[asterisk-users] Asterisk and Hardware Requirements
Hello, I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B. Let's assume : - Asterisk box in country A = GWA - Asterisk box in country B = GWB - Calling party number (located in country A) = CgPNA - Called party number (located in country B) = CdPNB - Second Called party number (located in country B) = sCdPNB - PSTN in country A = PSTNA - PSTN in country B = PSTNB The idea is: The end user in country B will have 2 phone numbers : CdPNB and sCdPNB CgPNA is dialling CdPNB through PSTNA. The call will be intercepted by GWA. GWA will process the call, will change the CdPNB to sCdPNB and direct it via IP to GWB. If the quality of the call via IP is not really good, then I want to redirect it through E1 lines via KPN which will be able to route the call to GWB. GWB will then transmit the call through the PSTNB and will reach sCdPNB. First Scenario -- GWA's requirements would be: - 1 E1 port receiving the calls through PSTNA - 1 E1 port connected to KPN to re-route the calls - Asterisk? being able to map ~5000 CdPNB to sCdPNB - Handling up to ~50 simultaneous calls - Using G711, G.723.1 & G.729.a codec - H323 and SIP compliant GWB's requirements would be: - 1 E1 port receiving the calls routed through KPN from GWA - 5 E1 port connected to PSTNB - Handling up to ~150 simultaneous calls - Using G711, G.723.1 & G.729.a codec - H323 and SIP compliant - Asterisk managing the whole thing Second Scenario --- GWA's requirements would be: - 5 E1 port receiving the calls through PSTNA - 10 E1 port connected to KPN to re-route the calls - Asterisk? being able to map ~5 CdPNB to sCdPNB - Handling up to 500 simultaneous calls - Using G711, G.723.1 & G.729.a codec - H323 and SIP compliant GWB's requirements would be: - 10 E1 port receiving the calls routed through KPN from GWA - 50 E1 port connected to PSTNB - Handling up to ~1500 simultaneous calls - Using G711, G.723.1 & G.729.a codec - H323 and SIP compliant - Asterisk managing the whole thing I was wondering whether Asterisk could handle each scenario and what kind of hardware I would need to support each scenario. I hope I've been clear enough in my explanations. Looking forward for your comments/inputs. Regards, cam ___ --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] Incoming calls
Hi,
I run an asterisk server. The configuration is very basic.
Here is my question :
When someone calls my phone line, which is connected to an FXO card,
asterisk is answering using the context :
; Incoming calls goes to this default context :
[incoming-rtc]
include => postes-sip
;
exten => s,1,Goto(menu,1)
exten => s,2,Hangup
;
exten => menu,1,SetVar(count=0)
exten => menu,2,Answer
exten => menu,3,Background(silence/1)
exten => menu,4,Background(josh/welcome-msg)
exten => menu,5,Background(silence/5)
exten => menu,6,SetVar(count=$[${count} + 1])
exten => menu,7,GotoIf($[${count} < 1]?4) ; Repeat 3 times
exten => menu,8,Goto(s,2)
When a friend calls, I would like for him to enter a 4 digit password
in order to access to a sub-menu, if no password is entered, then the
welcome msg is said ...
Any hints on how to do that ??
Thanks a lot !
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[Asterisk-Users] Asterisk and embedded system
Hi all, I'm kinda new with asterisk stuff. I'm running a Debian with asterisk and a digium X101P clone card in country #1. Since I'm going to work in another country (country #2), I would like to setup another Asterisk server + 1 FXO device in #2 as well as in #1. However I'm looking for a small solution. By "small", I mean I don't wanna have a big desktop running 24/7 ... just an small box like a WRT54GS + ATA Ethernet FXO for ex ... I've read some howtos from http://www.voip-info.org/wiki/view/Asterisk+embedded+systems but I'm still wondering what kind of hardware to choose. The network I would like to setup is basic, something like : X101P + asterisk (#1)asterisk + FXO (#2) Person in #1 will call the line connected to the X101P, then choose via a menu to call me. The call is transferred via IAX between the 2 asterisk boxes, then in #2, asterisk will dial the number via FXO With your experiences, which hardware/system (that can be reliable) do you recommend ? i was thinking of a WRT54GS + a GrandStream Handytone 488 (http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43151799552.htm), what do you think about this ? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is this doable?
I am trying to configure Asterick, having the following system setup on the Asterick server: * eth0 faces the external Internet interface, *but* it does not have IP address (it has a private one given to it by my ISP's DHCP server); * eth1 faces my internal network (say 10.1.1.0/24); * tun0 serves all mobile smartphones and connects to the internal network (it has a different ip range, say 10.1.2.0/24) - they are all connected via the Internet using OpenVPN; I would like to configure Asterick for internal calls between ourselves (eth1<->tun0) and I think I have no problem with configuring this part. I would also like to use one external VOIP provider to which Asterick registers on startup. I think I know how to do that and use the "register" option in sip.conf, though I am not sure for the rest of the NAT-related entries (see below). The purpose of registering this external account is so that both the smart phones (tun0) and the internal net (eth1) users could use this account to make external calls (starting with "0", i.e "_0[0-9]." pattern in extensioins.conf). Obviously, I need these calls to be routed properly via the external VOIP account. In addition to that, I would also need to receive calls from that external account to a nominated internal one (say on extension 20). Is this achievable? If so, I am not completely clear on whether I need to explicitly specify my public IP address (via externip/externhost) or whether Asterick is able to find it without this option? If not, then my plan is to use external program to find it and then use a script in Asterick to set it up as an environment variable. Would that work? That external IP address is going to change, but only in rare circumstances and in such cases I have to restart a lot of stuff (including Asterick) on that server (this is usually triggered by a monitoring program), so it won't be a problem once it is setup initially. I am also not sure whether to specify "nat=yes" or just have "nat=route" only - any ideas? Is there a comprehensive list of all the options available in sip.conf and what they do, because I was unable to find such a list? If the above is doable, I would also like to add the following 2 features: 1. Secondary external VOIP account, though I have no idea how to specify its port in "register" (it uses port 5065 instead of the standard 5060). That account would need to be used on a separate interface (eth2) with a different public IP address. Would it be possible to use externip/externhost inside that external account section to specify it? If this is not possible, then I am thinking of running a separate instance of Asterick with the second VOIP account/public IP address set up - would that work? 2. I would like to be able to configure the following work flow: for a specific set of (external) calling numbers (including where no Caller ID is available): a) these callers to be prompted to specify the "reason" for their call; b) their response to be temporarily "recorded"/stored (a short message of, say no more than 10 seconds long or when they press '#' for that recording to stop); c) Asterick then rings the nominated number for external VOIP calls (extension 20) and play that recorded message back; d) then asks for one of four possible outcomes: - accept this call (pressing, say 1) in which case the call is connected as normal; - reject it with a message that that number/person is "unavailable" (say, by pressing 0); - ask the caller to leave a message by transferring them to a voicemail (say by pressing 2); or - end the initial call completely with a message that the caller/number has been "blacklisted" (say, by pressing the 9 key); Could this be achieved? One final question about binding: in order to be able to use both tun0 and eth1 interfaces so that Asterick serves the calls from both eth1 and tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the tun0 interface - is this possible? Many thanks in advance! -- _ -- 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] Is this doable?
Whats asterick? I blame my spell checker! :-P Do you have anything to offer in terms of help or advice on the issues/questions I posted? -- _ -- 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] Is this doable?
I think you might want to split your questions first. I thought that instead of creating a dozen different threads (and clogging the ML in the process) it would be better to put everything into one place - just pick the issue (or issues) you could address and leave (i.e. delete) the rest out. 1. You can't have multiple externip, but it's not necessary to run two Asterisk instances, because you can set routes to different destinations via particular interfaces. I have no problems with the routing - that is already done. I am not certain how Asterisk handles a stream running across multiple interfaces and how the packet NAT is done. I am also aware that SIP packets "embed" the IP address in it so not sure how this is handled either. -- _ -- 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] Is this doable?
I can't see any reason it shouldn't be.
At this stage, after reading for the past couple of days, my two main
concerns are NAT handling of SIP as both the Asterisk & my clients will
be behind a firewall on a private net, and multitasking - the latter
*may* be solved by going with AGI (not sure yet as Asterisk is still
completely new to me).
I figured out most of the things which had me worried initially,
including how to get multiple "register" entries for external providers
using (non-standard) ports (in v10.0 there is a provision for this in
sip.conf).
If so, I am not completely clear on whether I need to explicitly specify
my public IP address (via externip/externhost) or whether Asterick is
able to find it without this option?
As I understand it, that depends on your router. If you have a Linux
router with the ip_nat_sip module, it'll "fix" your SIP packets so
that you don't need to use the externip setting. However, you'll need
to test to verify that.
Nope! My eth0 interface is not facing the public Internet directly - it
takes its IP address from my ISP's DHCP (which is private!) even though
it can forward/pass traffic through the public internet via that
interface, that is the problem.
Asterisk won't be able to figure out your external address on its own,
so if your firewall isn't fixing packets, then you'd need to specify
externip.
I had a brief look at the sip.conf(.sample) for v10.0 and there is a
provision for activating STUN (application/module) to figure out what my
"real" public address is - if it works, then I may as well go using
this, otherwise I will have to use a separate program to do that job.
http://www.voip-info.org/wiki/view/Asterisk+variables
According to the information here, you should be able to use
${ENV(externip)} to reference the value of an environment variable
named "externip".
Thanks, that was good, but this is better ;-) ->
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
For a SIP trunk... no, I don't. The above link may be useful as it
describes NAT issues with SIP. If you have to specify NAT options at
all, start with "yes" and try "route" if that doesn't work.
Very good find, thanks again!
Is there a comprehensive list of all the options available in sip.conf
and what they do, because I was unable to find such a list?
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
I wish I knew. The link above seems fairly complete, but also terse.
Top find again, thanks! It is a bit dated, but it certainly helps and
I've got a few ideas of my own from this page.
One final question about binding: in order to be able to use both tun0
and eth1 interfaces so that Asterick serves the calls from both eth1 and
tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
tun0 interface - is this possible?
Start with binding to 0.0.0.0.
