Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. According to the release plans, support for 1.4 was scheduled to close in April 2011 - basically now. After that, only security patches would be committed. This is already a delay from the original plan published by Russell Bryant. Unfortunately, I think this is way too early. I don't have first-hand experience or an opinion on this matter, but just wanted to comment on how refreshingly welcome it is to have this discussion at all - without Open Source, we'd simply be stuck with a Vista type software (if I believe those who expressed concerns about 1.8). I do have one question: what about the ecosystem? Many people don't use Asterisk by itself, but as part of distributions (PBX in a Flash, Trixbox, ...) and with tools such as FreePBX to configure it. How ready is the ecosystem for moving to a new Asterisk version? -- _ -- 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] wrong time retrieved from system command
Tilghman,
Could you remove your Reply-By header, please? Your deadline is two months in
the past (and in any case, list postings really shouldn't have a reply deadline
at all)
Here is your Reply-By header from your March 21 email:
Reply-By: Wed, 19 Jan 2011 16:20:00 -0600
Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Monday, March 21, 2011 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] wrong time retrieved from system command
On Monday 21 March 2011 06:45:37 asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date
in linux terminal
Mon Mar 21 19:43:35 HKT 2011
I am looking for help.
Do you have an file (or symlink) in /usr/share/zoneinfo called GMT+8? I
certainly don't, and I'm not running anything different from the standard set
of zone files. If you don't have that entry, then the timezone code will use
UTC (i.e. no local differentiations).
--
Tilghman
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Re: [asterisk-users] Two asterisk servers, two different service providers
Here is how I would do it: First, come up with a numbering scheme. For instance, all extensions in location 1 are 1xxx, extensions in location 2 are 2xxx. External calls are 9xxx-xxx- In Location 1: Sip Trunk 1 - goes to Location 2. The dial plan uses it as outgoing trunk for all calls with 2xxx Sip Trunk 2 - goes to Provider 1. The dial plan uses it as outgoing trunk for all calls starting with 9 In Location 2, you do basically the same thing in reverse. Sip Trunk 1 - goes to Location 1. The dial plan uses it as outgoing trunk for all calls with 1xxx Sip Trunk 2 - goes to Provider 2. You can probably also use both providers as backup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell Sent: Wednesday, December 15, 2010 5:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Two asterisk servers, two different service providers All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for the provider at the new location? Can someone point me to some sample conf files that do this? Thanks for all help, AND non smart aleck RTFM answers. Eddie -- _ -- 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] Asterisk on smartphone?
Do you mean, using the smart phone as an Asterisk server, or as a device (i.e., an extension)? I think running Asterisk in server mode would run up against blocking of SIP traffic on most voice networks. Also, you would probably run into issues with battery life, and with availability (what if you are flying and have to turn off your smart phone - suddenly the phone number wouldn't go to the IVR or voice mail any more). I would also be concerned about constantly changing IP addresses, firewalling within the carrier network, ... As an extension, I believe there are some softphone implementations available for Android (they might still run up against blocked SIP traffic, though). Alternatively, you can probably also use follow me or plain old forwarding to get calls to the cell phone, and DNDI for dialing outgoing calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Monday, November 29, 2010 2:09 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk on smartphone? Hello Some SOHO prospects only have a cellphone and I was wondering if someone had investigate running Asterisk on a smartphone, to perform tasks such as IVR, CID rewriting, voice-mail, notifications through e-mails, etc.? Thank you. -- _ -- 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] Someone has hacked into our system
Use IPTables to lock down your machine to only accept incoming connections from your local network and from the particular IPs that you are expecting connections from (such as your SIP trunk, maybe). That is of course assuming that these calls are made by SIP. Don't forget to also change all the passwords. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz Sent: Monday, November 22, 2010 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Someone has hacked into our system Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Thank you, Gary Kuznitz -- _ -- 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] Door Contacts via Asterisk?
