[asterisk-users] H.323 video support
Hi list! I asked this in this list some time ago, and now I was searching for evolution about this subject, but I found nothing. Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta brunch? If not, is this in the roadmap for 1.6 brunch? Regards. Diego. ___ -- 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] forwarding pots lines
will an ata directly connect to another remote ata thus emulating a long phone cord? also most of the ATA's I've seen drive a phone rather than accepting a line from the telco. Eric On Thu, May 22, 2008 at 10:17 PM, Alex Balashov [EMAIL PROTECTED] wrote: Eric Fort wrote: I'm looking for a simple hardware solution where I can connect POTS lines at one place and make them appear transparently at another location with only SIP and the internet between the locations. If I'm thinking this out right one location would need a box with a bunch of fxo interfaces and the other would need a box with a equal number of fxs interfaces. I'd like this to essentially emulate a really long piece of phone wire in as many ways as possible. What hardware should I use and what is the best way to provision this. I'd prefer to forgo the expense of 2 full asterisk servers as this seems unnecessary for the application. You can use devices called ATAs (Analogue Telephone Adaptors). They are much cheaper. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] forwarding pots lines
On Fri, May 23, 2008 at 10:04 AM, Eric Fort [EMAIL PROTECTED] wrote: will an ata directly connect to another remote ata thus emulating a long phone cord? also most of the ATA's I've seen drive a phone rather than accepting a line from the telco. Depending on the reliability needed (is this a way to talk to a girlfriend in another country or a mission-critical business use?) I'd say it's better to pay a small monthly fee to someone like OnSIP.com and use their centrex. If it's because you have the phone lines already installed and need to just use them at certain times, I do think there are FXO devices but I'm not sure they will help. You wouldn't need two asterisk servers at any rate but only one. The phones connect (through a router if need be) to the asterisk at the phone lines + FXO end. /r ___ -- 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] forwarding pots lines
Eric Fort wrote: will an ata directly connect to another remote ata thus emulating a long phone cord? also most of the ATA's I've seen drive a phone rather than accepting a line from the telco. Good, higher-end ATAs and IADS will be able to trunk to each other, and do FXO and FXS signaling on their analog ports. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Todays at 12 Noon EDT: Mike Trest on large volume calling with asterisk
Just before 12 Noon EDT, call : - SIP/[EMAIL PROTECTED],60,D(22622#1#) -or- (724) 444-7444 and enter 22622# 1# there is also a DNS: TS.X2Z.EU if you're too lazy to type 66.212.134.192 --- If you have a PIN, please use it rather than the 1# I didn't want to use Big and Fast as the subject and end up in everyone's spam bins, but that *is* the subject of today's conference. After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens Of Thousands of simultaneous calls can be made and Mike Trest has offered to take it all apart for us to look inside. Mike's recent projects include design and deployment of a 6,000 port voice bridge to support a Mobile application for SPRINT/NEXTEL. This platform is used to deliver NASCAR audio programming from inside the race cars to mobile subscribers. A similar project will support 100,000 ports for delivery of interactive entertainment media to mobile subscribers. All the details for hooking with us LIVE up can be found here: * http://VoipUsersConference.org -or- http://x2z.eu (permanent short URL) IRC for the back chatter is on Freenode.net #voip-users-conference If you want to try TringMe's click to call widget we talked about last week, you can find the demo here: * http://randulo.com/nwc/tringme/ It will only be active between 12 Noon and 1PM EDT and I think calls are limited to 20 minutes. Randy ___ -- 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] Adit 600 password reset
C F wrote: Then there is basicly no way to do this besides for cracking it? I Not that I am aware of, no. This subject went around several years back. They also talk about brute forcing the password as well. As far as I recall, nobody came back saying they were successful. have already figured out the username, now I just need to figure out the password. What is a good screen automation program that can bruteforce this for Windows? That I don't know. I suggest scanning the archives. It's been a few years and I may have missed the message. Do a search on Help need to reset Adit 600 for Asterisk There was a thread going on about this in November of 2005 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] H.323 video support
On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote: Hi list! I asked this in this list some time ago, and now I was searching for evolution about this subject, but I found nothing. Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta brunch? If not, is this in the roadmap for 1.6 brunch? Regards. Diego. When and where is the 1.6 brunch? ;-) Steve T ___ -- 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] Dear asterisk-users@lists.digium.com May 87% 0FF
On Fri, May 23, 2008 at 5:49 AM, VIAGRA (R) Official Site asterisk-users@lists.digium.com wrote: About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the Unsubscribe link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. (c)2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- 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 Is this the new DAHDI Viagra? 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
Re: [asterisk-users] asterisk virtualization on VMWARESX infrastructure
On Thu, May 22, 2008 at 9:51 PM, Sam Tam [EMAIL PROTECTED] wrote: Why if you have 50 operator then I would even consider using dual server running backup So the idea of using vmware may really be very risky, let alone not talk about performance issue well vmware will not be installed on a single machine, i intend an enterprise SX infrastructure with multiple nodes and auto failover policy. If Asterisk doens't suffer a virtualization, a service virtualized on a solid infrastructure is more scalable and hardware independent -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- 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] Dear asterisk-users@lists.digium.com May 87% 0FF