That was my initial intention as I was hoping Linux will map each
request/response using the appropriate interface (i.e. on which
interface it comes from), I realise binding on 0.0.0.0. is not ideal
from a security point of view (I'd rather issue separate udpbind
statements for the interfaces I want to use), but for now it have to do
if there isn't an alternative.
Many thanks for your input, much appreciated.
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Re: [asterisk-users] Is this doable?
and multitasking - the latter *may* be solved by going with AGI (not sure yet as Asterisk is still completely new to me). I don't really follow you. What I meant was that it may be necessary (at a later stage, once I get enough level of knowledge to be confident of doing something a bit more complex) to ask Asterisk to "multitask" - for example: I listed in one of my previous posts, that once the "essential" part (i.e. connectivity & security) is set up properly, my idea is to set up the proper logic for treating different call groups. One particular example of this, as I already pointed out in that earlier post, is I'd like to ask "anonymous" callers (i.e. callers without any caller IDs or callers placed in that group by myself/admin/other) to be asked by Asterisk what is the reason for their call before routing the call, then some sort of moh to be played while Asterisk - at the same time - rings a nominated number and plays what the callers just said (which should be recorded temporarily, obviously) and I would have 4 options - accept the call, in which case Asterisk transfers the caller to me, reject it with a "not available" message, reject it but allow the caller to leave a message, or reject the call returning a message to the caller that the number is blacklisted. This flow, I am almost certain, would require multitasking on the part of Asterisk - that is what I meant with my previous post. Nope! My eth0 interface is not facing the public Internet directly - it takes its IP address from my ISP's DHCP (which is private!) even though it can forward/pass traffic through the public internet via that interface, that is the problem. In this case, "your" router is the one that your ISP provided or is using, which performs NAT for your hosts. If it is Linux with ip_nat_sip, I believe that it'll "fix" packets without requiring you to configure your Asterisk host. I am not sure that is the case though - especially when I will be using "non-standard" sip port (5061 & 5065) - in my current configuration I use a STUN option on my existing client, but the connection is routed directly - no intermediaries at all, so don't know what Asterisk is going to do in such case, hence my concern. That was my initial intention as I was hoping Linux will map each request/response using the appropriate interface (i.e. on which interface it comes from), I realise binding on 0.0.0.0. is not ideal from a security point of view (I'd rather issue separate udpbind statements for the interfaces I want to use), but for now it have to do if there isn't an alternative. Linux *can* do that, but it requires a bit of configuration for route selection. All the routes are already configured - I have no problem with that bit (I use checkpoint as well as another - customised - module, which takes care of accounting & implements additional tracking). My only worry is from a security point of view - 0.0.0.0 binding is for all interfaces, which is not something I want, but can live with - for now. I am a bit baffled though - Asterisk has existed for quite a while now and I am not sure why this wasn't implemented sooner - everyone knows that using 0.0.0.0 is a security risk. Thank you for your input, as always! -- _ -- 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] Is this doable?
Great subject, BTW. It'll make everyone's contribution so much easier for 'the next guy' to search for. Indeed - one of the main purposes of this ML. It also taps into the (shared pool of) knowledge of others, which is always a good thing. -- _ -- 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] res_http_post.so questions
In short - is this module essential for the running of Asterisk? What is its function? Is there a help/list where I could find a description of what it does? 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] Binding to 0.0.0.0 a security risk?
Why do you see binding to 0.0.0.0 to be a security risk? Purely because a response from Asterisk can be received as a result of a connection on *any* interface on the system/machine. If I have Asterisk confined to, say, 2 interfaces - eth0 (10.1.1.1) and eth1 (10.2.1.1) then a request over a third/subsequent interface cannot be served - it is not normally possible. When Asterisk binds to 0.0.0.0 that is not the case and request over a third/subsequent interface *can* be served by Asterisk (provided the routing is setup properly, that is). If you only have 1 interface, what's the difference? I don't as evident from my initial post. If you have 2 interfaces, just bind to one or the other. I don't - see above. If you have 3 or more interfaces (or you need to just bind to some subset), you should have the skills to configure 'iptables.' I do, but that is not the point - do you rely on microsoft for the security of your own desktop system (if you have one running windows that is) or do you take it into your own hands and make sure it is properly implemented? I don't know about you, but I am firmly in the latter category. Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. What do you mean by that? If a request is received over eht1 are you saying that Asterisk does not respond over the same interface?! -- _ -- 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] Binding to 0.0.0.0 a security risk?
While usually thread hijacking is not something that should be done, in this case thank you for hijacking it as the OP on his original topic was way off topic. Why is that - I think I posted legitimate questions/queries with regards to the installation, configuration and running of Asterisk - how is that "off-topic"? Asterisk can quite nicely deal with NAT provided you set it up right. That said the answer to your question is it doable? yes it is. Next time do lots of hands on and you'll see for yourself. The reason I see this as off topic is because it was mainly routing questions you had which is linux and not asterisk. I disagree - my original questions were with regards to installing, setting up and using Asterisk in a multi-interface environment (NAT, as well as restricting Asterisk to which interface it needs to bind to is part of that process as far as I know). My questions are not how to do general NAT - I am perfectly capable of implementing that, having over 18 years experience with that sort of thing, thank you. My questions were more on how/whether Asterisk deals (or whether is capable of dealing) with NAT and all the other issues I raised in my initial post. One last thing though - drop the attitude - if you are not willing/capable of contributing anything to this thread just move along - there is nothing to see here. Everyone knows? Not me. From Steves post I understand that neither does he know. Do you mind explaining this? See my previous post. -- _ -- 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] Is this doable?
Your description sounds almost entirely like the existing call screening, so I'm pretty sure you'll be able to accomplish it. Start with call screening, and modify that to suit your needs. It is indeed. This is already implemented in Asterisk I take it then? If so, brilliant news! I'd encourage you not to give callers much information. If you tell callers that their number is blacklisted, or that the recipient is not available (and not offer them voicemail), they're likely to call back and provide different or no information. It'll be more effective to let them leave voicemail and then delete and ignore it. Just a suggestion. A good one, thanks for that - will take it on board. IP routing alone isn't actually sufficient (typically) to use multiple interfaces. Under Linux, you have to set up multiple routing tables, track connections, mangle reply packets (mark), and use 'ip rule' to select the proper routing table for the packet. If you haven't verified that replies go out the right interface, you should look. If you have, then ignore me. :) This is already done and works, though from my (admittedly limited) understanding of the sip protocol I know that internal IP address information is included in the actual packet. I know that I could use sip helpers (kernel modules), but just wanted to know whether I should rely on Asterisk to do this or whether I should do it via netfilter alone (in which case why are all the nat-related options present in Asterisk?). No... binding to 0.0.0.0 isn't a security risk. Typically applications bind to a specific address so that a single host can have multiple addresses, and an application or multiple applications can bind to specific addresses to implement virtual hosting. I disagree. Binding to 0.0.0.0 allows connections to be made from all interfaces (provided the routing allows it, of course) - see my previous post as I do not wish to repeat myself here. I do not wish to solely rely on iptables/netfilter/other means if I can constrain Asterisk to the interfaces it is supposed to be using. -- _ -- 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] Binding to 0.0.0.0 a security risk?
As far as I know, Asterisk would use the default Linux/Unix routing algorithms to send packets out, in which case yes: responses may not go out on the same interface packets were received on. E.g. if you receive packets with non-LAN IP addresses on eth0, while your default route is set to eth1, in the absence of custom routing Linux will send the responses over eth1. Thanks, another mystery solved then - Asterisk does rely on the Linux/Unix routing, in which case I would definitely need to take care of the SNAT/DNAT and proper routing/forwarding of packets between interfaces using core Linux/Unix tools. Am I correct in thinking that? -- _ -- 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] res_http_post.so questions
The primary goal was to upload audio for IVRs in the Asterisk GUI. Thanks, if I don't use the GUI is it safe to exclude it from the build (it is just that I want to avoid a bunch of other dependencies which come with that module)? -- _ -- 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] Is this doable?
It is indeed. This is already implemented in Asterisk I take it then? If so, brilliant news! More or less. I don't know if it's easy to trigger for specific caller ID values, or for none. You might need to to a little customization, but something mostly like what you describe is present. I am glad to see this! Which modules/functions present this functionality - do you know? I am almost certainly going to customise this as the screening of calls will be done using my own custom-defined criteria and the response options will also have to be customised/enhanced as well (how much really depends on what is currently implemented in Asterisk). Is there some kind of attack that you believe is possible on one interface that isn't on the other? I can't conceive of any way that making your service available on additional addresses increases your vulnerability. Of course it does - by making Asterisk service available on, say eth2 (by binding on 0.0.0.0 that is automatically enabled, i.e. Asterisk can receive packets coming from that interface). This is not what I want. If I could restrict Asterisk to bind only on the eth0 and eth1 for example, packets coming from that interface (eth2) won't affect Asterisk at all and they will either be dropped or rejected as nothing would listen on that address/port. I know that you may say "netfilter/iptables is there to protect you", but the system will be more secure if Asterisk don't have the (physical) ability to answer requests coming from "undesired" interfaces - regardless of whether I have a fully-functional netfilter/iptables in place (even if it is compromised), rather than having Asterisk potentially answering such requests (by binding to 0.0.0.0) even if netfilter/iptables are functioning. In other words, having physically restricted Asterisk from answering requests coming from undesired interfaces (short of directly forwarding/routing packets from/to that interface) is better than allowing it do so and relying solely on netfilter/iptables for protection. -- _ -- 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] Binding to 0.0.0.0 a security risk?
