in the UK, where the Performing Rights Organisation (PRO) were the equivalent body. -Ronan ___ --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] Asterisk with BT's broadband voice service.
I've just signed up for BT's broadband voice service in the UK. Running a packet trace on their soft phone shows it trying to connect to btsip.bt.com using the BT account username / phone number. I've tried replicating this in my sip.conf but get a 404 when it tries to register. Has anybody managed to get an asterisk box registered with BT? If so, how's it done? -Ronan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - PSTN calls not connecting properly
Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-+ PSTN -+ * +- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940) I can make calls from the FXS port to the PSTN or my IAX service provider without any problems. I can make calls from my SIP phone to my IAX service provider, also without any problems. I can receive calls to the FXS port and SIP phone without any problems. However, when I call from my SIP phone to the PSTN my calls die, repeatedly, after 2-3 minutes. The display on the phone shows 'Session Progress (in 183)' for the duration of the call, rather than 'Connected', so it looks like the SIP phone is not recognising call connection on the PSTN. Output from the console is as follows: -- Executing Dial(SIP/ronan-5e0e, Zap/4/xxx) in new stack -- Called 4/xxx -- Hungup 'Zap/4-1' == Spawn extension (default, xxx, 1) exited non-zero on 'SIP/ronan-5e0e' A packet trace from the * box shows: ... 16.758516 192.168.2.9 - 192.168.2.30 UDP Source port: 12230 Destination port: 31042 16.758595 192.168.2.30 - 192.168.2.9 UDP Source port: 31042 Destination port: 12230 16.778540 192.168.2.9 - 192.168.2.30 UDP Source port: 12230 Destination port: 31042 16.779004 192.168.2.30 - 192.168.2.9 UDP Source port: 31042 Destination port: 12230 16.790884 192.168.2.30 - 192.168.2.9 SIP Request: CANCEL sip:[EMAIL PROTECTED];user=phone 16.791266 192.168.2.9 - 192.168.2.30 SIP Status: 487 Request Terminated 16.791477 192.168.2.9 - 192.168.2.30 SIP Status: 200 OK (192.168.2.9 is the * box, .30 is the phone) This has been going on for some time, but I've put up with it as the majority of my calls are short so it's not a big issue. As a result I'm unsure when the problem started, so I've no idea what change I made to the config that caused it. I'm fairly sure the change is on asterisk as I've not touched the config on the 7940 in a long time. My zaptel.conf, zapata.conf and sip.conf files are below, any suggestions or clue transfer would be much appreciated. -Ronan # zaptel.conf loadzone=uk defaultzone=uk fxsks=4 fxoks=1-3 # zapata.conf [channels] group = 0 context = incoming-POTS signalling = fxs_ks rxgain=10.0 txgain=6.0 echocancel=yes echocancelwhenbridged=no echotraining=300 immediate=no busydetect=no busycount=5 answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes callwaiting=yes relaxdtmf=no progzone=uk useincomingcalleridonzaptransfer = yes usecallerid=no callerid=asreceived cidsignalling=v23 cidstart=polarity ukcallerid=yes channel = 4 # sip.conf [general] allow=ulaw allow=alaw allow=gsm allow=g723.1 context=incoming recordhistory=yes port=5060 bindaddr=0.0.0.0 srvlookup=yes tos=lowdelay defaultexpirey=120 nat=no localnet=192.168.0.0/255.255.252.0 [ronan] regextension=ronan regcontext=4L [EMAIL PROTECTED] callerid=Ronan Mullally 100 restrictcid=no callgroup=1,2 pickupgroup=1,2 host=dynamic language=en type=friend context=default username=ronan secret=x fromdomain=4L.ie canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm allow=g723.1 qualify=100 accountcode=ronan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]
Hi Colin, Nice post - thanks, but: HARDWARE: 2 X 4-PCI slot PC's 6 X Black Box LB300A VDSL Ethernet Extenders 8 X PCI Ethernet cards Why not just install an ethernet switch on both ends that supports trunking / etherchannel? Less configuration, less chance for operator error, and no hard disks. You'll most likely also need a switch on each end *anyway*... -Ronan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]
On Fri, 7 Apr 2006, Ronan Mullally wrote: Why not just install an ethernet switch on both ends that supports trunking / etherchannel? Less configuration, less chance for operator error, and no hard disks. You'll most likely also need a switch on each end *anyway*... Before I get toasted through and through - you would still need the VDSL extenders of course, but the Linux boxes seem like overkill... -Ronan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Indications for Ireland
Hi Sean,
This is what I've got in my zaptel zonedata.c file for a small * box in
Dublin:
{ 18, ie, Ireland, { 400, 200, 400, 2000 },
{
/* Dialtone = 400//425//450 */
{ ZT_TONE_DIALTONE, 425 },
{ ZT_TONE_BUSY, 425/500,0/500 },
/* Ringtone = 400+450//425 */
{ ZT_TONE_RINGTONE, 400+450/400,0/200,400+450/400,0/2000 },
{ ZT_TONE_CALLWAIT, 425/180,0/200,425/200,0/4500 },
{ ZT_TONE_INFO, 950/330,1400/330,1800/330,0/1000 },
{ ZT_TONE_STUTTER, 350+440 },
{ ZT_TONE_CONGESTION, 400/400,0/350,400/225,0/525 },
{ ZT_TONE_DIALRECALL, 350+440 } },
},
-Ronan
On Wed, 14 Sep 2005, Sean Rima wrote:
Hello asterisk-users,
Just curious if anyone has the indications for Ireland, tried
googling for it to no avail.