On Fri, May 23, 2008 at 3:19 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, May 23, 2008 at 5:49 AM, VIAGRA (R) Official Site asterisk-users@lists.digium.com wrote: About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the Unsubscribe link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. (c)2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- 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 Is this the new DAHDI Viagra? I think, spamfilter should ban every message mentioning Microsoft :p Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] forwarding pots lines
Two grandsreams, a 4008 and a 4108 would inexpensively do this for you. Instructions are on their site. -Original Message- From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 5/23/08 1:21 AM Subject: Re: [asterisk-users] forwarding pots lines Eric Fort wrote: I'm looking for a simple hardware solution where I can connect POTS lines at one place and make them appear transparently at another location with only SIP and the internet between the locations. If I'm thinking this out right one location would need a box with a bunch of fxo interfaces and the other would need a box with a equal number of fxs interfaces. I'd like this to essentially emulate a really long piece of phone wire in as many ways as possible. What hardware should I use and what is the best way to provision this. I'd prefer to forgo the expense of 2 full asterisk servers as this seems unnecessary for the application. You can use devices called ATAs (Analogue Telephone Adaptors). They are much cheaper. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410P install
We are trying to get our first B410P installation working but unable to get any L1 or L2 links. Connecting 3 ports to BRI lines via S/T interfaces on telco provided NT equipment. Using Debian etch 2.6.18-6 Ports are configured as TE Have tried both PTP and PTMP modes All 4 red lights are flashing. When we connect the ISDN cable, NOTHING HAPPENS and unable to receive or make calls. Have tried both Cross-over and straight through cables. Telco is adamant the ISDN lines are working fine. # misdn-init scan [OK] found the following devices: card=1,0x4 # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Interface is Poin-To-Point. - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) and so on The driver appears to load OK, except for some entries highlighted below. Syslog from starting the driver: May 23 22:27:22 cats kernel: Modular ISDN Stack core version (1_1_7_2) revision ($Revision: 1.40 $) May 23 22:27:22 cats kernel: mISDNd: kernel daemon started (current:c8ac7030) May 23 22:27:22 cats kernel: mISDNd: test event done May 23 22:27:22 cats kernel: ISDN L1 driver version 1.20 May 23 22:27:22 cats kernel: ISDN L2 driver version 1.32 May 23 22:27:22 cats kernel: mISDN: DSS1 Rev. 1.47 May 23 22:27:22 cats kernel: mISDN Capi 2.0 driver file version 1.21 May 23 22:27:23 cats kernel: mISDN: HFC-multi driver Rev. 1.68 May 23 22:27:23 cats kernel: HFC-multi: card manufacturer: 'Cologne Chip AG' card name: 'HFC-4S Digium Card' clock: normal May 23 22:27:23 cats kernel: PCI: Enabling device :00:0c.0 ( - 0003) May 23 22:27:23 cats kernel: ACPI: PCI Interrupt :00:0c.0[A] - GSI 17 (level, low) - IRQ 193 May 23 22:27:23 cats kernel: HFC-4S#1: defined at IOBASE 0xff00 IRQ 193 HZ 250 leds-type 2 May 23 22:27:23 cats kernel: HFC_multi: resetting HFC with chip ID=0xc revision=1 May 23 22:27:23 cats kernel: Setting GPIOs May 23 22:27:23 cats kernel: calling vpm_init May 23 22:27:23 cats kernel: VPM: Chip 0: ver 33 May 23 22:27:23 cats kernel: VPM: A-law mode May 23 22:27:23 cats kernel: VPM reg 0x20 is 11 May 23 22:27:23 cats kernel: NLP Thresh is set to 2 (0x2) May 23 22:27:23 cats kernel: VPM: A-law mode May 23 22:27:23 cats kernel: VPM reg 0x20 is 11 May 23 22:27:23 cats kernel: NLP Thresh is set to 2 (0x2) May 23 22:27:23 cats kernel: hfcpci_probe: DIPs(0x90) jumpers(0x0) May 23 22:27:23 cats kernel: register_layer: register_sysfs failed -17 st(0100) inst(4100) ç=== Bothered by these entries May 23 22:27:23 cats kernel: register_layer: register_sysfs failed -17 st(0200) inst(4200) May 23 22:27:23 cats kernel: register_layer: register_sysfs failed -17 st(0300) inst(4300) May 23 22:27:24 cats kernel: register_layer: register_sysfs failed -17 st(0400) inst(4400) May 23 22:27:24 cats kernel: 1 devices registered May 23 22:27:24 cats kernel: mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) May 23 22:27:24 cats kernel: mISDN_dsp: DSP clocks every 128 samples. This equals 4 jiffies. CLI misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 2 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 3 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 CLI As usual its rather urgent to get it running. If anyone can provide suggestions ,Id be very grateful. Chris ___ -- 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] forwarding pots lines
Sorry to jump in on this but I am also interested in this topic. In my scenario I have about 10 POTs lines brought into the front of a facility and the only infrastructure connecting the back of the facility is a 3000ft fiber backhaul. I've been asked to bring the POTs lines to the back of the facility. Are there any ATAs that trunk multiple POTs Lines? Like a multiplexer of some sort. If anyone has any information can you please provide the manufacturer and model of the device? Thank You, Dennis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, May 23, 2008 4:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines On Fri, May 23, 2008 at 10:04 AM, Eric Fort [EMAIL PROTECTED] wrote: will an ata directly connect to another remote ata thus emulating a long phone cord? also most of the ATA's I've seen drive a phone rather than accepting a line from the telco. Depending on the reliability needed (is this a way to talk to a girlfriend in another country or a mission-critical business use?) I'd say it's better to pay a small monthly fee to someone like OnSIP.com and use their centrex. If it's because you have the phone lines already installed and need to just use them at certain times, I do think there are FXO devices but I'm not sure they will help. You wouldn't need two asterisk servers at any rate but only one. The phones connect (through a router if need be) to the asterisk at the phone lines + FXO end. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] B410P install