All of that is true, but none of it appears to be a security concern, specifically. For you, may be, but from where I am sitting, I don't want to rely solely on netfilter/iptables to protect me when I could physically restrict Asterisk from binding to that interface (and answering such requests) - that will serve me well in the event netfilter/iptables is somehow compromised (see my previous post). It's possible for an application to bind a socket to a specific interface, but very few do. Generally speaking, server applications bind a socket to an address. The kernel decides what interface that packets are sent on. Normally that will be the interface that has the lowest cost default route, not necessarily the one on which a connection was initiated. That is why I noted previously that you have to use connection tracking, packet mangling, and ip rules for multi-homed hosts. If you've never verified that your packets are being routed out the interface you expect (probably with tcpdump), perhaps you should. Yeah, that was already clarified by another poster - I assumed (wrongly, as it turned out) that Asterisk, somehow, could "automagically" take care of directing sip/voip packets between interfaces and also take care of all the other related issues. As I understand it now, I will have to reconfigure this myself by using the standard Linux/Unix tools (ip & iptables mostly). Thanks for the clarification yet again! -- _ -- 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] Is this doable?
http://www.asterisk.org/astdocs/node66.html
Thanks, never knew that!
Yes, I understand that it's not what you want, but that doesn't make
it a security concern. If Asterisk is publicly available on one
interface, making it available on another interface doesn't make you
less secure.
You lost me. What I want/don't want is largely irrelevant. The issue is,
as you rightly pointed out, whether it is considered more secure or less
secure when Asterisk binds to 0.0.0.0 as oppose to using a specific set
of interfaces, selected at startup.
If one has internal networks, accessible via, say eth1 and tun0, and
implements Asterisk to act as the internal/private PBX (without exposing
it to the outside world), then having been forced to use 0.0.0.0 will,
of course, expose Asterisk to any other - undesirable - interfaces,
including those pointing to the outside world.
By having the option to specify which interfaces Asterisk should use to
bind to (via multiple {udp,tcp}bind statements or by any other means)
Asterisk is *not* exposed to any undesirable interfaces and thus, the
risk is not there. I thought I have made that clear by now, obviously I
haven't, it seems.
It's fine if you want to take that step, but please drop the "everyone
knows this is a security risk" thing. You appear to be alone in that
opinion, and unable to explain why you think it's a security risk.
Moreover, you're speaking for others without warrant or welcome.
If you can't see why binding to 0.0.0.0 carries greater risk than
restricting Asterisk which interfaces to use, then you are truly blind
and beyond help, I am afraid.
--
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Re: [asterisk-users] Is this doable?
I don't get this. Didnt EVERYONE know it's insecure? Can you read? -- _ -- 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] PBX / Asterisk integration
Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so VoIP users can talk to the PSTN as well. We don't actually control the University's PBX, so it is highly desirable that any changes to the PBX are very minimal! We were thinking of setting up a single extension which is associated with a huntgroup on the PBX that somehow connects (via an E1?) to the Asterisk box. If a user leaves his office he sets his extension to re-direct calls to that extension number (or to re-direct on no-reply). The PBX would route the call to the asterisk huntgroup. Asterisk would "see" the original extension number called (assuming this is possible!!), and (knowing which extension maps to which user to which IP address) route the call to the user over the IP network. I can handle the IP stuff without any problems. My question is: is this a possible/sensible approach to implementing this type of service? If not, what's a better solution? TIA for any suggestions/comments, josh. -- --- Josh Howlett, Networking & Digital Communications, Information Systems & Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960s
Cisco and bugtraq need to know this! josh. On Fri, 2003-07-11 at 09:21, Matthew Hardeman wrote: > Cisco should really be ashamed of this product... > > While it is physically well constructed, and has excellent sound > quality along with a very pleasant user interface, the device has > SERIOUS stability issues, unless you run your network with an iron > fist... > > Quite by accident, while configuring my Asterisk system to connect to > a Cisco 7960 via SIP in a standard office PBX type arrangement, I > discovered something interesting... > > By screwing around with both the source IP address of a SIP message, > along with certain IP addresses in the SIP message itself, it's quite > easy to crash the Cisco. > > In short, it would be trivial to DOS (by forcing continuous crashes > and the subsequent reboots) any Cisco 7960 that you can route UDP > packets to... > > Matt Hardeman > PaperSoft > > -- --- Josh Howlett, Networking & Digital Communications, Information Systems & Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 403 FORBIDDEN Help!
I'm new too, but alot of my 403 forbidden messages when adding extensions were due to context rules.. make sure that the client dialing the extension is included in the same context your extension is in. just my thoughts on it, as it resolved a lot of 403 errors for me. - Original Message - From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:31 AM Subject: [Asterisk-Users] 403 FORBIDDEN Help! Hello, I have a question. I set up an extension to 1234 exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank you in advance. Bartosz Jozwiak
Re: [Asterisk-Users] 403 FORBIDDEN Help!
is the sip extension on the vocal sip server also 1234? if not, that could be why it's not working... when you're dialing sip, you have to use the format: exten => LOCEXT,1,Dial(SIP/[EMAIL PROTECTED]:port) so it would be something like exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060) where REMEXT is the sip extension you're trying to dial. pardon me with the context stuff, i just woke up recently, and didn't think to ask if the remote extension was the same. - Original Message - From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:50 AM Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help! Asterix PBX is loggin to Vocal and the extension number is also loggin on the same vocal server. I cannot make it work :( - Original Message - From: Josh Roberson To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 11:43 AM Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help! I'm new too, but alot of my 403 forbidden messages when adding extensions were due to context rules.. make sure that the client dialing the extension is included in the same context your extension is in. just my thoughts on it, as it resolved a lot of 403 errors for me. - Original Message - From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:31 AM Subject: [Asterisk-Users] 403 FORBIDDEN Help! Hello, I have a question. I set up an extension to 1234 exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank you in advance. Bartosz Jozwiak
[Asterisk-Users] Brooktrout PRI-ISA48 card... info..
I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1 card, which, upon checking with brooktrout, is supported for linux 2.x, but before I do this, I want to check and see what the opinions of your, the list members, and Mark, of course, as far as asterisk being able to use this card. ANY information would be helpful, as this offer will expire to me very soon. Thank you. :)
Re: [Asterisk-Users] Brooktrout PRI-ISA48 card... info..
It did I think, however I do still have an ISA slot to use My question was, will it work with asterisk? - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 19, 2003 10:44 AM Subject: Re: [Asterisk-Users] Brooktrout PRI-ISA48 card... info.. > On Tue, 2003-08-19 at 04:08, Josh Roberson wrote: > > I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1 > > card, which, upon checking with brooktrout, is supported for linux > > 2.x, but before I do this, I want to check and see what the opinions > > of your, the list members, and Mark, of course, as far as asterisk > > being able to use this card. > > > > ANY information would be helpful, as this offer will expire to me very > > soon. > > It is ISA. Didn't the ISA bus go away with PC99 specs? It is getting > extremely rare to find ISA motherboards now days. Don't waste your time > on it. If you consider your time with more than minimum wage then the > time spent making it work will be more than buying a digium card. > > > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic cards...
Are the dialogic DTI series cards supported in asterisk? I know there's standard API, but I don't know if it supports only the cards listed on the digium site, or if it will support ALL dialogic cards.. Sorry, I *AM* a newbie to this stuff, just trying to get my hands on a good card. Thanks.
[Asterisk-Users] OT: My congestion music.
Wanna cheap laugh? IAXTel: 17005334094. -Josh
[Asterisk-Users] Newbie IVR question
let me first say this is an amazing product. ok here is my question what I want to do is be able to have people call me and answar questions. The answars to there questions would need to be stored in a mysql database. so call comes in asterisks plays question asterisk waits for answar caller presses 3 into a sql table goes (callerid,questionnum,3) on to next question. Can this be done in asterisks, if so, can anyone point me in the correct direction? howto's code examples etc Help protect your PC: Click here to go to the Mcafee.com free online virus scan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
Are there any examples for ther psql or agi scriptscan I use php withagi Thanks Josh >From: Steven Critchfield <[EMAIL PROTECTED]> >Reply-To: [EMAIL PROTECTED] >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Newbie IVR question >Date: 31 Aug 2003 15:54:53 -0500 > >On Sun, 2003-08-31 at 15:39, Josh Edwards wrote: > > let me first say this is an amazing product. > > > > > > ok here is my question > > > > what I want to do is be able to have people call me and answar > > questions. The answars to there questions would need to be stored in a > > mysql database. > > > > so > > > > call comes in > > > > asterisks plays question > > > > asterisk waits for answar > > > > caller presses > > 3 > > > > into a sql table goes (callerid,questionnum,3) > > > > on to next question. > > > > > > Can this be done in asterisks, if so, can anyone point me in the > > correct direction? howto's code examples etc > > >If you are stuck on mysql, then you will need to do this in agi. If you >could do it in postgres, then there is the PSQL commands. >-- >Steven Critchfield <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users Get MSN 8 and enjoy automatic e-mail virus protection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
