Re: [Asterisk-Users] Tux/Asterisk logo for Cisco phones
Mark Phillips wrote: I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT! Anyone know where I can download this file please? http://www.loligo.com/asterisk/cisco/79xx/2003-04-27.examples/asterisk-tux.bmp ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] iax.cc opinion request
[EMAIL PROTECTED] wrote: On 7/10/2005, trixter wrote: I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Well - here we have a quandary. Opinion? Bad. (But so good) No outages that I can place on Sixtel - 24/7 rock solid - think a router hiccupped once for a couple of hours, but it wasn't theirs. Packet loss - again - as good or better that cell phones. Can't fault them (or him) there. UK termination (DID?) - can't say - thought they (or him) were US only. Tech support? Hahahahahahahahahahaha Ouch - my sides hurt! Took a month to get a DID. This pretty much sums it up for me as well. Except that it took two months for my DID to become active. On the other hand, I've had zero downtime and my 800 number was active within a day. I'm not noticing any problems with call quality either. They claimed in an email from early June to have instituted a new support responsiveness guarantee, but like I said earlier, since the DID went active I've had zero (!) problems. The server that you would be connecting to is iax2.sixtel.net, so run a few tracroutes from your site. Best, Dave ___ 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] Uniden UIP 200 and Asterisk.
Heath Oderman wrote: Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone is 172.28.184.105 In sip.conf - [uip200] username = heath secret = happy type = friend qualify = no host = dynamic defaultip = 172.28.184.105 dtmfmode = rfc2833 context = sip nat=no In unidenMAC.txt - # Sip Settings MyLcdDisplay 31521 MyDialNumber 703XXX DisplayName 31521 UserNameForProxy heath PasswordForProxy happy UserNameForRegistrar heath PasswordForRegistrar happy The output from asterisk is, of course: *CLI Jul 4 15:33:15 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Jul 4 15:33:45 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Jul 4 15:34:15 NOTICE[22905]: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.28.184.105' Remove the 'defaultip = 172.28.184.105' line from sip.conf. Do a 'sip reload' from the asterisk console and see if the registration works. Best, Dave ___ 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] Re: Polycom vs. Cisco IP Phones
Noah Miller wrote: My experience is that the Cisco and Polycom phones are both about in terms audio quality and useability. Neither one does exactly what I'd expect with respect to multiple lines. They both take a little extra setup in this regard, but you can read the wiki for that stuff. Snoms do exactly what I'd expect for a multiple line phone, are very easy to setup, but the audio quality and usability do not compare favorably with either Cisco or Polycom. If you've considered the Snom, you might also want to test a Zultys 4x4 or 4x5. I picked a 4x5 up off of ebay recently and have been pleasantly surprised by it. While I don't currently have a Polycom to compare it with, I would rank the audio quality equal to the Cisco's. It also just 'does the right thing' with multiple lines - only one registration, no hints needed. Can be configured through TFTP with both default and phone specific config files. Software updates are freely available from the Zultys website. ___ 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] 7912G via SIP, looking for comments
Marty Mastera wrote: I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation like this (built in switch, display, speaker phone...) I'm very happy with the one on my desk. Sound quality is the best I have found yet, and everything works well with SIP. Getting the firmware was not a pleasant experience, but eventually it all came through. TechData (800-237-8931) was the vendor I used for the $8 service contract described in the wiki. The handset has a good heft to it and sits firmly in the cradle, unlike, say, the Snom's. The buttons have an excellent feel to them and are legible at arm's length. What I like most about the phone, though, is the angle it sits at - I wish more manufacturers would realise that this is a significant benefit. There are 2 configuration issues that you'll want to be aware of. First is that you can not turn off the phone's built-in 'Forward to Voicemail' feature, you can only put an excessive delay on it so that the asterisk server gets a chance to handle things. Second is that the DTMF mode and silence suppression features that are controlled by the same setting in the config file. This is confusing because the Cisco docs want you to do bitwise math to figure out the correct setting. I use these in my gkdefault.txt file: AudioMode:0x0010 ForwardToVMDelay:200 One thing I don't like is the Cisco program for changing the logo that appears on the LCD. I haven't been able to find a version of Windows that it will run on. Doesn't work with wine or DOSBox either. I also considered the Polycom IP600 and the Zultys 4x5. Best, Dave ___ 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] OT: iax.cc/sixTel local DID question