On Fri, May 23, 2008 at 8:41 AM, Chris Curtis [EMAIL PROTECTED] wrote: We are trying to get our first B410P installation working but unable to get any L1 or L2 links. Connecting 3 ports to BRI lines via S/T interfaces on telco provided NT equipment. Using Debian etch 2.6.18-6 Ports are configured as TE Have tried both PTP and PTMP modes All 4 red lights are flashing. When we connect the ISDN cable, NOTHING HAPPENS and unable to receive or make calls. Have tried both Cross-over and straight through cables. Telco is adamant the ISDN lines are working fine. # misdn-init scan [OK] found the following devices: card=1,0x4 # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Interface is Poin-To-Point. - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) and so on The driver appears to load OK, except for some entries highlighted below. Syslog from starting the driver: May 23 22:27:22 cats kernel: Modular ISDN Stack core version (1_1_7_2) revision ($Revision: 1.40 $) May 23 22:27:22 cats kernel: mISDNd: kernel daemon started (current:c8ac7030) May 23 22:27:22 cats kernel: mISDNd: test event done May 23 22:27:22 cats kernel: ISDN L1 driver version 1.20 May 23 22:27:22 cats kernel: ISDN L2 driver version 1.32 May 23 22:27:22 cats kernel: mISDN: DSS1 Rev. 1.47 May 23 22:27:22 cats kernel: mISDN Capi 2.0 driver file version 1.21 May 23 22:27:23 cats kernel: mISDN: HFC-multi driver Rev. 1.68 May 23 22:27:23 cats kernel: HFC-multi: card manufacturer: 'Cologne Chip AG' card name: 'HFC-4S Digium Card' clock: normal May 23 22:27:23 cats kernel: PCI: Enabling device :00:0c.0 ( - 0003) May 23 22:27:23 cats kernel: ACPI: PCI Interrupt :00:0c.0[A] - GSI 17 (level, low) - IRQ 193 May 23 22:27:23 cats kernel: HFC-4S#1: defined at IOBASE 0xff00 IRQ 193 HZ 250 leds-type 2 May 23 22:27:23 cats kernel: HFC_multi: resetting HFC with chip ID=0xc revision=1 May 23 22:27:23 cats kernel: Setting GPIOs May 23 22:27:23 cats kernel: calling vpm_init May 23 22:27:23 cats kernel: VPM: Chip 0: ver 33 May 23 22:27:23 cats kernel: VPM: A-law mode May 23 22:27:23 cats kernel: VPM reg 0x20 is 11 May 23 22:27:23 cats kernel: NLP Thresh is set to 2 (0x2) May 23 22:27:23 cats kernel: VPM: A-law mode May 23 22:27:23 cats kernel: VPM reg 0x20 is 11 May 23 22:27:23 cats kernel: NLP Thresh is set to 2 (0x2) May 23 22:27:23 cats kernel: hfcpci_probe: DIPs(0x90) jumpers(0x0) May 23 22:27:23 cats kernel: register_layer: register_sysfs failed -17 st(0100) inst(4100) ç=== Bothered by these entries May 23 22:27:23 cats kernel: register_layer: register_sysfs failed -17 st(0200) inst(4200) May 23 22:27:23 cats kernel: register_layer: register_sysfs failed -17 st(0300) inst(4300) May 23 22:27:24 cats kernel: register_layer: register_sysfs failed -17 st(0400) inst(4400) May 23 22:27:24 cats kernel: 1 devices registered May 23 22:27:24 cats kernel: mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) May 23 22:27:24 cats kernel: mISDN_dsp: DSP clocks every 128 samples. This equals 4 jiffies. CLI misdn show stacks BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 2 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 * Port 3 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 CLI As usual its rather urgent to get it running. If anyone can provide suggestions ,I'd be very grateful. Chris Is your card compatible with BRIStuff? Thanks, Steve Totaro ___ -- 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] B410P install
As usual its rather urgent to get it running. If anyone can provide suggestions ,I'd be very grateful. Chris Is your card compatible with BRIStuff? Thanks, Steve Totaro Don't forget to modprobe qodah ;-) Steve T ___ -- 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] forwarding pots lines
There are a couple of companies out there that make 24 port fxo and fxs boxes. If you have some unused fibers you cout do this very reliably with two channel banks... One with fxs ports and the other with fxo ports and t1 media converters. The grand stream solution mentioned in an earlier post does 8 ports, you could get one 4 port model and one 8 port model of fxs and the same of fxo and accomplish your goal rather inexpensively as well. Joe -Original Message- From: Dennis P. Clark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 5/23/08 8:43 AM Subject: Re: [asterisk-users] forwarding pots lines Sorry to jump in on this but I am also interested in this topic. In my scenario I have about 10 POTs lines brought into the front of a facility and the only infrastructure connecting the back of the facility is a 3000ft fiber backhaul. I've been asked to bring the POTs lines to the back of the facility. Are there any ATAs that trunk multiple POTs Lines? Like a multiplexer of some sort. If anyone has any information can you please provide the manufacturer and model of the device? Thank You, Dennis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, May 23, 2008 4:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines On Fri, May 23, 2008 at 10:04 AM, Eric Fort [EMAIL PROTECTED] wrote: will an ata directly connect to another remote ata thus emulating a long phone cord? also most of the ATA's I've seen drive a phone rather than accepting a line from the telco. Depending on the reliability needed (is this a way to talk to a girlfriend in another country or a mission-critical business use?) I'd say it's better to pay a small monthly fee to someone like OnSIP.com and use their centrex. If it's because you have the phone lines already installed and need to just use them at certain times, I do think there are FXO devices but I'm not sure they will help. You wouldn't need two asterisk servers at any rate but only one. The phones connect (through a router if need be) to the asterisk at the phone lines + FXO end. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Thanks :-D change the context to default and everithing works fine. I assigned the sip context because that was the context on the example. Thanks :-) Nomar Alex Balashov wrote: Nomar Mora wrote: Alex Balashov wrote: Do you have dial plan routes for internal extension calls? Do you mean if I have configured the extension.conf? Yes, I config the extensions on the extension.conf file otherwise, no I have not. Thanks in Advance Nomar In the 'sip' context? -- 2008 Año del satélite Simón Bolívar ___ -- 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] New York Asterisk Users
This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many Asterisk users there are in NY we'll work out what to do from there. Cheers, Dean ___ -- 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] H.323 video support
Yes, you are right... sorry for my fast and poor English. I rewrite my questions: Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta branch? If not, is this in the roadmap for 1.6 branch? Regards. 2008/5/23 Steve Totaro [EMAIL PROTECTED]: On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote: Hi list! I asked this in this list some time ago, and now I was searching for evolution about this subject, but I found nothing. Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta brunch? If not, is this in the roadmap for 1.6 brunch? Regards. Diego. When and where is the 1.6 brunch? ;-) Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] B410P install