So I guess what i need is an agi app that goes is passed callerid and ext then does a "wait for digit" then writes (callerid,ext,) to a database. I do not know perl, so I guess my real question is, is this hard to write, has anyone else done it? Josh >From: Steven Critchfield <[EMAIL PROTECTED]> >Reply-To: [EMAIL PROTECTED] >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Newbie IVR question >Date: 31 Aug 2003 15:54:53 -0500 > >On Sun, 2003-08-31 at 15:39, Josh Edwards wrote: > > let me first say this is an amazing product. > > > > > > ok here is my question > > > > what I want to do is be able to have people call me and answar > > questions. The answars to there questions would need to be stored in a > > mysql database. > > > > so > > > > call comes in > > > > asterisks plays question > > > > asterisk waits for answar > > > > caller presses > > 3 > > > > into a sql table goes (callerid,questionnum,3) > > > > on to next question. > > > > > > Can this be done in asterisks, if so, can anyone point me in the > > correct direction? howto's code examples etc > > >If you are stuck on mysql, then you will need to do this in agi. If you >could do it in postgres, then there is the PSQL commands. >-- >Steven Critchfield <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users Enter for your chance to IM with Bon Jovi, Seal, Bow Wow, or Mary J Blige using MSN Messenger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
I downlaoded it and tried it, SIPPS. Nice featureful sip client, however, I haven't been able to get it to pass dtmf to *. I don't know if this is a software restriction or not, but I have emailed nero asking them for their opinion of this, as it is, in my case, a LARGE restriction when trying to deal with IVR's, and esp. * voicemail. - Original Message - From: "Dave Packham" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]> Sent: Monday, September 01, 2003 10:23 PM Subject: [Asterisk-Users] Sip Software from Nero Folk? > http://www.nero.com/us/631911127302064.html > > > Have you all seen this? > > Its a SIP softphone put out by the people that do the CD burning software Nero... > > Check it out it works with * > > Dave > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Actually, I do have that. i've tried inband, as well as rfc2883. Neither work. I'm going back and forth with ahead software on the issue, and they're doing a little bit of looking into it. Doesn't even work when clicking on the numbers, as required by the software, as someone else pointed out, that was an obvious "feature" i noticed right off the bat. -Josh - Original Message - From: "Gavin Hollinger" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, September 02, 2003 1:58 AM Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? > > haven't been able to get it to pass dtmf to *. I don't know if this > > Do you have > dtmfmode=inband > in sip.conf? > > http://www.sippstar.com/en/631927444894185.html > > Q.: DTMF generated by SIPPS is not recognized by other > applications. > > SIPPS generates DTMF based on the standard set-op for DTMF for PSTN > telephones. SIPPS transmits DTMF as tones and not as events. Hence, any > application awaiting an event instead of a tone will not be able to work > with SIPPS > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions.conf issue
Question below, here is the file in question exten => 9,1,system,/usr/local/bin/hetest 01 onexten => 9,2,system,/usr/local/bin/hetest 02 onexten => 9,3,system,/usr/local/bin/hetest 03 onexten => 9,4,system,/usr/local/bin/hetest 04 onexten => 9,5,system,/usr/local/bin/hetest 05 onexten => 9,6,system,/usr/local/bin/hetest 06 onexten => 9,7,system,/usr/local/bin/hetest 07 onexten => 9,8,system,/usr/local/bin/hetest 08 onexten => 9,9,system,/usr/local/bin/hetest 09 on When I dial 9 it runs the first item, then it exits and gives me a busy, why does it not go through all of the items then exit Josh Get MSN 8 and help protect your children with advanced parental controls. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf issue
Nothing, it is a shell script that runs another program. do I need to have it return something? Josh >From: Steven Critchfield <[EMAIL PROTECTED]> >Reply-To: [EMAIL PROTECTED] >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] extensions.conf issue >Date: Tue, 02 Sep 2003 15:11:34 -0500 > >On Tue, 2003-09-02 at 14:54, Josh Edwards wrote: > > Question below, here is the file in question > > exten => 9,1,system,/usr/local/bin/hetest 01 on > > exten => 9,2,system,/usr/local/bin/hetest 02 on > > exten => 9,3,system,/usr/local/bin/hetest 03 on > > exten => 9,4,system,/usr/local/bin/hetest 04 on > > exten => 9,5,system,/usr/local/bin/hetest 05 on > > exten => 9,6,system,/usr/local/bin/hetest 06 on > > exten => 9,7,system,/usr/local/bin/hetest 07 on > > exten => 9,8,system,/usr/local/bin/hetest 08 on > > exten => 9,9,system,/usr/local/bin/hetest 09 on > > > > > > When I dial 9 it runs the first item, then it exits and gives me a > > busy, why does it not go through all of the items then exit > >what does hetest return? >-- >Steven Critchfield <[EMAIL PROTECTED]> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users Get MSN 8 and enjoy automatic e-mail virus protection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering "answer"
Hello Mickey,
I had a similar problem with the mp3 functions a while back, but I
handled it off list, but since you're having the same issue, here's how I
noted to fix it:
1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work,
and it has to be the REAL mpg123.
2. Make sure that the system has already passed the Answer call for the
extension. For example:
exten => 69,1,Wait(5)
exten => 69,2,Answer
exten => 69,3,MP3Player,/path/to/music.mp3
This example is the only way I found to make the mp3 player work. I haven't
been able to test fully the music on hold functionality, as my system is'nt
fully functional yet, and I don't have other clients to test with.
-Josh
- Original Message -
From: "Mickey Binder" <[EMAIL PROTECTED]>
To: "Asterisk maillist (E-mail)" <[EMAIL PROTECTED]>
Sent: Wednesday, September 03, 2003 11:13 AM
Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering "answer"
> Hi
>
> I have kind of an odd problem.
> When dialing in from an outside line via a TE410P card it seems like
> MusicOnHold and MP3Player doesn't work properly (for me anyway). The
remote
> end which is calling * doesn't hear the music but just keeps ringing. But
if
> I insert a Playback("file_which_dont_exist") just before the Moh or
> MP3Player I can hear the music. Actually I observed the same behavior
> internally when I used H323 for my Welltech Wellgates (which I have now
> changed to SIP).
>
> What can cause this kind of problem?
> Its not a huge issue since I can use the Playback to trigger the call, but
> it would be nice to find the source of the problem.
>
> regards
> Mickey Binder
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering "answer"
Strange.. I had a symbolic link, and it wouldn't work. After I finally got
it working properly, i even tried to remove it from /usr/bin and symlink it,
and it wouldn't work again... couldn't for the life of me figure out why.
- Original Message -
From: "Joseph Finley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, September 03, 2003 4:20 PM
Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering
"answer"
>
> I used a symbolic link and it works just fine for me.
>
> -Joe
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson
> Sent: Wednesday, September 03, 2003 4:30 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering
> "answer"
>
>
> Hello Mickey,
>
>I had a similar problem with the mp3 functions a while back, but I
> handled it off list, but since you're having the same issue, here's how I
> noted to fix it:
>
> 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work,
> and it has to be the REAL mpg123.
>
> 2. Make sure that the system has already passed the Answer call for the
> extension. For example:
>
> exten => 69,1,Wait(5)
> exten => 69,2,Answer
> exten => 69,3,MP3Player,/path/to/music.mp3
>
> This example is the only way I found to make the mp3 player work. I
haven't
> been able to test fully the music on hold functionality, as my system
is'nt
> fully functional yet, and I don't have other clients to test with.
>
> -Josh
>
> - Original Message -
> From: "Mickey Binder" <[EMAIL PROTECTED]>
> To: "Asterisk maillist (E-mail)" <[EMAIL PROTECTED]>
> Sent: Wednesday, September 03, 2003 11:13 AM
> Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering
"answer"
>
>
> > Hi
> >
> > I have kind of an odd problem.
> > When dialing in from an outside line via a TE410P card it seems like
> > MusicOnHold and MP3Player doesn't work properly (for me anyway). The
> remote
> > end which is calling * doesn't hear the music but just keeps ringing.
> > But
> if
> > I insert a Playback("file_which_dont_exist") just before the Moh or
> > MP3Player I can hear the music. Actually I observed the same behavior
> > internally when I used H323 for my Welltech Wellgates (which I have
> > now changed to SIP).
> >
> > What can cause this kind of problem?
> > Its not a huge issue since I can use the Playback to trigger the call,
> > but it would be nice to find the source of the problem.
> >
> > regards
> > Mickey Binder
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering "answer"
Along the same lines, though, I do agree that MP3Player app should cause
the system to trigger the answer without having to do it manually, but I
could see where you might not want it to, as well. In theory, the Playback
app (pardon, this is what i've gathered by toying with it) triggers the
'answer' function if the call is not already answered. Couldn't we get the
MP3Player app to do the same? I'm not that skilled of a programmer,
otherwise, I'd hack it up and do it myself.
-Josh
- Original Message -
From: "Mickey Binder" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, September 04, 2003 4:34 AM
Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering
"answer"
> > -Original Message-
> > From: Joseph Finley [mailto:[EMAIL PROTECTED]
> > Sent: 3. september 2003 23:21
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering
> > "answer"
> >
> >
> >
> > I used a symbolic link and it works just fine for me.
> >
> > -Joe
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Josh Roberson
> > Sent: Wednesday, September 03, 2003 4:30 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering
> > "answer"
> >
> >
> > Hello Mickey,
> >
> >I had a similar problem with the mp3 functions a while back, but I
> > handled it off list, but since you're having the same issue,
> > here's how I
> > noted to fix it:
> >
> > 1. Make sure you have mpg123 in /usr/bin. Symbolic links
> > will NOT work,
> > and it has to be the REAL mpg123.
> >
> > 2. Make sure that the system has already passed the Answer
> > call for the
> > extension. For example:
> >
> > exten => 69,1,Wait(5)
> > exten => 69,2,Answer
> > exten => 69,3,MP3Player,/path/to/music.mp3
> >
> > This example is the only way I found to make the mp3 player
> > work. I haven't
> > been able to test fully the music on hold functionality, as
> > my system is'nt
> > fully functional yet, and I don't have other clients to test with.
> >
> > -Josh
> Ok I get same results when using Answer, so I'll just stick with that
>
> thx
> Mickey
> >
> > - Original Message -
> > From: "Mickey Binder" <[EMAIL PROTECTED]>
> > To: "Asterisk maillist (E-mail)" <[EMAIL PROTECTED]>
> > Sent: Wednesday, September 03, 2003 11:13 AM
> > Subject: [Asterisk-Users] MusicOnHold and MP3Player not
> > triggering "answer"
> >
> >
> > > Hi
> > >
> > > I have kind of an odd problem.
> > > When dialing in from an outside line via a TE410P card it
> > seems like
> > > MusicOnHold and MP3Player doesn't work properly (for me anyway). The
> > remote
> > > end which is calling * doesn't hear the music but just
> > keeps ringing.
> > > But
> > if
> > > I insert a Playback("file_which_dont_exist") just before the Moh or
> > > MP3Player I can hear the music. Actually I observed the
> > same behavior
> > > internally when I used H323 for my Welltech Wellgates (which I have
> > > now changed to SIP).
> > >
> > > What can cause this kind of problem?
> > > Its not a huge issue since I can use the Playback to
> > trigger the call,
> > > but it would be nice to find the source of the problem.