Sean
--
+---+
|VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie |
|GPG Key http://thecivvie.fastmail.fm/thecivvie.asc |
+---+
Strange things happen under the midnight sun
when Men and Dogs go hunting for gold
___
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[Asterisk-Users] Matching on '@' in extensions
Hi, I'm using 1.0.7. I'm trying to write a rule in my dial plan to recognise and dial URLs. As far as I can see, the expression matching logic is not really up to this - all it can handle is Z, X, N and '.'. If I write a rule that matches '_.' then it starts catching calls to extensions like 'h', which makes me nervous. Is there any straigtforward way to match a @ in an extension? If not, are there any plans to add some normal regular expression functionality? -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS signalling for Ireland
Hi, I've installed a TDM400 card in an aging Dell Optiplex GXa (see my post a few weeks ago). The machine powers the card okay, it shows up in an 'lspci', and asterisk runs fine with it. I've tried both 1.0.4 and 1.0.5. The box in question is running a 2.4 kernel. However... I'm having trouble getting the card to work with a variaty of handsets attached to the FXS ports. One handset is a standard Telecom Eireann 'Boyne' handset, the other an NTL DECT handset purchased in the UK (RJ11 plug, works fine attached to the PSTN). Funnily enough, I've got an identical card installed in a machine in the UK attached to a Siemens Gigaset DECT phone purchased in Ireland, which is running without any trouble... I'm seeing the following rather wierd behaviour: - Incoming Pots calls, sent directly to the echo test app work fine - Outgoing calls from FXS ports to the PSTN work fine - Calls between FXS extensions work fine for the most part - Incoming calls sent to an FXS port will ring the FXS port. Upon pickup the bridging of the two fails. The incoming call continues to ring. If I drop the FXS, pick it up again, drop it, and pick it up a third time it will finally answer. The first two attempts result in a distorted ring tone being heard on the handset. I suspect the problem is down to FXS signalling. I've tried a variety of loop, kewl and ground start as well as EM. Does anybody have any suggestions regarding the zaptel / zapata config for these handsets? I'd none of these problems when I installed my TDM400 card in my UK box which runs 1.0.4 and a 2.6 kernel. Any suggestions welcome... -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISP connection to the PSTN using Asterisk
Hi Ashling, On Mon, 24 Jan 2005, Ashling O'Driscoll wrote: Could someone let me know the most common way that an Internet ISP would allow customers access to the PSTN?? Do they buy multiple fxo cards such as the TDM400P and rent multiple lines from a larger provider?? Do you mean how does an ISP allow its customers to make outbound voice calls using its infrastructure? Most ISPs these days will use something like a cisco AS5300 or AS5800, or an Ascend MAX TNT (it's been 5 years since I've played the ISP game, these model numbers are probably dating me ;-). They take multiple PRIs from the telco(s) which can be terminated as either individual B channels (for ISDN calls) or into digital modems (for analogue modem dialup). AFAIK the latter can be used as DSPs to permit in/outbound voice traffic. Would the best way be to connect to a third party voice/pstn gateway?? Is that simply a matter of forwarding all sip traffic destined for the pstn to another provider with a gateway and then they have to worry about the number of lines etc??And if that is the case, I presume no extra hardware is required? Depending on the economics involved this might be a sensible way of doing it - a third party could be enlisted to take the SIP traffic and deliver it to it's ultimate destination. The hand off would most likely be across a private interconnect (ethernet within a co-lo, or a dedicated circuit between sites) to ensure the link is not congested. -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 in aging Dell Optiplex
I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is successfully running an X100P card. I'm hoping to upgrade to a TDM400. Has anybody tried running these cards in old Optiplex machines? I'm not particularly worried about horsepower - more about the motherboard having a PCI bus that's able to power up the card... -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay before dialing extension on Zap channel
Hi, After using Asterisk with a SIP hardphone for a couple of weeks I've just installed a TDM400P card. My hardphone - a 7940 - allows me to use a dialplan to decide when a particular extension is complete and automatically trigger dialing. This works well with my internal extensions, which are all of the form Z00. When trying to dial these extensions from a handset connected to a Zap channel there appears to be a delay of about 4 seconds between the time I dial Z00 and the time asterisk decides I've finished dialing and connects me. Is there any way to reduce this delay? I'd ideally like asterisk to dial the extension as soon as it matches a valid extension. -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04B vs Dell revisited
