Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There is. If you build a reliable network, the phones will simply never have a problem. We've got customers with phones that have never lost contact for years. Re-registering is just a crutch for a network defect. -- Carlos Alvarez TelEvolve 602-889-3003 This is so true! -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
2013/1/31 Ishfaq Malik i...@pack-net.co.uk On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There is. If you build a reliable network, the phones will simply never have a problem. We've got customers with phones that have never lost contact for years. Re-registering is just a crutch for a network defect. -- Carlos Alvarez TelEvolve 602-889-3003 This is so true! If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation
Hi Olivier! Chan_mobile is a old project and work very poor. Search about chan_dongle and get a usb modem it really works. You can't use another machine to connect your hardware. Best regards and sorry for top-posting i'm at blackberry phone. Emiliano. Emiliano Vazquez | PcCentro S.R.L. Office: +54 (11) 4635-7764 ext. 4 Celular: 15.6253.7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -Original Message- From: Olivier oza_4...@yahoo.fr Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 31 Jan 2013 08:25:42 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation
On 31/01/13 07:25, Olivier wrote: Hello, On a LAN, is it possible to install a bluetooth dongle on one workstation (at this time, this workstation OS is not specified) and use it with chan_mobile ? I've read some USB over IP (or Ethernet) middleware exist but I'm not certain I'm looking at the right direction. Regards Hi Oliver, I have used chan_mobile over the years on a number of occasions with several different Nokia phones. I would say that even if in theory it might be able to work with some USB over IP software for the bluetooth dongle, it's probably not worth the hassle in practice. It's quite likely that it will create too many problems, which will probably outweigh the benefits of what you are trying to do. Bluetooth already introduces a certain delay/latency in the communication path - by adding and IP link in between, that will only get worse. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation
2013/1/31 Sebastian Arcus s...@open-t.co.uk On 31/01/13 07:25, Olivier wrote: Hello, On a LAN, is it possible to install a bluetooth dongle on one workstation (at this time, this workstation OS is not specified) and use it with chan_mobile ? I've read some USB over IP (or Ethernet) middleware exist but I'm not certain I'm looking at the right direction. Regards Hi Oliver, I have used chan_mobile over the years on a number of occasions with several different Nokia phones. I would say that even if in theory it might be able to work with some USB over IP software for the bluetooth dongle, it's probably not worth the hassle in practice. It's quite likely that it will create too many problems, which will probably outweigh the benefits of what you are trying to do. Bluetooth already introduces a certain delay/latency in the communication path - by adding and IP link in between, that will only get worse. Sebastian Sebastian, What I had in mind is to use someone's cellphone as a presence detector. Let me explain: - as the first thing you take along when leaving a room or location, is your own cellphone, why not use chan_mobile and a bluetooth dongle on your on PC (as you're not supposed to be within bluetooth range from an asterisk server ;-)) to advertise you're away from your desk - it seems that chan_mobile is not up to expectations for voice delivery but would it remain the same for presence detection, if may call it this way ? Thoughts ? -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 3:08 AM, Leandro Dardini ldard...@gmail.com wrote: 2013/1/31 Ishfaq Malik i...@pack-net.co.uk On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation
On Thu, Jan 31, 2013 at 4:15 AM, Olivier oza_4...@yahoo.fr wrote: What I had in mind is to use someone's cellphone as a presence detector. Let me explain: - as the first thing you take along when leaving a room or location, is your own cellphone, why not use chan_mobile and a bluetooth dongle on your on PC (as you're not supposed to be within bluetooth range from an asterisk server ;-)) to advertise you're away from your desk - it seems that chan_mobile is not up to expectations for voice delivery but would it remain the same for presence detection, if may call it this way ? What you're trying to do would probably be better served by interfacing with AMI and firing an Originate command ( https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate) to trigger your dialplan and update presence state. Or use existing software to accomplish the same thing locally on the PC by updating a user's XMPP status, and have Asterisk subscribe to that. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. You need the register to be able to have good battery life on those. If you use TCP, the softphone will go to sleep, OS will keep the stream alive. When a SIP packet comes in (INVITE, OPTIONS etc), the OS will wake up the softphone and the softphone will handle the packet. No register means no stream and the softphone will just sleep forever. Z -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 9:45 AM, joachim zoach...@securax.org wrote: If you have no NAT or dynamic IP in your network, you can just remove the registration process and assign to each peer its IP address. This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3Com 3101SP phone on Asterisk?