I don't really know as I am unfamiliar with BRIstuff. If fact the whole ISDN world is a new one for me as its not that common for small business here in Australia. We are using the Digium B410P. A quick Google of B410P and BRIstuff is inconclusive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 23 May 2008 10:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] B410P install Is your card compatible with BRIStuff? Thanks, Steve Totaro ___ -- 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] Extension not found
Dear Randulo, Thanks for your suggention. Now i am able to communicate between 2 computers. Regards, Baskar --- randulo [EMAIL PROTECTED] wrote: On Mon, May 19, 2008 at 8:44 AM, bas karan [EMAIL PROTECTED] wrote: [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 handle_request_invite: Call from 'Phone3' to extension '5' rejected because extension not found. -- Registered SIP 'Phone3' at 192.168.1.101 port Extension.conf enteries are, exten = 3,1,Dial(SIP/Phone3,30,tr) exten = 4,1,Dial(SIP/Phone4,30,tr) exten = 5,1,Dial(SIP/Phone5,30,tr) Where is the [sip] context named in the phones context= statement ? ___ -- 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 Has your work life balance shifted? Find out - http://in.search.yahoo.com/search?fr=na_onnetwork_mail_taglinesei=UTF-8rd=r1p=work+life+balance ___ -- 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] Adit 600 password reset
Quoting Doug Lytle [EMAIL PROTECTED]: C F wrote: Then there is basicly no way to do this besides for cracking it? I Not that I am aware of, no. This subject went around several years back. They also talk about brute forcing the password as well. As far as I recall, nobody came back saying they were successful. have already figured out the username, now I just need to figure out the password. What is a good screen automation program that can bruteforce this for Windows? I had the same problem with one of mine. I smply forgot the password. I seem to recall that the adit had a flaw in it, where it was obvious by the error message returned if you had the correct length username and password, which should make your brute-force attempt much easier. --Shane ___ -- 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] B410P install
On Fri, 2008-05-23 at 22:41 +1000, Chris Curtis wrote: We are trying to get our first B410P installation working but unable to get any L1 or L2 links. Which mISDN and kernel version are you using? There was a problem with very recent kernels mISDN so you might want to check the mISDN mailing list archives[1]. If your kernel version is 2.6.25 or newer than mISDN 1.1.7.2 will not work and you will need to install mISDN from git cause Christian committed some fixes[2]. The card does come with support from Digium. Have you tried calling them? Regards, Patrick [1] https://www.isdn4linux.de/mailman/listinfo/isdn4linux [2] http://www.isdn4linux.de/pipermail/isdn4linux/2008-May/003441.html ___ -- 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/OpenSER users in Porto, Portugal?
Friends, I will be spending a few days in Porto, Portugal in the beginning of June. Any Asterisk and/or OpenSER users there that wants to go out and have dinner and Open Source Voip talk? Respond off list, and we'll see if we can meet. Have a nice weekend! /Olle ___ -- 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] New York Asterisk Users
Do you mean the city or the state of New York? I'm in NY, but a long ass way from NYC. This is an email to all* New York* based Asterisk users. For some time it’s been bugging me that we don’t have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I’m going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many Asterisk users there are in NY we’ll work out what to do from there. Cheers, Dean ___ -- 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 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 269.24.0/1462 - Release Date: 5/23/2008 7:20 AM ___ -- 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] (no subject)
In the setup tutorial @ http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation it states the potential issue regarding setting up UniqueID as the primary key, but doesn't state how to rectify this? What is the proper way to make sure this is done right? Also, has anyone built a simple front end for non technical folk to utilize for accessing the data simply for overview when billing etc is not important (small company)? Thanks! jlc ___ -- 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] H.323 video support
Remind me to pick on your poor Spanish next time I see you for a mid-morning meal. :) Steve Totaro wrote: On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote: When and where is the 1.6 brunch? ;-) ___ -- 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] New York Asterisk Users
:) Same here On Fri, May 23, 2008 at 10:13 AM, Adam Moffett [EMAIL PROTECTED] wrote: Do you mean the city or the state of New York? I'm in NY, but a long ass way from NYC. This is an email to all* New York* based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many Asterisk users there are in NY we'll work out what to do from there. Cheers, Dean ___ -- 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 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 269.24.0/1462 - Release Date: 5/23/2008 7:20 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New York Asterisk Users
Hey Adam, Yes I was thinking NYC - basically I was surprised at the lack of response about Ming from Voiceroute wanting to organize a physical meeting event (btw it got moved to the 2nd of June) This bugged me as when you look at other opensource community groups in NY I belong to they have a much stronger face to face relationship with each other. So short answer is yes if you want to get together with each other in NYC on a face to face basis this is what I had in mind. Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Friday, 23 May 2008 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New York Asterisk Users Do you mean the city or the state of New York? I'm in NY, but a long ass way from NYC. This is an email to all* New York* based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many Asterisk users there are in NY we'll work out what to do from there. Cheers, Dean ___ -- 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 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 269.24.0/1462 - Release Date: 5/23/2008 7:20 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adit 600 password reset
On Fri, May 23, 2008 at 01:20:58AM -0400, C F wrote: serial, I don't know the IP address. Loopback cable. nmap -sP 0/1 nmap -sP 1/1 Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] reload stopping EVERYTHING on CLI and causing havoc.