> > >
> > > regards
> > > Mickey Binder
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
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>
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[Asterisk-Users] Call Forwarding
I am trying to setup call forwarding, I have 2 x100p cards.I want the calls forwarded to a static number. Is this done using meetme? Any help or HOWTOs would be great Get 10MB of e-mail storage! Sign up for Hotmail Extra Storage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Indications
Is there anyway in asterisk using the tdmcard, to make the dial tone "sound funny" when you have voicemail. The local bell does this here, and I was thinking that mabye asterisk did it Use custom emotions -- try MSN Messenger 6.0! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelized T1 Question/Request
Using a channelized T1 here for our * box, we get ANI and DNIS information inband over DTMF, in the format *1234567890*222333*, where 1234567890 is the ANI and 222333 is the DNIS. Any hope for processing this effectively without resorting to AGI scripting? Right now, * gets confused and processes as an invalid extension. -- Josh Rollyson Technical Support - [EMAIL PROTECTED] Cape Lookout Internet Services 1-800-262-8371 / 1-252-240-2335 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channelized T1 Question/Request
Steven Critchfield wrote:
Pattern matching.
exten = _*NN*NN*,1,SetVar(ANI=${EXTEN:2:10})
exten = _*NN*NN*,2,SetVar(DNIS=${EXTEN:13:10)
exten = _*NN*NN*,3,Goto(DifferentContext|${DNIS}|1)
Thanks, this looks like it should work!
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RE: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
This may just be me, but When replying to a message from a digest, it would be proper to remove all the context except that to which you are replying so as not to have to scroll an entire mile to see your reply. I know if I was the person you were replying to, I probably wouldn't scroll all the way through the other 15 messages just to see a reply. Just my .02, Sorry if I seem a bit irrational, just irritated. -Josh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shimul Kanti Barua Sent: Wednesday, September 17, 2003 4:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs <<<<>>>>>> > Message: 16 > Date: Sat, 13 Sep 2003 16:32:32 +0300 > From: Michael Manousos <[EMAIL PROTECTED]> > Organization: inAccess Networks > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway > Reply-To: [EMAIL PROTECTED] > > Cerrajetto wrote: > > Hello: > > > > I am testing Asterisk with oh323. > > > > My question is: can Asterisk route some calls thru a second h323 gateway (a > > h323 <-> PSTN gw)? > > > > - Asterisk ip: 192.168.1.10 > > - h323<->PSTN gw: 192.168.1.20 > > > > I've tried: > > > > exten => _9,1,Dial(OH323/192.1.1.20) > > > > or > > > > exten => _9,1,Dial(OH323/[EMAIL PROTECTED]) > > I guess that "192.1.1.20" is a typo, right? > You will have to give more info in order to be able to > find the problem. > Try to set these params in oh323.conf file: > > wrapLibTraceLevel=3 > libTraceLevel=3 > libTraceFile=/tmp/trace.txt > > Rerun and send me the "/tmp/trace.txt" file, "oh323.conf" > and the screen log (off-list). > > > > > but it does not work at all. > > > > If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls are > > routed to the PSTN perfectly. > > > > What is the correct way to route some calls from Asterisk to another h323 > > gateway? > > > > Thank you, > > Mark > > > > > Michael. > Hi Mark, Yes, it is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream exten => _01XX,1,Dial(OH323/[EMAIL PROTECTED]) ; Crilium - exten => _9XX,1,Dial(OH323/[EMAIL PROTECTED]) Shimul > > > > --__--__-- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxComm - IAX client for Win32
The copy I downloaded from the website never did register with *. It would make authenticated calls, but wouldn't actually register with the server. Even checked the IAX peers, and nope, wasn't registered. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar Sent: Wednesday, September 17, 2003 1:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 On Wed, 17 Sep 2003 11:27:25 +0200, Florian Overkamp <[EMAIL PROTECTED]> wrote: >At 19:55 16-9-2003 -0500, you wrote: >>iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port >>of the >>iax library. >> >>iaxComm is a client written in c++ using wxWindows. There is a Win32 >>binary on >>the site. I think that it should be compilable on Linux and MacOSX, but can't >>test it. >> >>Feedback is welcome. > >Well, this looks like a big improvement, but I cant seem to find the option >to register at the asterisk server. Is it impossible, or am I missing it ? >Would be a hefty requirement for real use, I think... It automatically registers with all asterisk servers that have been configured in the Options|Directory dialog. I dial out and register from two different servers. I previously had an "auto register" checkbox, but changed to registering all servers when I moved the servers list from a listcontrol to a combobox. I'm thinking that you would want to register with any server through which you may want to make outbound calls. When the servers are read from the registry, they are read in alphabetical order, and registration is attempted in that order. (The order may be different on other platforms). You should see "Registration accepted" in the status bar after the last server is registered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programming 976 numbers from dialing out.
Or you can set them up correctly, as patterns. Add a leading _ to the extension. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, September 17, 2003 2:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming 976 numbers from dialing out. Just as simple to call your telco and have those turned off then its not an issue ever! bkw On Wed, 17 Sep 2003, Ariel Batista wrote: > I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! > > exten => 1900XXX,1,Congestion > exten => XXX976,1,Congestion > exten => XXX976,1,Congestion > exten => 1XXX976,1,Congestion > exten => 91900XXX,1,Congestion > exten => 9XXX976,1,Congestion > exten => 91XXX976,1,Congestion > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxtel and iax.conf
Bkw, Yes, and we have documented a bug on it... seems to only happen with the iaxtel entry, so we're not sure if it's IAXTel that's at fault, or a bug with * causing iaxtel to read the wrong entry. http://bugs.digium.com/bug_view_page.php?bug_id=296 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, September 23, 2003 12:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iaxtel and iax.conf I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has anyone else seen this happen? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
FW: RE: [Asterisk-Users] iaxtel and iax.conf (HTML CONTENT, FYI)
I hate to post HTML to the list, but I refuse to respond to this, and I would like to say that whomever is using this service is kinda stupid for subscribing an email address to the list using this service. I hope they learn a lesson by us, the list users, NOT responding to this, and eventually, after not receiving any list mail, they’ll wonder “hmm… why is this list so dead?” Just my .02. Again, sorry for the html post. -Original Message- From: AntiSpam UOL [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 3:27 AM To: [EMAIL PROTECTED] Subject: RE:RE: [Asterisk-Users] iaxtel and iax.conf Olá, Você enviou uma mensagem para [EMAIL PROTECTED] Para que sua mensagem seja encaminhada, por favor, clique aqui Esta confirmação é necessária porque [EMAIL PROTECTED] usa o Antispam UOL, um programa que elimina mensagens enviadas por robôs, como pornografia, propaganda e correntes. As próximas mensagens enviadas para [EMAIL PROTECTED] não precisarão ser confirmadas*. *Caso você receba outro pedido de confirmação, por favor, peça para [EMAIL PROTECTED] incluí-lo em sua lista de autorizados. Atenção! Se você não conseguir clicar no atalho acima, acesse este endereço: http://tira-teima.as.uol.com.br/challengeSender.html?data=""> Hi, You´ve just sent a message to [EMAIL PROTECTED] In order to confirm the sent message, please click here This confirmation is necessary because [EMAIL PROTECTED] uses Antispam UOL, a service that avoids unwanted messages like advertising, pornography, viruses, and spams. Other messages sent to [EMAIL PROTECTED] won't need to be confirmed*. *If you receive another confirmation request, please ask [EMAIL PROTECTED] to include you in his/her authorized e-mail list. Warning! If the link doesn´t work, please copy the address below and paste it on your browser: http://tira-teima.as.uol.com.br/challengeSender.html?data=""> Use o AntiSpam UOL e proteja sua caixa postal --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003
RE: [Asterisk-Users] Meetme question
Basically what you need to do, unless you are installing zap devices, is uncomment the ztdummy line in the zaptel makefile, make install, and modprobe ztdummy. This will install a 'pseudo channel' driver for zap. (ie, emulating a zap device for applications needing zap interfaces). Hope this helps. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Johnson Sent: Thursday, September 25, 2003 1:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meetme question Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing '/etc/asterisk/meetme.conf': Found WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to open pseudo channel It then hangs up.. Anyone seen this before?? -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
Since * and MySQL have had a licensing scuffle, is there a way to set it up so that we can specify wether or not it's in the mysql database, or use the plaintext file that * generates with cdr_csv.so? Just a thought.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Carl Sent: Saturday, September 27, 2003 1:42 AM To: Asterisk Users (E-mail); Asterisk Dev (E-mail) Subject: RE: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, New versions available. Now written in PHP with totals for Billing Seconds and Duration. Help yourselves and please send me more suggestions!!! Thanx! J > -Original Message- > From: Dimitri Bellini [mailto:[EMAIL PROTECTED] > Sent: Friday, 26 September 2003 10:40 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] CDR Web Search Frontend > > > *This message was transferred with a trial version of > CommuniGate(tm) Pro* > Hi Carl > i see web frontend i action is very good!! The total > time at end is good > thing. > Thanks for great work. Can you put the script in some place > to download. > > Dimitri > > > *This message was transferred with a trial version of > CommuniGate(tm) Pro* > > > > Hey all, > > > > I've just done a quick (but functional) web front end for > searching the > > CDRs in a MySQL database. Anyone interested in trying it out? I'm > > wondering what to add to it next. > > > > So far you can seach using source, destination, CLI, > channel and date > > ranges. It also displays ALL fields in the database table. > > > > If interested, email me on [EMAIL PROTECTED] Do not reply > directly to > > this email, it will bounce. Depending on the level of > interest, I may > > post this somewhere for your free downloading pleasure. > > > > Regards, > > > > Jamie Carl > > Jazz Inc. > > http://www.jazz-inc.net > > Email: [EMAIL PROTECTED] > > JID: [EMAIL PROTECTED] > > Phone: +61-414-365466 > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
You wanted suggestions, didn't you? Well, you got them! :P Also another major search function... Allow to search via account code, for accounting purposes (obviously). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Carl Sent: Saturday, September 27, 2003 8:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* Gimmie a break, I only learnt PHP yesterday.. :) J - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, September 27, 2003 10:13 PM Subject: Re: [Asterisk-Users] CDR Web Search Frontend > *This message was transferred with a trial version of CommuniGate(tm) Pro* > > Since * and MySQL have had a licensing scuffle, is there a way to set it > > up so that we can specify wether or not it's in the mysql database, or > > use the plaintext file that * generates with cdr_csv.so? > > Or do something really smart like the Perl guys and have a > backend-mostly-independent DB infrastructure. Hell I think that PHP > finally smartened up and went this way, too. > > Regards, > Andrew > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) ------ Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault. Nowhere do they say the phone is not available. I was suprised when they told me it wasnt even out. When I sent a message to this thief, he said its for the 101 and they are hard to get. Thats why he jacked up the price. He did cancel my bid after telling me what a bad person I am for wasting his time. -- Original Message -- From: Andrew Kohlsmith <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Wed, 1 Oct 2003 23:32:51 -0400 >> Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as >> the 102D. And to make matters worse he starts the bid at $90.00 Beware. > >There's no need to beware -- anyone who doesn't shop around deserves to get >suckered. > >Regards, >Andrew >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX and IAXTEL
Apparently a lot of people forget about bugs.digium.com :P http://bugs.digium.com/bug_view_page.php?bug_id=296 DIRECTLY relates to your problem. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, October 02, 2003 5:36 AM To: ASTERISK USERS; bill black Subject: Re: [Asterisk-Users] IAX and IAXTEL Sometime yes sometimes no :) But thats the life :) Ok but I fixed it. Just put the "guest" "section" in iax.conf all the way on the end. And right now it works for me. :) -- Bart - Original Message - From: "bill black" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 02, 2003 12:47 AM Subject: Re: [Asterisk-Users] IAX and IAXTEL Hello Bart: Did anyone ever follow up to your question? I have the same issue. thanks, Bill On Wednesday 01 October 2003 07:27, Bartosz Jozwiak wrote: > Hello, > > Could somebody tell me what I should change in iax.conf file to be able to > receive calls from iaxtel. I am already registered and I can make calls to > IAXtel users but what I should do in iax.conf to be able to receive call > also. > > -- Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
My bad... It's a .net, not a .com :P Oops... Sorry JMB (sheepish grin) http://www.chagres.net -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam Sent: Thursday, October 02, 2003 6:22 AM To: '[EMAIL PROTECTED] ' Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault. Nowhere do they say the phone is not available. I was suprised when they told me it wasnt even out. When I sent a message to this thief, he said its for the 101 and they are hard to get. Thats why he jacked up the price. He did cancel my bid after telling me what a bad person I am for wasting his time. -- Original Message -- From: Andrew Kohlsmith <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Wed, 1 Oct 2003 23:32:51 -0400 >> Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as >> the 102D. And to make matters worse he starts the bid at $90.00 Beware. > >There's no need to beware -- anyone who doesn't shop around deserves to get >suckered. > >Regards, >Andrew >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-v
RE: [Asterisk-Users] eBay Sip Phone Scam.