Basically, any door control system that works with DTMF tones should work - in theory. You will probably need to play around with the length of the DTMF tones, and maybe also with the level. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Monday, November 15, 2010 10:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Door Contacts via Asterisk? Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -- _ -- 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] Integrating With Asterisk
There are many ways to do this, and very little information to go on. For instance, if you have Exchange 2007 and a lot of money, you can integrate it with Asterisk using Exchange UM (Unified Messaging). Microsoft charges extra for the premium CALs you need to actually do that. There are also some pitfalls (Microsoft uses SIP over TCP, in Asterisk that mode is experimental. Asterisk usually uses SIP over UDP). Since your Windows application already exists, you must already have a way to generate voice mails for non-Asterisk systems. It is entirely possible, and in fact quite likely, that you can leverage whatever mechanism you are using for that. The more information you give us about your existing Windows application and how it interfaces with phone systems to begin with, the better information you will get. Also don't forget version information. Which version of Windows, which version of Asterisk, and is there any other software involved? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shyamala Devi Sent: Monday, November 08, 2010 2:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Integrating With Asterisk Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -- _ -- 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 client floods port 5060 and gets blocked
I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct - if the SIP authentication fails, you can see this happening. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, October 28, 2010 12:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client floods port 5060 and gets blocked Hello, Is there any reason why an IP-phone would pounder on port 5060 ? My firewall blocks the public IP because it thinks the remote IP is port scanning on port 5060. I think the phone is just registering but for some reason it does this repeatedly in a very short time. Oct 28 09:01:48 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48073 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:49 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48074 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:50 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48075 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:52 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48076 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:01:56 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48077 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:00 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48078 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:04 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48079 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:08 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48083 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:12 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48084 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:16 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48085 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Oct 28 09:02:20 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0 OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00 SRC=remote_ip DST=server_ip LEN=696 TOS=0x00 PREC=0x00 TTL=53 ID=48087 DF PROTO=UDP SPT=2367 DPT=5060 LEN=676 Any input on this ?! Kind regards, Jonas. -- _ -- 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] Audio problems on cable modem link
I've had to rip out VoIP in two cable-modem situations because the call quality was too poor. Bandwidth isn't the main characteristic you are looking for; most Internet connections have plenty of that. Latency and jitter matter far more. Latency describes how long each packet travels from your Asterisk system to the other end of the IAX connection. The easiest way to measure it is with ping. Jitter describes how consistent the data transfer speed is. Cable modems are to some extent designed for burst data. This will obviously kill call quality. You can go to http://myvoipspeed.visualware.com/ to get an idea. Run the same test multiple times, both with no traffic on the cable modem, and with a VoIP call going on. Then compare the jitter numbers. What I found was that on my connection (also cable modem) the average jitter was supposedly acceptable, but it was highly variable - in three tests with minimal other traffic all in the middle of the night, the jitter was 1.6, 3.5 and 4.6 milliseconds. 4.6 is in the borderline quality area. And if there is more traffic, quality may well go down further, into the poor-quality zone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, October 15, 2010 7:53 AM To: Asterisk Users List Subject: [asterisk-users] Audio problems on cable modem link We have a small office installation running over a cable modem. (8M down, 500k up confirmed with numerous speed test sites) When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is 175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the internet connection looks fine, etc. We have a linux router in place running wondershaper that seems to be running fine (same as our other installations). Can someone suggest where to look? Could this be the ITSP? -- _ -- 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] Moving from DSL to T1