Andrew Thompson wrote: When you choose to add an unlimited local DID to your account from their control panel, do you get to pick the prefix/NXX, or just the area code? Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. (I don't want to click the 910 area code to find out they giving me a Wilmington or Raleigh local DID that's just useless to me.) The form lets you choose the NXX. ___ 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] OT: iax.cc/sixTel local DID question
Andrew Thompson wrote: David Mallwitz wrote: Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. snip The form lets you choose the NXX. Actually, it didn't. I asked in a ticket what happens and the response came back that I would have gotten an email about it. I've sent the request back, so we'll see what happens. Odd. I signed up for a DID with them yesterday, and the form gave me a choice of several NXX's in my area. ___ 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] Re: grandstream bt100 upgrade 1.0.5.18
Rodney Acosta Coya wrote: [113] type=friend context=test username=113 fromuser=113 callerid=113 usecallerid=yes hidecallerid=no host=172.16.4.226 Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from 'sip:@172.16.4.249' failed for '172.16.4.226' You appear to be using DHCP to assign addresses to your phone - host=dynamic should fix your problem. Dave ___ 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] SIP Phones-Receptionist Setup
Bob Goddard wrote: Not all over $500 - a quick search finds: http://www.xenarcdirect.com/search_results.asp?txtsearchParamCat=6txtsearc hParamType=ALLtxtsearchParamMan=ALLtxtsearchParamVen=ALLiLevel=1 Product ID: 700TSCategory: 7 LCD Monitor 700TS - 7' USB Touch Screen LCD Monitor with VGA input Description: 4-wire Resistive Touch Screen (USB); VGA Input × 1; Supports 640 x 480 ~ 1600 x 1200 display resolution; For PC, Server, GPS, and Standard VGA Use; On Screen Display Control; Available in Silver or Black Price: $429.00 That is still too dear. http://www.mini-itx.com/store/?c=9#p503 I bought one of these Xenarc displays last year for a different devel project. If you turned it sideways and placed it next to the phone you would have a device that looks very similar to the Cisco 7914. There are some drawbacks to the display - the power supply connector is not well placed, and it only runs at 18 bit color depth, but it performed very well as a touchscreen monitor. I saw the option for a Xenarc driver in the last kernel compile I did, but haven't had a chance to check it out yet. There could be a couple of interesting uses for this as a receptionist device in addition to routing phone calls - monitoring security cameras, automated alert systems, setting up conferences... Basically, you'd need to create a hard phone with a vga output and a usb input using a cpu powerful enough to drive a display. Something like this: http://www.applieddata.net/products_bitsyX.asp , maybe. Set up a manager account for the whole thing... You know, this looks do-able. If anyone wants to take a run at it, I'm in. Dave ___ 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] VOIP security on an IAX connection.
Gregory Junker wrote: I use an OpenVPN tunnel as well, and IAX over that, and it works dandy. I highly recommend it. It's definately the easiest to configure, understand, and to get across diverse links. It is NAT-friendly, all UDP, etc. In my opinion, OpenVPN is to IPSEC as IAX is to SIP or H323. Does OpenVPN support PFS? Greg Perfect Forward Security? Yes, OpenVPN can easily be configured for dynamic re-keying at any specified interval and provides all the ciphers that the openssl library supports. I use and highly recommend it. Dave ___ 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] TDM400P hardware problems (fix)
Cirelle Enterprises wrote:
now in wcfxs.c on lines approximately 2127 or there abouts
this key needs to be added like so:
static struct pci_device_id wcfxs_pci_tbl[] = {
{ 0xe159, 0x0001, 0xa159, PCI_ANY_ID, 0, 0, (unsigned long) wcfxs },
{ 0xe159, 0x0001, 0xe159, PCI_ANY_ID, 0, 0, (unsigned long) wcfxs },
{ 0xe159, 0x0001, 0xb100, PCI_ANY_ID, 0, 0, (unsigned long) wcfxse },
{ 0xe159, 0x0001, 0xa9fd, PCI_ANY_ID, 0, 0, (unsigned long) wcfxsh },
{ 0xe159, 0x0001, 0xa904, PCI_ANY_ID, 0, 0, (unsigned long) wcfxsh },
{ 0xe159, 0x0001, 0xa900, PCI_ANY_ID, 0, 0, (unsigned long) wcfxsh },
{ 0xe159, 0x0001, 0xa901, PCI_ANY_ID, 0, 0, (unsigned long) wcfxsh },
{ 0 }
};
in this case this procedure needed to happen several times until 0xa904 allowed
the reboot (note the a901, a900)
recompile and see if it works for you...
Yep, I can verify this fixed it for me. Thanks for posting.
Dave
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