Also what do I do if I see deadlocks all over the CLI? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: May 23, 2008 12:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Hi Sherwood, I've done the backtrace. Maybe you can submit yours too. http://bugs.digium.com/view.php?id=12709 Thanks, Mark. Original Message Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. From: Sherwood McGowan [EMAIL PROTECTED] Date: Thu, May 22, 2008 7:13 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com snip True True...it's only been a minor annoyance for me, but in the interest of improving Asterisk I should go ahead and rebuild using the debug settings and submit a backtrace. Sherwood MCGowan snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer
Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So - is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that's OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin... the carriers here don't like it if your sending CLI for other countries, that don't match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine... lol Thanks, Adrian ___ -- 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 chan Skype
Hello! Iam configuring chan Skype on my asterisk box, doing some test calls I saw that asterisk answer the calls but hungs up before the call are stablished. Is this a license problem? Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- 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] Adit 600 password reset
yes that's how I figured out the username. since it returned incorrect login before the password prompt on the wrong username. I a don't know the password however. On 5/23/08, Shane Young [EMAIL PROTECTED] wrote: Quoting Doug Lytle [EMAIL PROTECTED]: C F wrote: Then there is basicly no way to do this besides for cracking it? I Not that I am aware of, no. This subject went around several years back. They also talk about brute forcing the password as well. As far as I recall, nobody came back saying they were successful. have already figured out the username, now I just need to figure out the password. What is a good screen automation program that can bruteforce this for Windows? I had the same problem with one of mine. I smply forgot the password. I seem to recall that the adit had a flaw in it, where it was obvious by the error message returned if you had the correct length username and password, which should make your brute-force attempt much easier. --Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adit 600 password reset
seen that thread, it doesn't help me much since I only have seriel access and no linux machine with a seriel port On 5/23/08, Doug Lytle [EMAIL PROTECTED] wrote: C F wrote: Then there is basicly no way to do this besides for cracking it? I Not that I am aware of, no. This subject went around several years back. They also talk about brute forcing the password as well. As far as I recall, nobody came back saying they were successful. have already figured out the username, now I just need to figure out the password. What is a good screen automation program that can bruteforce this for Windows? That I don't know. I suggest scanning the archives. It's been a few years and I may have missed the message. Do a search on Help need to reset Adit 600 for Asterisk There was a thread going on about this in November of 2005 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] forwarding pots lines
Will fax and dial-up internet work through the gateway? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Friday, May 23, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines There are a couple of companies out there that make 24 port fxo and fxs boxes. If you have some unused fibers you cout do this very reliably with two channel banks... One with fxs ports and the other with fxo ports and t1 media converters. The grand stream solution mentioned in an earlier post does 8 ports, you could get one 4 port model and one 8 port model of fxs and the same of fxo and accomplish your goal rather inexpensively as well. Joe -Original Message- From: Dennis P. Clark [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 5/23/08 8:43 AM Subject: Re: [asterisk-users] forwarding pots lines Sorry to jump in on this but I am also interested in this topic. In my scenario I have about 10 POTs lines brought into the front of a facility and the only infrastructure connecting the back of the facility is a 3000ft fiber backhaul. I've been asked to bring the POTs lines to the back of the facility. Are there any ATAs that trunk multiple POTs Lines? Like a multiplexer of some sort. If anyone has any information can you please provide the manufacturer and model of the device? Thank You, Dennis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, May 23, 2008 4:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines On Fri, May 23, 2008 at 10:04 AM, Eric Fort [EMAIL PROTECTED] wrote: will an ata directly connect to another remote ata thus emulating a long phone cord? also most of the ATA's I've seen drive a phone rather than accepting a line from the telco. Depending on the reliability needed (is this a way to talk to a girlfriend in another country or a mission-critical business use?) I'd say it's better to pay a small monthly fee to someone like OnSIP.com and use their centrex. If it's because you have the phone lines already installed and need to just use them at certain times, I do think there are FXO devices but I'm not sure they will help. You wouldn't need two asterisk servers at any rate but only one. The phones connect (through a router if need be) to the asterisk at the phone lines + FXO end. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proposed changes for queue timeout
Hello, I've been annoyed quite some time by behavior of queue timeout (specified as argument to Queue app). Basically if I specify timeout for queue 5 minutes, and ring time to agent for 15 seconds, and ring to agent starts at 4:59, agent will receive ring only for 1 second, after which call attempt will terminate. So, the question is - if anybody needs exact queue timing, with possibility that agent calls are terminated without finishing ring timeout? Please see issue http://bugs.digium.com/view.php?id=12690 - there's table of calculations, which explains how values are calculated now, and how I'm proposing. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] (no subject)
the subject of this thread has been on this list way too many times just search the archives. On 5/23/08, Joseph L. Casale [EMAIL PROTECTED] wrote: In the setup tutorial @ http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation it states the potential issue regarding setting up UniqueID as the primary key, but doesn't state how to rectify this? What is the proper way to make sure this is done right? Also, has anyone built a simple front end for non technical folk to utilize for accessing the data simply for overview when billing etc is not important (small company)? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New York Asterisk Users
I am in Philadelphia, keep me updated and will try to make time to attend. On Fri, May 23, 2008 at 10:25 AM, Dean Collins [EMAIL PROTECTED] wrote: Hey Adam, Yes I was thinking NYC - basically I was surprised at the lack of response about Ming from Voiceroute wanting to organize a physical meeting event (btw it got moved to the 2nd of June) This bugged me as when you look at other opensource community groups in NY I belong to they have a much stronger face to face relationship with each other. So short answer is yes if you want to get together with each other in NYC on a face to face basis this is what I had in mind. Regards, Dean Collins [EMAIL PROTECTED] Cognation Limited +1-212-203-4357 +61-2-9016-4652 (Sydney indial) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Friday, 23 May 2008 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New York Asterisk Users Do you mean the city or the state of New York? I'm in NY, but a long ass way from NYC. This is an email to all* New York* based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many Asterisk users there are in NY we'll work out what to do from there. Cheers, Dean ___ -- 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 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 269.24.0/1462 - Release Date: 5/23/2008 7:20 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc.