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Sunday, September 12, 2010 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving from DSL to T1 On Sun, 2010-09-12 at 15:32 -0700, Kevin Keane wrote: In terms of telephony, a T-1 can make a huge difference over DSL. DSL gives you a lot of raw bandwidth, true, but for voice that really doesn't matter all that much. Voice calls only take a relatively small amount of bandwidth anyway; you can fit dozens of concurrent calls into a DSL or T-1. When used strictly for telephony (non-VoIP), a T-1 is designed for 24 concurrent calls, each one takes up 56kbit. For VoIP use, most providers tell you that a phone call takes up about 80kbit/s. What really matters is the latency, and T-1 is a huge improvement over DSL in that area. The easiest way to measure latency is the ping time to a server that is close to you Internet-wise. A DSL has latencies of between 40ms (if it's extremely good and not too many other people are using it) and 1000ms (if there is a problem somewhere). A good T-1 may have latencies as low as 5 ms or so. Also, with a T-1 the bandwidth is guaranteed and bidirectional. With a DSL line, you may get burstable bandwidth - you don't actually have that bandwidth, you just get to compete for excess bandwidth with your neighbors. You consider 40ms extremely good??? Either your isp or youself must have a considerable number of hops to cross. At home (cheap abo) i got following transit delays (round trip) national 15 ms international 17-35 transatlatic or satelite is above 200ms At work national 3-4 ms international 20-25 ms (sorry about the weird quoting - Outlook insists on top-posting!) Wow. I think I have to move to the Netherlands. The European telecom landscape is quite a bit different from US, so I'm not completely surprised. The USA no longer has the fastest Internet in the world anyway. My numbers are from an ATT DSL line in California, suburban San Diego county, and just around the corner from the central office. So it is not the distance (with DSL, the distance does make quite a difference). On the other hand, there are several hops just to get to the Internet backbone. Your work numbers sound like what I have seen with a T-1 here. Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. Please define does not work at all over satellite ??? Sure, it is not studio HIFI quality, but is th same quality as you get from official commercial telco providers. We still have voip over S-band and X-band satelites running NOW between NL and afghanistan. All the people are more than satisfied. ** Should have been more specific. I was talking about Internet over satellite in the USA. I believe those are geostationary TV satellites. I am not familiar with S-band and X-band, but assume they are in lower orbit. That would explain how it can work for you. -- _ -- 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] Moving from DSL to T1
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, September 13, 2010 12:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Moving from DSL to T1 On Mon, 2010-09-13 at 00:32 -0700, Kevin Keane wrote: Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. Please define does not work at all over satellite ??? Sure, it is not studio HIFI quality, but is th same quality as you get from official commercial telco providers. We still have voip over S-band and X-band satelites running NOW between NL and afghanistan. All the people are more than satisfied. ** Should have been more specific. I was talking about Internet over satellite in the USA. I believe those are geostationary TV satellites. I am not familiar with S-band and X-band, but assume they are in lower orbit. That would explain how it can work for you. No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, ** In that case, my guess is that they have a dedicated channel for the voice, maybe even some kind of clocking mechanism. Some T-1 lines here in the USA also have that (one more reason why T-1 works better than DSL/Cable for VoIP). The consumer internet satellite services just mix all kind of Internet traffic, so one packet may have a very low latency while the next one may have a much higher latency, or get lost altogether. Another thing about the consumer satellites is that they are probably optimized for TCP rather than UDP. For TCP, they are using huge retransmission window sizes. That allows large chunks of data to arrive without waiting for confirmation, and the satellite can organize the data into a stream. With UDP, each packet basically stands on its own. Just a guess about another area where these two could be different. -- _ -- 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] Moving from DSL to T1
In terms of telephony, a T-1 can make a huge difference over DSL. DSL gives you a lot of raw bandwidth, true, but for voice that really doesn't matter all that much. Voice calls only take a relatively small amount of bandwidth anyway; you can fit dozens of concurrent calls into a DSL or T-1. When used strictly for telephony (non-VoIP), a T-1 is designed for 24 concurrent calls, each one takes up 56kbit. For VoIP use, most providers tell you that a phone call takes up about 80kbit/s. What really matters is the latency, and T-1 is a huge improvement over DSL in that area. The easiest way to measure latency is the ping time to a server that is close to you Internet-wise. A DSL has latencies of between 40ms (if it's extremely good and not too many other people are using it) and 1000ms (if there is a problem somewhere). A good T-1 may have latencies as low as 5 ms or so. Also, with a T-1 the bandwidth is guaranteed and bidirectional. With a DSL line, you may get burstable bandwidth - you don't actually have that bandwidth, you just get to compete for excess bandwidth with your neighbors. Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. To answer the OP's question: assuming that you will be using the T-1 for mixed VoIP and data (the most likely scenario in this case), a T-1 is really not much different from a DSL line. Both provide you with IP connectivity. Just make sure that QoS is set up correctly on your router and firewall to give priority to VoIP calls. If you are using VoIP and DSL concurrently and your router/firewall supports that configuration, you may also need to modify routing tables to make sure calls go in and out over the correct link. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Sunday, September 12, 2010 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Moving from DSL to T1 On 09/12/2010 02:34 PM, Kyle Kienapfel wrote: Really it depends on what the capabilies of dsl were assuming you are just using both dsl and t1 as internet connections. a dsl that has close to 1mb/sec out and 10mb/sec or so in, is going to be pretty comparable to a t1 actually so not really sure why you would make that switch in the first place. as long as there is a static ip for the server on either, you wouldn't see much difference. (t1 is actually usually delivered over hdsl which is basically the same thing as adsl except the bandwidth is more symetric.) if you have a low speed dsl, such as like 128kb/sec up and 512 down you'll see much faster performance, but again not much big diff if both are just internet connections. This is also assuming your carrier doesn't particularly grossly oversell either service. You need to make sure you are getting transit, not burstable, or quality may suffer depending on how its oversold. On Sun, Sep 12, 2010 at 10:43 AM, Richard Stuppi rich...@stuppi.commailto:rich...@stuppi.com wrote: I work in a small office and have fallen into the role of network support based on knowing enough about networking to be dangerous. Our office is moving from DSL to a T1. Were using Asterisk as our PBX and I'm looking for hints or resources that might help me make the transition as error free as possible. Are there well known gotchas that I shoud be aware of? Thanks in advance, Richard Stuppi rich...@stuppi.commailto:rich...@stuppi.com 626-221-8010 You should be more specific, A)Are you switching from voip over DSL to voip over T1 B) ... or using the T1 for phones? C)Are you switching from analog lines + DSL to just a T1 for voice and data? -- _ -- 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] Moving from DSL to T1
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Sunday, September 12, 2010 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving from DSL to T1 On Sun, 12 Sep 2010, Kevin Keane wrote: What really matters is the latency, and T-1 is a huge improvement over DSL in that area. The easiest way to measure latency is the ping time to a server that is “close to you” Internet-wise. A DSL has latencies of between 40ms (if it’s extremely good and not too many other people are using it) and 1000ms (if there is a problem somewhere). A good T-1 may have latencies as low as 5 ms or so. Also, with a T-1 the bandwidth is guaranteed and bidirectional. With a DSL line, you may get burstable bandwidth – you don’t actually have that bandwidth, you just get to compete for excess bandwidth with your neighbors. You are confusing DSL with cable. Both, actually. The latency numbers are actual numbers measured at a customer site. The sharing of excess bandwidth also happens with both, just in different places. -- _ -- 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] asterisk on Vmware
Are you talking about VMware Server, ESX/ESXi, or one of their other products? The only VMWare product that I can even conceive might work is ESX/ESXi. Others have already pointed out that in VMware, you won't get direct access to the hardware. VMWare does have some limited capability to directly interface with hardware, but I agree with everybody else that it is likely not going to work. The second problem with VMWare in a virtual machine is timing. On a physical machine, Asterisk and Linux have almost complete control over the timing. That's important for voice data streams that need to send data at precise points in time. In a virtual machine, I would expect poorer sound quality due to dropouts. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino Sent: Wednesday, August 11, 2010 1:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk on Vmware Hello, Is it possible to install Asterisk on Vmware(centos) from source. Is there any difference or disadvantage for this compared to asterisk running on physical machine. -- _ -- 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] Monitor asterisk
Do you have a Nagios server? Then you could use that to monitor various aspects of Asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu Sent: Sunday, August 08, 2010 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Monitor asterisk Thanks Nasri, I don't want to only be able to use the CLI because I need the Helpdesk and application support Unit to be able to monitor, and they are not all the techy with CLI and stuff.. On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal na...@ictinnovations.commailto:na...@ictinnovations.com wrote: Hi following asterisk cli commands can help show channels, show uptime and show sysinfo here is an example asterisk -x core show sysinfo On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugmailto:richard.z...@time.co.ug wrote: Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ughttp://www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- 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 -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ughttp://www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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] Using Vertical IP2007 phones with Asterisk?