Ok, see, now you're confusing what I said. Nowhere did I say I had the 102D. I said he never mentioned that it was the 102, irregardless of the D. I *DO* have the 101, which is what he was talking about. No, it doesn't mention it's the 101. This argument has now proved silly, especially since you're confusing what I'm saying, with what he supposedly is. *I CLAIM END OF THREAD!* -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 7:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Ok, in addition you are confusing the 102 with the 102D. If you had done your homework you would have noticed that the 102D (see the big D?) is a different model. Than one has the 16x2 LCD and 3 way conferencing. I spent a lot of time studying these phones. So no, you don't have that phone. check http://www.chagres.net -- Original Message -- From: "Josh Roberson" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 07:31:43 -0500 >My bad... It's a .net, not a .com :P > >Oops... Sorry JMB (sheepish grin) > >http://www.chagres.net > >-- >Josh Roberson >Indigent Networks >1.877.677.9647 x1 >[EMAIL PROTECTED] > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam >Sent: Thursday, October 02, 2003 6:22 AM >To: '[EMAIL PROTECTED] ' >Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. > >Josh, > >Pls can you confirm that URL, www.chagres.com doesn't seem to mention >the sale of any Grandstream phones > >Adam > >-Original Message- >From: Josh Roberson >To: [EMAIL PROTECTED] >Sent: 02/10/03 13:04 >Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. > >Actually, had you taken the time to READ the auction details, He says >(direct copy/paste from auction) > >-Begin Copy/Paste- > > >Flash Based OS > >Easy to install and manage, >Cost effective, >Easy to use - Friendly GUI for 1st time user, >Easy to learn - User's guide and on-line tutorial > >Big information and management LCD blue back light >User friendly keypad >Universal AC/DC adapter >Ergonomic design > > > > >25-button keypad >12-digit caller ID LCD >Universal Switching Power Adaptor >Input: 100-240VAC >Output: +5VDC, 400mA, > 1. Auto-sensing 10/100 Base-TX Ethernet Port >2.UL/CE/FCC >3.Power Supply : Universal 90 ~ 264V > > >Support all major Network Operating Systems (Windows, MAC, Linux/Unix) > >Web-Based Management > >TCP/IP Configuration with DHCP support > >Free Flash Firmware update > >No User Licenses > >System Restart/ Shutdown > >Password Access control > >1 x 10/100Mbps Ethernet Port (RJ-45 Interface) > >Support STUN and SIMPLE extension > >Interoperable with 3d parties Proxy, Registrar and gateway products > > > > > DSP technology for the best voice quality > >Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 >(alaw >and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) > >In and out-off-band DTMF > >Support 3-way conferencing (Model 102D), full duplex hands-free >speakerphone, >redial, call log, volume control, voice mail with indicator > >Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, >DiffServ, MPLS) > >Remote software upgrade capability via TFTP > >Support Silence Suppression, VAD (Voice Activity Detection), CNG >(Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC >(Automatic Gain >Control) > >-End Copy/Paste- > >Nowhere does he make the claims you're stating. He DOES, have (Model >102D) in one of the descriptions, but that is a direct quote from >Grandstream's product brochure. > >Also, this phone *IS* out on the market.. I own one, and I'm quite happy >with it.. I will tell you this though: > >Go order one from Chagres (http://www.chagres.com). They are an >asterisk supporter/user on this list, and the price is MUCH better. ;) > > > >-- >Josh Roberson >Indigent Networks >1.877.677.9647 x1 >[EMAIL PROTECTED] > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of costas >Sent: Thursday, October 02, 2003 4:50 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. > >I did shop around. Nowhere does he say the phone is the
RE: [Asterisk-Users] IAX and IAXTEL
Well, that's odd.. Can you, then, with IAX, determine in which section (first, second, last, etc...) you read your configuration in iax.conf, rather than matching up with passwords? -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Thursday, October 02, 2003 7:16 AM To: ASTERISK USERS Subject: Re: [Asterisk-Users] IAX and IAXTEL The location of the "guest" / "iaxtel" section having to be at the end is, as it turns out, a configuration error on iaxtel. I hope to have it straightened out shortly. Mark On Thu, 2 Oct 2003, Bartosz Jozwiak wrote: > Sometime yes sometimes no :) But thats the life :) > > Ok but I fixed it. Just put the "guest" "section" in iax.conf all the way on > the end. > And right now it works for me. :) > > -- Bart > > - Original Message - > From: "bill black" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, October 02, 2003 12:47 AM > Subject: Re: [Asterisk-Users] IAX and IAXTEL > > > Hello Bart: > > Did anyone ever follow up to your question? I have the same issue. thanks, > Bill > > On Wednesday 01 October 2003 07:27, Bartosz Jozwiak wrote: > > Hello, > > > > Could somebody tell me what I should change in iax.conf file to be able to > > receive calls from iaxtel. I am already registered and I can make calls to > > IAXtel users but what I should do in iax.conf to be able to receive call > > also. > > > > -- Bart > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transfer from IAX call
I recreated your problem using the 09/09 cvs source tree. The only way I found around that is to create the extensions.conf context for [NANPA], like this: [NANPA] include => local ; Allow transfers from IAX calls to cellphone through NuFone in local pbx context (ie, 1000 for queue, 1001 for Admin, 1002 for Tech) Works like a charm. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Friday, October 03, 2003 7:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Transfer from IAX call I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called [EMAIL PROTECTED]/1515480 -- Call accepted by 65.127.126.42 (format GSM) -- Format for call is GSM -- IAX2[NuFone]/3 is ringing -- IAX2[NuFone]/3 stopped sounds -- IAX2[NuFone]/3 answered Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 -- Playing 'pbx-transfer' -- Unable to find extension '1' in context 'NANPA' -- Playing 'pbx-invalid' -- Stopped music on hold on Zap/1-1 How do I make this work? dave -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Ringing from PSTN
Well, the ATA uses SIP to communicate with the * box. SIP by default doesn't generate a ringing indicator when the far side is ringing, you indeed DO have to tell it to ring, using the r flag in the extension. **Note, this is just from my experience. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, October 09, 2003 3:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringing from PSTN That does make a ringing sound, but any idea what's causing the problem? Stephen Subject: Re: [Asterisk-Users] No Ringing from PSTN You can send a fake ring by using something like: exten => 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: > Here is my Configuration > > PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 > > When I call from the pstn to the ATA, the ATA rings but I don't hear > anything on the calling side until the call is picked up. > > When I call from the ATA, everything seems to work fine. > > When I bypassed ASTERISK, everything seems to work fine. > > Anyone know what I might have configured wrong? > > Thanks, > > Stephen > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
I would like to beta test this tool. :) Looks like it could be a good thing. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: Friday, October 17, 2003 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 2003-10-22 at 15:38, Michael T Farnworth wrote: > On Wed, 22 Oct 2003, Michael T Farnworth wrote: > I take this back, as a protocol tftp is hideously complex compared to > http and would take a lot more code. RFC 1350 (tftp v2): 11 pages RFC 2616 (http/1.1) : 114 pages josh. -- --- Josh Howlett, Networking & Digital Communications, Information Systems & Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip and budgetone
Is there a way to checkout a two week old version of the cvs?