Nevermind. Attached atleast two backtraces, one with Asterisk running and not coredumping, and two with Asterisk built coredumps using DO_CRASH. If anyone is interested, please check http://bugs.digium.com/view.php?id=12709 Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: May 23, 2008 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Also what do I do if I see deadlocks all over the CLI? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: May 23, 2008 12:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Hi Sherwood, I've done the backtrace. Maybe you can submit yours too. http://bugs.digium.com/view.php?id=12709 Thanks, Mark. Original Message Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. From: Sherwood McGowan [EMAIL PROTECTED] Date: Thu, May 22, 2008 7:13 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com snip True True...it's only been a minor annoyance for me, but in the interest of improving Asterisk I should go ahead and rebuild using the debug settings and submit a backtrace. Sherwood MCGowan snip ___ -- 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
[asterisk-users] Strange State 6 on Channel X
In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard ___ -- 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] reload stopping EVERYTHING on CLI and causing havoc.
Mark Hamilton wrote: Also what do I do if I see deadlocks all over the CLI? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: May 23, 2008 12:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Hi Sherwood, I've done the backtrace. Maybe you can submit yours too. http://bugs.digium.com/view.php?id=12709 Thanks, Mark. Original Message Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. From: Sherwood McGowan [EMAIL PROTECTED] Date: Thu, May 22, 2008 7:13 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com snip True True...it's only been a minor annoyance for me, but in the interest of improving Asterisk I should go ahead and rebuild using the debug settings and submit a backtrace. Sherwood MCGowan snip ___ -- 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 There's a section on debugging a running deadlock on this page: http://www.voip-info.org/wiki-Asterisk+debugging That's the route you'll probably want to go ___ -- 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] Strange State 6 on Channel X
In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard ___ -- 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] Strange State 6 on Channel X
Daniel Lockard wrote: In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard ___ -- 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've wondered the same thing, but it hasn't caused me any issues. I've searched google quite a few times, haven't found anything useful. Sherwood McGowan ___ -- 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] reload stopping EVERYTHING on CLI and causing havoc.
Mark Hamilton wrote: Nevermind. Attached atleast two backtraces, one with Asterisk running and not coredumping, and two with Asterisk built coredumps using DO_CRASH. If anyone is interested, please check http://bugs.digium.com/view.php?id=12709 Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: May 23, 2008 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Also what do I do if I see deadlocks all over the CLI? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: May 23, 2008 12:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. Hi Sherwood, I've done the backtrace. Maybe you can submit yours too. http://bugs.digium.com/view.php?id=12709 Thanks, Mark. Original Message Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing havoc. From: Sherwood McGowan [EMAIL PROTECTED] Date: Thu, May 22, 2008 7:13 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com snip True True...it's only been a minor annoyance for me, but in the interest of improving Asterisk I should go ahead and rebuild using the debug settings and submit a backtrace. Sherwood MCGowan snip ___ -- 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 I'm monitoring the issue, and will submit a backtrace from my system as well. ___ -- 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] OOH323 to Avaya S8500?
Has anyone tried using ooh323 in Asterisk to talk H.323 to an Avaya S8500 running Communications Manager 4 software? I have a potential customer who has such a system, and wants an Asterisk box to talk to it. Apparently they don't have SIP installed. I've successfully got ooh323 talking between two Asterisk boxes, so am just after some confidence regarding the Avaya, or any gotchas to be aware of. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] Strange State 6 on Channel X
Well, then it might not be that that is causing me issues? I have no idea why I would be able to call in, hear ringing on my phone, and then have the CLI tell me that it has answered... Daniel Lockard On 5/23/08, Sherwood McGowan [EMAIL PROTECTED] wrote: Daniel Lockard wrote: In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard ___ -- 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've wondered the same thing, but it hasn't caused me any issues. I've searched google quite a few times, haven't found anything useful. Sherwood McGowan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom LDAP Corporate Directory
Any more information on this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of faraz Sent: Friday, April 18, 2008 6:30 PM To: Watkins, Bradley Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory please do!. how much did the 50 cost you? On Fri, 2008-04-18 at 18:22 -0400, Watkins, Bradley wrote: I actually just ordered 50 licenses to give this and the other applications a try. I'll post my results to the list once I get them and have had a chance to play around. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of faraz Sent: Friday, April 18, 2008 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory I havent tried it. I have quite a few polycoms and didnt even know polycom had this feature! :) This is obviously a separate peice of software that must be purchased and installed on the phones. Looks amazing though- any idea on pricing?. On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote: Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip /applications/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange State 6 on Channel X
I posted about this exact same problem about a month or two ago and got no replies. The problem was an A200D with Wanpipe 3.2.1. We've since made some changes to this installation and the problems have gone away. The changes were specifically getting a new local loop from the telco on the problem lines and recabling from the demarc to the * box. So, check your cabling carefully to make sure it is in good repair. If everything checks out, hopefully you can convince your local telco to get you a new loop. Luckily, the telco we use primarily in our area is relatively small and has fantastic support. Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Danny Lockard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 23, 2008 11:54:09 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Strange State 6 on Channel X Well, then it might not be that that is causing me issues? I have no idea why I would be able to call in, hear ringing on my phone, and then have the CLI tell me that it has answered... Daniel Lockard On 5/23/08, Sherwood McGowan [EMAIL PROTECTED] wrote: Daniel Lockard wrote: In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard ___ -- 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've wondered the same thing, but it hasn't caused me any issues. I've searched google quite a few times, haven't found anything useful. Sherwood McGowan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer
Adrian Marsh wrote: Hi All, In my old telco days (SS7), if I was wanting to hand back a call to the network for transfer to a different PSTN number, there was a specific SS7 action I could take, which send the call back to the network, which in turn then routed the call appropriately. It added a transfer-number into the SS7 headers so that the originating number, dialed number and transfer number all stayed to specs, and everyone was happy. In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to have at least the control packets go via my SIP server), and use a Dial out to the far end. So – is there a way of handing the call back to the network in asterisk ? My detailed problem is this: When a call comes in, I want to send it onto users mobiles, if I hairpin the call that’s OK, except the CLI needs to be that of the originator (from the USERS point of view) so they can decide if they want to accept the call. Here in the UK, this is where the issues begin… the carriers here don’t like it if your sending CLI for other countries, that don’t match what they think they should receive from that connecting carrier. Eg, if a call coming to them is 13 digits, but they only expect 11 from that carrier, then they cut the digits. This turns a US originated call into a Southampton UK originated call! So I was hoping that handing the call back to the network in the traditional sense would make it their problem and not mine… lol Thanks, Adrian ___ -- 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 http://www.google.com/search?hl=enq=asterisk+302+redirect+sipbtnG=Search I believe you're looking for a 302 Redirect? Sorry if you're not, but that sounds like what you want ___ -- 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-dev] Asterisk 1.6 Realtime Database must use ', ' not '|'
On Fri, May 23, 2008 at 01:25:43PM -0400, Donny Kavanagh wrote: This is getting downright abusive, and is totally uncalled for, this is not a list for personal attacks. You thought that Steve suggesting JT step in was abusive? If that's not what you meant, then you need to either a) be clearer, or b) reply to the proper message. And hackers ignoring pleasantries to get right down to the technical issues isn't abusive at all. See Jargon File; see also Asperger's Syndrome, How To Ask Good Questions. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Weird NAT issue ...