From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: Sunday, July 25, 2010 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Kevin Keane Subject: Re: [asterisk-users] Using Vertical IP2007 phones with Asterisk? Kevin Keane wrote: I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I would like to use the phones with an Asterisk system instead, but there doesn't seem to be much information on it on Google. Is it even possible? These phones claim that they are SIP phones. Thanks! Kevin Good luck getting much help from Vertical. For those who don't know, Vertical seems to have bought all the semi-crappy business phone system companies in recent years, Comdial, Vodavi (who was previously merged with Isoetec and Executone ) and perhaps others. The only decent product in the mix was the Keyvoice voice mail line. Previous dealers seem to have been shut out of support, and there is little available without being a Vertical dealer. Seems they may be using the Cisco model. Either try and sign up with Vertical, or put the system on eBay and cut your losses John Novack I have done some more research and found that it is indeed possible to use these phones with Asterisk - at least, they register (I haven't fully tested everything yet). I have tried it with Firmware version 3.2.32 - G729. Firmware updates unfortunately are only available through Vertical dealers. In case somebody else uses Google to find it, here is what I found out. The phones, as far as I can tell, do not support DHCP option 66. It does support a configuration file on a TFTP server, but it will only load the configuration file when you tell it to do so in the Web interface. It might be possible to automate that with a wget script. To create the configuration file, it is best to start by manually configuring one phone with a Web browser. After a factory reset, the user name/password are admin/1234. The connection to Asterisk is configured in the SIP tab. The Asterisk server goes into the Registrar Server and Registrar Outbound Server fields. It MUST be an IP address; the phone does not accept a DNS name here. Phone Number and Authorized ID should be the extension. The Phone number will be displayed on the phone's screen. The User name will also be displayed; you can type what you want here, but be aware that you cannot use a space! The secret (Authorized Password) also accepts only certain characters; it is best to stick to just alphanumeric and avoid punctuation altogether. Once you are done, you can save the configuration to a file. In the SW Upgrade tab, click on Download Settings. Move the downloaded file to your TFTP server and rename it IP2007.cfg . Edit as appropriate for the next phone. It is a text file. Be sure to adhere to the same restrictions as the Web interface. Otherwise, the file will not load, and there is no indication what is wrong. Then log on to the next phone's Web interface. Go to the SW Upgrade page. Change the IP address of the TFTP server (no DNS names accepted). Click on Save (otherwise, the phone will use the previous setting for the TFTP server!). Make sure the file name under Profile is correct, and click on the Update button next to it. I hope this helps the next person trying these phones! -- _ -- 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] Fail2ban - SuSEfirewall
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Monday, July 26, 2010 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fail2ban - SuSEfirewall Randy R wrote: On Mon, Jul 26, 2010 at 10:36 AM, Brent A. Torrengali...@torrenga.com wrote: Why isn't the Asterisk box on a static IP on the LAN? That seems to be asking for trouble using DHCP. Not really; the trick is to assign an fixed IP address to the Mac address (with a host statement in ISC DHCP, or a reservation in Windows DHCP). I am a big fan of centralized management, so I prefer to do that rather than have static IP addresses on the network (except of course where absolutely essential). For the OP: maybe a workaround is to assign a fixed IP address from your DHCP server and use a very long lease time? -- _ -- 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] Cisco 7960 phone can't leave a queue
I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I'd appreciate help with tracking down which setting might cause this! 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] Cisco 7960 phone can't leave a queue
Stupid question (sorry, I'm pretty much an Asterisk beginner) - where do I find the dialplan.xml? As far as I can tell, there is no TFTP server in this network. I found the IP address that the phone tries to use for TFTP (192.168.1.7 in this case) but there is nothing at that device. Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue Check your dialplan.xml file that the affected phones are loading. Thanks, --Warren Selby On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause thi s! 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 -- _ -- 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] Cisco 7960 phone can't leave a queue
I'm not sure if ALL use the same TFTP address, but I believe so. My guess is that it is actually the TFTP server that the previous phone vendor used for the phone's initial configuration before shipping it to us. So in that sense it would be an old config. Is there a way to extract the current configuration somehow to regenerate the XML? To make matters worse, I don't have the phone's admin password (it's not the default cisco). Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue It may be pulling a tftp server from dhcp, or it may just have an old config. Do all the phones (even the ones that work properly) use the same tftp address? Thanks, --Warren Selby On Jul 25, 2010, at 4:47 PM, Kevin Keane subscript...@kkeane.com wrote: Stupid question (sorry, I'm pretty much an Asterisk beginner) - where do I find the dialplan.xml? As far as I can tell, there is no TFTP server in this network. I found the IP address that the phone tries to use for TFTP (192.168.1.7 in this case) but there is nothing at that device. Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue Check your dialplan.xml file that the affected phones are loading. Thanks, --Warren Selby On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause thi s! 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 -- _ -- 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 -- _ -- 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
[asterisk-users] Using Vertical IP2007 phones with Asterisk?
I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I would like to use the phones with an Asterisk system instead, but there doesn't seem to be much information on it on Google. Is it even possible? These phones claim that they are SIP phones. Thanks! Kevin -- _ -- 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] Cisco 7960 phone can't leave a queue
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mstults tds.net Sent: Sunday, July 25, 2010 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue On Sun, Jul 25, 2010 at 9:52 AM, Kevin Keane subscript...@kkeane.commailto:subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I'd appreciate help with tracking down which setting might cause this! Thanks! I would also check to see if they are static members. That would explain why some can leave and some can't. Mike Most people indeed are static members, but this problem only affects dynamic members. It is really a dialing issue, not a queue issue (the phone won't let them dial the second *). Thanks for the thought! -- _ -- 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