Hello Good Day List I incorporate 3Com 3101SP phone (3C10401SPKRB) to ASTERISK Has anyone managed to connect the phone to Asterisk or if anyone knows how to incorporate these SIP phones Greetings and Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? From what i understood from the original post, Xbrian is looking for a way to work around broken phones that fail to register when they should. I doubt his idea of 100% availability is the same as yours or he would/should be using a different brand/model of phones. + The mobile phone will survive a power outage, because of the register you could be behind NAT as it will open the bindings, you can take it to the bathroom etc. I'm just trying to illustrate the possible advantages of a register before XBbrian redoes his network config. Z -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip unregister ' (where is the peer name) will unregister a peer - however, I want to force registration of a peer from the CLI. Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? A REGISTER request originates from the peer. How do you propose Asterisk ask the unregistered peers to REGISTER in a device agnostic fashion? Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen at? I don't think I've seen anything that has a register command, but lots of devices can get a check your config or reboot command via SIP NOTIFY. I'm more wondering why the peer is unregistered but we still expect to communicate with it. Other than a network problem or the device being unplugged...neither of which could be fixed from the server. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
On Thu, Jan 31, 2013 at 9:26 AM, Adam Moffett adamli...@plexicomm.netwrote: Maybe it's possible to send a NOTIFY to a peer on the last IP it was seen at? I don't think I've seen anything that has a register command, but lots of devices can get a check your config or reboot command via SIP NOTIFY. If you can notify, you can call. This fixes nothing other than refreshing NAT if that's involved. I'm more wondering why the peer is unregistered but we still expect to communicate with it. Other than a network problem or the device being unplugged...neither of which could be fixed from the server. I have a feeling that some people in this discussion have a lack of understanding about the SIP protocol and the underlying networking that could affect it. The original post failed to say whether this was on a LAN without routing, on a LAN with routing, or a WAN. Each of those could result in totally different results and solutions. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Another option would be a VPN between the phone and the LAN the Asterisk box is on. VPN software may handle IP address changes better than the Softphone. This way the IP of the softphone doesn't change. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of joachim Sent: Thursday, January 31, 2013 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip register peer (the quest for near 100% availability) This is the answer. If 100% availability is critical, your IP addresses shouldn't be changing anyway, so take the registration process out entirely. This advice is not valid for android / iphones though. That's absurd. Why would you use a battery-powered smartphone if you are trying to have 100% availability? From what i understood from the original post, Xbrian is looking for a way to work around broken phones that fail to register when they should. I doubt his idea of 100% availability is the same as yours or he would/should be using a different brand/model of phones. + The mobile phone will survive a power outage, because of the register you could be behind NAT as it will open the bindings, you can take it to the bathroom etc. I'm just trying to illustrate the possible advantages of a register before XBbrian redoes his network config. Z -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? For those peers which are at known, fixed, predictable IP addresses (e.g. in-house phones which have statically-configured IP addresses or which get non-dynamic addresses from a DHCP server you control) you do not need to use registration at all. You can simply hard-code the peer's address into sip.conf (or, I presume, an equivalent realtime table). When you Dial() such a peer, Asterisk will start sending out the INVITE packets, regardless of whether it has heard anything at all from that peer in the last hour or fifty. No need for qualify although you can use this to keep track of whether the peers are actually alive or not. If you take this approach, you'll save yourself a great deal of heartburn if you can figure out an automated way of keeping the IP addresses synchronized, between Asterisk and whatever hand out the addresses mechanism the phones use (DHCP, TFTP-based provisioning files, etc.). Keep a master list of peers and addresses in a simple table or file somewhere, and use this to populate the other pieces of software which need to know. For peers which can move around to arbitrary IP addresses, and where your server system won't know what those addresses may be in advance, using REGISTER from the device is really the only good approach. If you've got a setup where devices change their IP address frequently and need to be on-line constantly, I'd say you have a fundamental problem with no easy solution. Using a short registration time limit (e.g. 30 seconds) is probably the least awful way to handle this, and if you're talking about a very large number of phones you may want to set up a dedicated SIP proxy to handle this registration burden and keep Asterisk from having to deal with it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (SOLVED) Call parking in a multi-tenant system
I figured I'd follow up on this in case anyone else cares. The documentation is simply awful and it took a lot of experimenting to make it work. In features.conf for each company: [parkinglot_televolve] parkext = 700 parkpos = 701-720 context = televolve#parking parkinghints = yes parkingtime = 75 courtesytone = beep parkedplay = both findslot = next Include the context declared above in extensions.conf under the dial context. include = televolve#parking In the defaults for the customer's sip.conf file add: parkinglot=parkinglot_televolve NOTE: The parkinglot_ in the name is required! On Tue, Jan 15, 2013 at 2:08 PM, Bakko asannu...@gmail.com wrote: Hello, from 1.6.2 version, Asterisk suport multi-tenant parking Look at features.conf for a example. Regards El 15/01/2013 15:58, Carlos Alvarez escribió: We use Asterisk as a hosted PBX. We've had a couple of requests for parking, but none of the documentation shows any way to make it aware of contexts or otherwise make it multi-tenant. Have I missed something and does anyone know how to make this work? Would be on Asterisk 1.6 for now, 1.8 some time soon. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected DNS queries from Asterisk
Keep in mind this is 1.2... I have a peer in sip.conf: [my-uk900] context = uk900 host= a.b.c.d type= friend Why am I seeing DNS queries like my.example.com and uk900.example.com? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExtenSpy - no audio
Hi, Does anyone have any experience debugging the ExtenSpy function? Asterisk 1.6 (yes, I know it's old) on Debian. core show channels: Channel Location State Application(Data) SIP/570-00031ac1 808@monitor:1Up ExtenSpy(808@desks,w) SIP/node1-00031a number-removed@outbound: Up AppDial((Outgoing Line)) SIP/808-00031aac number-removed@outbound: Up Dial(SIP/0226075066@node1, The problem is that the channel spying (monitoring) has no audio (at all). cheers, Jan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