sorry for not replying to this sooner! but the canreinvite=no trick worked a treat. thank you -- Alan Williamson Try our free registrationless email/sms reminder http://yourli.st/ b: http://alan.blog-city.com/ Steve Davies wrote: If the two phones attempt to refer to each other using their external (NAT) IP addresses rather that their internal addresses, then it will all go horribly wrong. You do not provide enough information about asterisk IP addresses or firewalls for a possible solution, but assuming you are using SIP and asterisk, you could try canreinvite=no against the 2 phones to see if keeping the Asterisk server in-the-loop helps. Also look on the VoIP wiki for externip and localnet in the sip.conf configuration. Regards, Steve On Mon, Mar 17, 2008 at 1:59 PM, Alan Williamson [EMAIL PROTECTED] wrote: Afternoon one and all. I am having some interesting fun with our Asterisk setup. We have two CISCO handsets (7960) sitting on the same network (NAT). Each phone can successfully originate calls. Each phone can be called successfully from outside Each phone can be directly called by other extensions OUTSIDE the network HOWEVER -- when those 2 phones try to call each other; the connection is made, but no voice is heard. Any advice as to where i need to look? ___ -- 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] Discover connected Zap lines
2008/5/22 C. Chad Wallace [EMAIL PROTECTED]: When it says FXS only, I think it's reasonable to assume that FXO is excluded. FXS is the signalling of FXO cards. I have only FXO cards. -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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 virtualization on VMWARESX infrastructure
Hi, If Asterisk doens't suffer a virtualization, a service virtualized on a solid infrastructure is more scalable and hardware independent at the beginning of my project I was thinking to do so too (with Xen), but I was told that delays etc. in a virtualized environment will be a significant problem on some load! So finally every server has installed one Asterisk and should serve at about 600 users. On my private system I'm going to run Asterisk under a Xen-DomU, but it is only one Asterisk with three phones and two lines :-) -- Chau y hasta luego, Thorolf ___ -- 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] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
This does not do the trick, because while the voice path is not created until the digit 1 is dialed, when the first extension picks up the others stop ringing. What is needed is something where all extensions continue ringing until the digit is dialed. On Mon, May 12, 2008 at 10:54 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: srv04*CLI show application Dial srv04*CLI -= Info about application 'Dial' =- [Synopsis] Place a call and connect to the current channel *SNIP* p- This option enables screening mode. This is basically Privacy mode without memory. P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if it is provided. The current extension is used if a database family/key is not specified. n- This option is a modifier for the screen/privacy mode. It specifies that no introductions are to be saved in the priv-callerintros directory. N- This option is a modifier for the screen/privacy mode. It specifies that if callerID is present, do not screen the call. On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote: GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? ___ -- 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
[asterisk-users] dialplan syntax error: need new eyes
I'm trying to set the outgoing caller id to the DID number, but only if
the extension is greater than 140. MAINSTUB is simply the first 7 digits
of the main number. sip.conf sets the CALLERID(num) to the extension.
exten =_1NXXNXX,n,Set(CALLERID(num)=${MAINSTUB}${CALLERID(num)})
works. But I want to set the caller id to the main number unless the
extension is 141 or higher.
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
ast_yyerror(): syntax error: syntax error, unexpected '', expecting
$end; Input:
140
I've counted my parens, checked IF syntax, and now need some new eyes to
look at this.
Thanks.
sean
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[asterisk-users] *#%! Polycom...