On Jan 13, 2005, at 11:10 AM, Michael Swan wrote: A week or so ago I wrote about the problems I was having using a Digium TDM04B card in a Dell PowerEdge 750 IU running Fedora Core 1. Digium steadfastly indicates their card won't work in any PowerEdge 650, 700 and 750 series machines do (sic) to a failure in interrupt handling between the pci bus and the card. My question to the list is: for those of you using a TDM04B card, can you recommend a vendor/model of hardware to replace the Dell? I need it to be rack-mountable and, of course, to work well with *, preferably on FC 1. I saw this whilst booting my 2.6 machine yesterday - might be of use: PCI: Using ACPI for IRQ routing ** PCI interrupts are no longer routed automatically. If this ** causes a device to stop working, it is probably because the ** driver failed to call pci_enable_device(). As a temporary ** workaround, the pci=routeirq argument restores the old ** behavior. If this argument makes the device work again, ** please email the output of lspci to [EMAIL PROTECTED] ** so I can fix the driver. -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] kernel: Out of storage space while 900 MB free?
df -i - you're out of inodes on that filesystem. -Ronan On Wed, 1 Dec 2004, Roger Schreiter wrote: Hi, after loading the zaptel driver wct4xxp I have strange log lines in the syslog: Out of storage space. free tells, that more than 900 MB are still free. Disk space is also available. I'm using a dual opteron in 64 bit mode. Any ideas? Roger. Syslog: Dec 1 02:18:37 ipphone4 kernel: TE410P: Launching card: 0 Dec 1 02:18:37 ipphone4 kernel: TE410P: Setting up global serial parameters Dec 1 02:18:38 ipphone4 kernel: Found a Wildcard: Wildcard TE410P-Xilinx Dec 1 02:18:38 ipphone4 kernel: Registered tone zone 2 (France) Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 1 configured for CCS/HDB3/CRC4 Dec 1 02:18:38 ipphone4 kernel: SPAN 1: Primary Sync Source Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 2 configured for CCS/HDB3/CRC4 Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 3 configured for CCS/HDB3/CRC4 Dec 1 02:18:38 ipphone4 kernel: TE410P: Span 4 configured for CCS/HDB3/CRC4 Dec 1 02:18:38 ipphone4 kernel: SPAN 4: Quaternary Sync Source Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/2 Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/3 Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/4 Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/3 Dec 1 02:18:38 ipphone4 kernel: Zaptel: Master changed to TE4/0/1 Dec 1 02:18:48 ipphone4 kernel: Out of storage space Dec 1 02:18:48 ipphone4 last message repeated 121 times ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: TDMoE over bonded NIC's
On Mon, 22 Nov 2004, Kevin Brennan wrote: It happens servers come with twin GB NIC's, bonding is for redundancy not capacity. /Kev/ If all you're after is redunancy then have a look at failover rather than bonding - you'll have nothing to worry about with regard to frames potentially arriving out of order. -Ronan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens GSM terminal with Wildcard FXO
Hi Jafar, You want to look at Fixed Cellular Telephones (FCTs) like the Nokia Premicell or the Ericsson F251. As I understand it these present a PSTN interface which you can plug into an FXS interface. -Ronan On Sun, 7 Nov 2004, jafar mohammed wrote: Hi, I would like to implement GSM origination for a VOIP system i am developing. I am purchasing a Siemens M20 Terminal and would like to know if i can plug it into my Wildcard FXO device to get incoming GSM calls routed to the Asterisk server. If anyone has been able or successful in using this terminal please let me know. And if any of you have this terminal can you hook it up to a telephone headset and see if incoming calls will ring the headset. Thank you. __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens GSM terminal with Wildcard FXO
On Sun, 7 Nov 2004, Martin List-Petersen wrote: or you can look at chan_bluetooth (http://www.crazygreek.co.uk/content/chan_bluetooth). It's a work in progress, but seems to do the job, if you can get audio working. A cellphone and a bluetooth module are usually quite a lot cheaper than a GSM-to-PSTN adapter (usually 500 EUR and up) and a FXO or FXS (100 EUR and up) device. You can get FTCs for well under 500 Euros - I can get my hands on Ericsson F151s for about 230 Euros + VAT, assuming reasonable (a dozen or so) quantities. -Ronan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