Jon
CVS has an incredably flexable date specification routine:
-D date_spec
Use the most recent revision no later than date_spec (a
single argument, date description specifying a
date in the past). A wide variety of date formats are
supported, in particular ISO ("1972-09-24 20:05")
or Internet ("24 Sep 1972 20:05"). The date_spec is
interpreted as being in the local timezone, unless
a specific timezone is specified. The specification is
``sticky'' when you use it to make a private
copy of a source file; that is, when you get a working
file using -D, cvs records the date you speci-
fied, so that further updates in the same directory will
use the same date (unless you explicitly over-
ride it; see the description of the update command). -D
is available with the checkout, diff, history,
export, rdiff, rtag, and update commands. Examples of
valid date specifications include:
1 month ago
2 hours ago
40 seconds ago
last year
last Monday
yesterday
a fortnight ago
3/31/92 10:00:07 PST
January 23, 1987 10:05pm
22:00 GMT
So there should be no problem with cvs update -D "2 weeks ago"
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RE: [Asterisk-Users] grandstream ntp
I've noticed the same problem on the BT-102. I would also like to know this... (cc'ed grandstream to get their opinion) -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rodger Sent: Friday, November 07, 2003 7:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] grandstream ntp I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I hate to do this but..
I hate to bring this thread back to life, but... > it may be possible to get it supported, do you think the price >point is remotely competitive with Digium hardware? Also as I am not >about to divulge my information to them to look in the downloads >section, what is the licensing of their SDK? What is the licensing of >the driver? > >Steven >On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote: >> Is there any current/planned support for Aculab hardware? >> >>http://www.aculab.com >> >> Looks like they have Linux drivers and an SDK. Has any advancement taken place in this? Has someone developed a working channel driver for this product? I have one, and would like to see if it would be a possibility to get working... Thanks -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Graphical Interface
I would like to propose the name… “astmaster control”… in all seriousness…. I agree, this isn’t a name for an actual possible business implementation, but I think it has a nice ring to it for a project name.. J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Friday, November 14, 2003 8:57 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Graphical Interface Hello, I don't have a project name yet, any suggestions? What in your mind should a full client app have in it? This program pretty much has everything that my company needs from a client app in it. What other things (within the limitations of a Zap/Sip Asterisk system with unmodified source code) need to be added to it to make it complete? MATT--- -Original Message- From: marin blu [mailto:[EMAIL PROTECTED] Sent: Friday, November 14, 2003 9:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Graphical Interface Hi, What is the project name? Do you thing that your project could be a step to a full client appl ? Best Regards, Marin Blu mattf <[EMAIL PROTECTED]> wrote: Hello, I have developed a graphical interface using Perl/TK that has the following features: I'm still cleaning up the client code, but it will be released before the end of the month on Sourceforge. Here are some of the things I have added to the code: - Recording of any Zap channel by extension they are connected to at the click of a button - A refreshing list of active Zap channels - dialing a number by entering in a number or selecting from a list of recently dialed numbers and clicking a DIAL button - Asterisk based conference-calling of up to 6 external channels(even on single-line phone) - Admin section that allows you to Hangup any Zap channel at the click of a button - Call Parking and retrieval from specific extensions - Runs on Linux and Windows On the server side you will need a MySQL server, a couple AGI scripts and some custom dialplan extensions, but the Asterisk code itself is unaltered. On the Client side you just need to have perl and Tk/tcl modules installed on Linux and on windows you just need Activestate perl, you also need to make sure you have the Net:Telnet and Net::MySQL perl modules loaded on both(these are easy to get and have no prerequisites). MATT--- -Original Message- From: David Winkler [mailto:[EMAIL PROTECTED] Sent: Thursday, November 13, 2003 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Graphical Interface Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003
RE: [Asterisk-Users] MeetMe problem
Also, unless something has changed, If you don't have any zap devices, you'll need to have the ztdummy module loaded to provide zap timing to the meetme app. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, November 15, 2003 8:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MeetMe problem On Saturday 15 November 2003 08:59, [EMAIL PROTECTED] wrote: > Hi guys, > Having a bit of a problem trying to get conference bridges working. > In my meetme.conf file I have the following line > [rooms] > conf => 6000 > > > In my extensions.conf file I have: > exten => 1000,1,MeetMe,6000 > > My problem is that when I dial into extension 1000 it is telling me > "this is not a valid conference number". Can anybody telling me what > I'm doing wrong here? You need to do a restart after defining new conference numbers, otherwise they won't work (i.e. not on a reload). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.538 / Virus Database: 333 - Release Date: 11/10/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.538 / Virus Database: 333 - Release Date: 11/10/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App Queue
Does anyone have a good HOWTO on queues Is your computer infected with a virus? Find out with a FREE computer virus scan from McAfee. Take the FreeScan now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Van Donselaar
> Sent: Tuesday, November 18, 2003 10:12 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for
WinXP,
> Red Hat 9.0
>
> On Tue, 18 Nov 2003 17:02:42 +0200, "Dan" <[EMAIL PROTECTED]> wrote:
>
> >Hi,
> >
> >Tried on WinXP Pro and it loads, but in the background (no window).
> >There is something needed from the wxWindows package to just run the
> >executable?
>
> Nothing needed from the wxWindows package. I think it's because it
can't
> find
> the rc directory.
>
> I'm sorry that I didn't put this in the README. Bad coder. No donut.
>
> You must run iaxComm from the installation directory beacuse it looks
for
> rc
> files in ${cwd}/rc.
>
> Steve put an error dialog on failure in the CVS sources, but I'm
working
> on a
> better solution.
>
> Please let me know if this solves it, or if the problem lies
elsewhere.
Nope still crashes on XP on load. Ran from directory extracted to, etc.
Below are crash details:
AppName: iaxcomm.exe AppVer: 0.0.0.0 ModName: iaxcomm.exe
ModVer: 0.0.0.0 Offset: 0008e98c
Don't know if that helps any at all, but the other details screen is WAY
too long to attach.
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.538 / Virus Database: 333 - Release Date: 11/10/2003
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Re: [Asterisk-Users] PSTN intercepted announcement
Michael Ulitskiy wrote:
Hi,
I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN
gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI.
Everything works fine with one exception. I seem to be unable to figure out why I cannot hear
PSTN intercepted announcement ("number is not in service" etc.) when I'm calling
a disconnected number through asterisk.
AFAIK, A PRI is normally expected to signal number not in service
conditions out of band, so * should be signalling the out of service
condition in a manner appropriate for the channel type (as a recording,
or as the most accurate protocol specific out of service response code
available, casung the out of service condition to be indicated by the
target equipment)
With my SNOM phone, the PRI signals not in service or no route or
whatever, then * relays that in the form of a SIP error response, then
the SNOM phone executes an internal recording and shows the error on the
display as well.
This results in more reliable call handling all the way through, and
some bandwidth savings.
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RE: [Asterisk-Users] cisco 7960 power suplies?
Also, I see them on eBay all the time for around $35 US. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Lists > Sent: Sunday, November 30, 2003 5:49 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] cisco 7960 power suplies? > > Does anyone know where to get cisco 7960 power suplies? What should they > cost? > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T400P and 2.4.23 kernels
Anyone able to confirm whether the T400P (or any other Zap device) works with the 2.4.23 kernels? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XBOX as and * Dedicated Server
I have done this, but I haven't put the server in place yet... It appears to run absolutely fabulous, with the exception that OSS/dsp is noisy as all get-out. Alsa drivers tend to fix this problem, though. Other than that, I can pass at least 5-10 calls through with no problem whatsoever. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Miguel Cavazos > Sent: Thursday, December 04, 2003 6:43 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] XBOX as and * Dedicated Server > > Hello guys, i have been on this mailing list for some weeks now, and i > was wondering if someone here has installed linux on the XBOX and use it > as a dedicated server. Its a 200 USD computer and could make it perfect > to asterisk, its little and doesnt really take much space. My question > is could this make it for a stable server??? > > here are some links i found for linux on XBOX > http://xbox-linux.sourceforge.net/ > > some intresting screenshots found on that URL > http://xbox-linux.sourceforge.net/docs/screenshots.html > > The only real thing that i dont know is where am i going to put the > X100p. > > Miguel > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XBOX as and * Dedicated Server
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Senad Jordanovic > Sent: Friday, December 05, 2003 5:18 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] XBOX as and * Dedicated Server > > Miguel Cavazos wrote: > > On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote: > > > >> During Phreaknic, Mark was showing off a Xbox running asterisk with 4 > >> S100U interfaces connected to the game ports on the front. It was > >> interesting. In the end, I don't think it is cost effective as a real > >> PC since you can also build a PC of similar or better specs for that > >> price now and you get PCI slots. > > > > the S100U is a good idea, and yes you can get a pc for what 30bucks a > > P200, but i was looking for something small and good looking, i dont > > have a big room and another CPU. > > > > Miguel > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > I like you idea. Very Cool :) > Is RAM upgradable on xbox? > > Thanks > Yes the ram is upgradable... *IF* you can do extremely small surface mount soldering. You can upgrade it to a whopping 128 megs. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
Brian West wrote: Have you had issues with the Hold button and flash button? The MUTE button is my biggest concern. Its CRITICAL for that to work in a callcenter enviroment! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID trunks -- equipment requirement
>Hi guys, > > >I'm looking to setup a 16 extension / 10-14 phone line Asterisk install >for a customer who would like to have DID numbers for the extensions, >since they're currently on Centrex and already have the 1-to-1 >correspondence. Since I'm in a less populated area of the country, SBC >doesn't seem to have much in the way of fractional T1 products (on the >scale that we need them) available, so I think my only option for DID is >to use (analog) DID trunks for incoming calls and POTS lines for >outbound calls. I'd look into ISDN, both PRI and BRI. If the costs are not too prohibitive, this would be the most flexable option. ISDN uses out of band signaling and has a number of features which complement a DID enviroment, such as DNIS (dialed number information service), where the number dialed is passed along with an incoming call. If your enviroment never uses all its outside lines at the same time, this can be cost effective, because you can have direct dial numbers for all the phones without a one to one correspondance of outside lines to extensions. PRI is usually too expensive, but sometimes BRI is affordable, however you should check on pricing for both to see if they may be cost effective. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent setup
CW_ASN wrote:
Shad:
Using the AddQueueMember. Launching this command 3 times in different
queues, logs one phone to that 3 queues...
*CLI> show application AddQueueMember
-= Info about application 'AddQueueMember' =-
[Synopsis]:
Dynamically adds queue members
[Description]:
AddQueueMember(queuename[|interface]):
Dynamically adds interface to an existing queue
Returns -1 if there is an error.