I used to do lots of Asterisk, but got an offer I couldn't refuse, and went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want to set up a test system. One thing I'd really like to get my hands on is recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old source, seems to have been kaput about as long as I've been a sysadmin; are there any other sources out there? (And, yeah, if anyone wants to e-mail them to me directly, I won't say no.) Thanks much, -Ken ___ -- 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] forwarding pots lines
On May 23, 2008 11:25:55 am Dennis P. Clark wrote: Will fax and dial-up internet work through the gateway? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll Sent: Friday, May 23, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -Non-Commercial Discussion Subject: Re: [asterisk-users] forwarding pots lines There are a couple of companies out there that make 24 port fxo and fxs boxes. If you have some unused fibers you cout do this very reliably with two channel banks... One with fxs ports and the other with fxo ports and t1 media converters. The grand stream solution mentioned in an earlier post does 8 ports, you could get one 4 port model and one 8 port model of fxs and the same of fxo and accomplish your goal rather inexpensively as well. In generaly this is a bad idea (especially dialup internet). If both the gateways you use support T.38 origination/termination then faxing will not be a problem at all. However, in your case I assume you are only transporting the calls over LAN, and there is no WAN/Internet involved... which means you will probably achive a high success rate for both dialup and fax... I wouldn;t be surprised if you can;t max out the baud on your dialup internet connections though... i'd expect a slight reduction in speed (and errors, though error correction built into your modem would hopefully take care of this, at the cost of a a little speed due to re-transmissions) -- Matt http://www.mattgwatson.ca ___ -- 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] *#%! Polycom...
On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote: I used to do lots of Asterisk, but got an offer I couldn't refuse, and went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want to set up a test system. One thing I'd really like to get my hands on is recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old source, seems to have been kaput about as long as I've been a sysadmin; are there any other sources out there? (And, yeah, if anyone wants to e-mail them to me directly, I won't say no.) My source was google, and I came across this almost right away: http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html presumably their RPM includes firmware for all of the polycom's I don't use polycom's and never actually downloaded the RPM... but it seems to me thats what you are looking for. You should also be able to contact whomever you bought your Polycom's from to obtain the most recent versions. -- Matt http://www.mattgwatson.ca ___ -- 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] *#%! Polycom...
On Friday 23 May 2008 16:27:49 Ken D'Ambrosio wrote: I used to do lots of Asterisk, but got an offer I couldn't refuse, and went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want to set up a test system. One thing I'd really like to get my hands on is recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old source, seems to have been kaput about as long as I've been a sysadmin; are there any other sources out there? (And, yeah, if anyone wants to e-mail them to me directly, I won't say no.) Polycom no longer requires a reseller agreement to obtain firmware directly from them: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip430.html http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_2_2_2_release_sig.zip This isn't the latest firmware, but it's the latest that most people seem to be running at this time. I think people are still waiting for version 3.0 to stabilize. -- Tilghman ___ -- 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] forwarding pots lines
Thanks for the tip on the 400x, 401x, and 4024 grandstreams. They will work quite nicely (and they'll do t.38 fax if necessary). My application is that I help move business offices from place to place and during the move period. This solution is helpful so the move can be done during the phone cutover period and all their phones still work transparently as usual through the other commotion which ensues. Eric On Fri, May 23, 2008 at 5:37 AM, Joe Carroll [EMAIL PROTECTED] wrote: Two grandsreams, a 4008 and a 4108 would inexpensively do this for you. Instructions are on their site. -Original Message- From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 5/23/08 1:21 AM Subject: Re: [asterisk-users] forwarding pots lines Eric Fort wrote: I'm looking for a simple hardware solution where I can connect POTS lines at one place and make them appear transparently at another location with only SIP and the internet between the locations. If I'm thinking this out right one location would need a box with a bunch of fxo interfaces and the other would need a box with a equal number of fxs interfaces. I'd like this to essentially emulate a really long piece of phone wire in as many ways as possible. What hardware should I use and what is the best way to provision this. I'd prefer to forgo the expense of 2 full asterisk servers as this seems unnecessary for the application. You can use devices called ATAs (Analogue Telephone Adaptors). They are much cheaper. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
[asterisk-users] Strange ring or moh quality
Hi, when i use Dial(SIP/201SIP/202SIP/,60,r) the ring sound is very strange. This happens after an IVR menu. Instead of ing..ing..ing is rrrcrcrcrcrcrcrc..rrccrcrcrcrcrcrcrcrc discontinous and horrible. This happens also with music on hold, when dial multiple channels. I'm sure that is not a bandwith issue, because when dial a single channel or during a conversation, the quality is perfect. How i can obtain the same ring as Ringing() application? - Asterisk 1.4.18 on Debian4 (channels on a Thomson, Astra and X-Lite) -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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 syntax error: need new eyes
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
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Re: [asterisk-users] Strange ring or moh quality
Remove the r. Asterisk will provide the proper ringing by default. If it is not doing so then something is wrong in the config. Compressed codecs (any codec other than ulaw/alaw) do not handle non-voice very well (i.e. ringing, MoH, etc). Vinz486 wrote: Hi, when i use Dial(SIP/201SIP/202SIP/,60,r) the ring sound is very strange. This happens after an IVR menu. Instead of ing..ing..ing is rrrcrcrcrcrcrcrc..rrccrcrcrcrcrcrcrcrc discontinous and horrible. This happens also with music on hold, when dial multiple channels. I'm sure that is not a bandwith issue, because when dial a single channel or during a conversation, the quality is perfect. How i can obtain the same ring as Ringing() application? - Asterisk 1.4.18 on Debian4 (channels on a Thomson, Astra and X-Lite) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Grandstream
Hello, Do you redirected the rtp ports to your phone? usually 1 - 2 defautl rtp ports Best Regards Carlos Rojas On Thu, May 22, 2008 at 8:48 AM, Phibee Network Operation Center [EMAIL PROTECTED] wrote: I have a problem connecting a Grandstream ipphone to an asterisk. The ipphone is behind a nat router, I redirected UDP 5060 and 5004 to my phone. It connects well to the asterisk server. I can call outside and receive calls from outside without any problems. But if I call from this ipphone to another ipphone connected on the same asterisk server, using internal dialing, I can hear my correspondant, but he cannot. Do you have any idea? Thanks for advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan syntax error: need new eyes
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
Thanks for the response.
Tried it this way:
exten =_1NXXNXX,n,Set(CALLERID(num) = ${IF($[ ${CALLERID(num)}
140] ? $
{MAINSTUB}${CALLERID(num)} : ${MAINNUMBER})})
Same result.
sean
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