Example: AddQueueMember(techsupport|SIP/3000)
*CLI>
Or, you must declare only one extension for all agents, i.e:
;LogOn and LogOff in queue: cola
exten => icola,1,EAGI(opinc.php,cola)
exten => icola,2,Hangup
exten => dcola,1,EAGI(opdel.php,cola)
exten => dcola,2,Hangup
Easier but poorly documented solution. AgentCallbackLogin()
AgentCallbackLogin delivers callo for a logged in agent to an extension.
- they continue to get calls until they log out (by logging in to a null
extension (pressing # when prompted for extension)
The AGIs contains:
---> opinc.php:
#!/usr/bin/php -q
ob_implicit_flush(true);
set_time_limit(0);
function sm($texto) {
echo("VERBOSE \"".$texto."\"\n");
}
function se($comando) {
echo("exec $comando\n");
}
//MAIN PROCEDURE
$err = fopen("php://stderr","w");
$in = fopen("php://stdin","r");
while (!feof($in)) {
$temp = str_replace("\n","",fgets($in,4096));
$s = split(":",$temp);
$agi[str_replace("agi_","",$s[0])] = trim($s[1]);
if (($temp == "") || ($temp == "\n")) {
break;
}
}
$cid=trim($agi["callerid"]);
$ext=trim($agi["extension"]);
$tec=trim($agi["type"]);
$rt1=strpos($cid,"<");
$rt2=strpos($cid,">");
if (strpos($cid,"<")>0) {
$cid=trim(substr($cid,($rt1+1),(($rt2-$rt1)-1)));
} else {
$cid=trim($cid);
}
$cid=stripslashes($cid);
$cola=$argv[1];
se("AddQueueMember $cola $tec/$cid");
se("Playback agent-loginok");
sm("Agent '$tec/$cid' was included in queue '$cola'");
?>
---> opdel.php:
#!/usr/bin/php -q
ob_implicit_flush(true);
set_time_limit(0);
function sm($texto) {
echo("VERBOSE \"".$texto."\"\n");
}
function se($comando) {
echo("exec $comando\n");
}
//MAIN PROCEDURE
$err = fopen("php://stderr","w");
$in = fopen("php://stdin","r");
while (!feof($in)) {
$temp = str_replace("\n","",fgets($in,4096));
$s = split(":",$temp);
$agi[str_replace("agi_","",$s[0])] = trim($s[1]);
if (($temp == "") || ($temp == "\n")) {
break;
}
}
$cid=trim($agi["callerid"]);
$ext=trim($agi["extension"]);
$tec=trim($agi["type"]);
$rt1=strpos($cid,"<");
$rt2=strpos($cid,">");
if (strpos($cid,"<")>0) {
$cid=trim(substr($cid,($rt1+1),(($rt2-$rt1)-1)));
} else {
$cid=trim($cid);
}
$cid=stripslashes($cid);
$cola=$argv[1];
se("RemoveQueueMember $cola $tec/$cid");
se("Playback agent-loggedoff");
sm("Agent '$tec/$cid' was removed from queue '$cola'");
?>
I know, the code is dirty... but it works for me.
Hope this helps,
Regards,
Gus
pgp0.pgp
Description: PGP signature
RE: [Asterisk-Users] Re: Grandstream Early Dial
I've never had early dial working, however, I resolved my multiple digit issue by simply putting both the GS phones and asterisk in INFO mode. This worked on both 10.0.3.81 firmware on the budgetone and the ATA286, as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but doubelcheck your lines in asterisk to be dtmfmode=info and the gs devices are on SIP INFO method, and your DTMF Payload type is 101. Just my $.02 -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Stephen R. Besch > Sent: Wednesday, December 31, 2003 12:59 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Re: Grandstream Early Dial > > > > I've just checked my voicemail with 1.0.4.30 and get the same multiple > > digits problem. sip.conf and GS config are both at info, for me this is > > a new problem voicemail has always worked perfectly with the GS. > > > This has come up many times in this list, with no consensus for a > solution. According to Grandstream, the "multiple digit" problem arises > from a difference in the interpretation of the SIP standard. I'm not > sure I really understand this, so no flames please, but, paraphrasing a > conversation I had with GS, apparently they retransmit the digit as long > as the key is pressed and expect asterisk to know that it is a > re-transmission by examining other data in the packet. Asterisk does not > handle the SIP packet in the way GS expects, resulting in multiple digit > transmission. This flaw (?) is avoided by setting DTMF to INBAND. Why > this behaviour is not repeatable on everyones installations escapes me. > However, I have noticed one thing that may be a clue. I have one phone > that is older hardware (redial button instead of send and an unused > battery compartment on the bottom). This phone behaves differently than > all the other, later, models. For example, it is the only phone on > which the flash button actually works to answer the alternate line (eg > when an incoming call waiting call arrives). All phones are using 3.81 > firmware. > > > Early dial has never worked for me, and I just upgraded to the > > 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, > > making it impossible to check my voice mail. > > This is an acknowleged bug on the GS. They have connected to my * > server and acknowleged the problem. A fix has been promised but not yet > delivered. Until then, the only solution is to turn early dial off and > let the phone send the entire dial string in one packet. Since this > does not affect later single digit transmission for IVR's, etc, the only > consequence is the irritating delay between the last entered digit and > the actual placing of the call. But, you can always hit the send key. > > Stephen R. Besch > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
Well, since everyone else is top-quoting on this message, so will I :P I'm no veteran either. As a matter of fact, I have had ZERO prior knowledge to the telcom industry or more than 'user level' experience with telecommunications in general. I decided that I wanted to expand my knowledge, and actually LEARN a few things, so I jumped into asterisk. I was, and quite frankly, IMO, still AM a 'n00b' to *. However, after playing around, and learning what things do, by reading the documentation that IS there, searching the archives, and just trolling the list and IRC, I have learned more in the last 4-5 months of having * than a lot of people I've noticed have learned in a lifetime of experience.I now have a fully functional (well, minus MOH, because mpg123 isn't yet compiled on my new box), * implementation, serving myself and my roommates strictly over VoIP, and a couple ata's and a Internet PhoneJack card. I love it. And I'm STILL learning to this date. Asterisk is not something you can expect everyone to just drop what their doing and help you with. Sure, it can be frustrating, but if you are so dense that you can't sit down an play with it and learn what happens when you type something in the cli, or change a few things in your dialplan, then get out, I agree. If you liked taking apart mom's hairdryer as a kid and seeing how it worked, and then later on, rewired up a few things to do what you wanted them to, or even took a hex editor to command.com in msdos to change what it says to suit your taste (mucho guilty on that one.. lol), then you will have no problem finding out what you can and can't change simply by editing files, and trying things out. Take off your training wheels, and just TRY IT. - Josh R. [EMAIL PROTECTED] > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of SW > Sent: Wednesday, December 31, 2003 4:13 PM > To: [EMAIL PROTECTED] Digium. Com > Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. > > Hello, > > I am not a veteran here, but would like to share my thoughts on this > subject. > > True, * is opensource and freely available, but it is not a computer > program > that you download and run. It is a very versatile telecommunication > product > you would otherwise pay at least 100 K to buy from a telecom vendor, if > not > more based on modules and usage, license hash-codes etc. > > Even to try * one would need some pre requisite knowledge in telecom, if > not > many years in the field. I work for a large telecom company and my > specialty > is voice over broadband (or xDSL). I worked with asterisk for couple of > months now and I am amazed to see areas of telecom that * touch upon with. > Starting from Linux, to SIP, H323, DSL technologies (PPP, PPPoE, PPPoA, > DHCP, NAT), Call routing(Dial Plan), IVR, Transcoding, STUN are few areas > that one would have to master even thinking about *. > > True one would know the syntax, and howtos etc, but also would have to > have > the ability to troubleshoot. For last two-three months in this list, I > have > not seen any newbi posting a sip trace (from a ethereal or a TCP dump) and > asking a question about it. I have seen many question for instance, asking > syntax of h.323 dial, but never seen a question asked on a h323 trace. > > I think, having * openly available is like keeping an airplane openly > available in a airfield, so that anybody can try flying. Tell me how many > of > us would go try and fly that airplane if we do not know how to fly :) > > Point that I want to make here is simple, please try to understand what * > is > all about. If you like it's features and would like it to run in a > production environment try to get some professional help. If you are > learning these technologies for fun then get educated, use tools available > to troubleshoot. Hooking up couple of phones and making a call is far from > knowing *. > > Asterisk is a great product (thanks Mark and many others) and if you know > what you are doing, you can do wonders with it. Don't put it down, because > you do not have the background to understand it or work with it. > > Cheers > > SW > > > > Message: 4 > Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) > From: Me <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. > Reply-To: [EMAIL PROTECTED] > > As a newcomer to Asterisk, you will not be welcomed > with open arms. First, you will find almost no > documentation on it's features. Second, if you try to > ask questions, you will be flamed and pointed to > worthless how-tos and 'the wiki
[Asterisk-Users] Cisco SIP license?
In order to use a cisco phone and the SIP image, do you need a license, or just the firmware. Is this like saying that you can get music, all you need is something like kazza? Or if you get the phone and the image are you legit? What does it take to get the lic? Josh _ Get reliable dial-up Internet access now with our limited-time introductory offer. http://join.msn.com/?page=dept/dialup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Hmm Just my $.02 - no flames please. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Balaji NJL > Sent: Saturday, January 03, 2004 8:31 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... > don't walk. > > Hi All, > > Can we stop this thread pl. This lady has no > intentions to learn asterisk. > She is just a troll and wasting our time. With her > corporate attitude, what > she expects is support that available with paid > commercial products. Her > company has enough money to buy commercial products, > let she go there. Hey > lady, whoever u are, dont waste our time. this is not > for u. > > Lets move on to something useful pl. > -B > > - Original Message - > From: "Steven Critchfield" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, January 03, 2004 5:36 PM > Subject: RE: [Asterisk-Users] New to asterisk? RUN... > don't walk. > > > > On Sat, 2004-01-03 at 14:12, Me wrote: > > > Mr. West, > > > > > > Sorry to burst your bubble, but that is not me. > My > > > name is Barbara Simpson. Either you are lying or > > > someone is trying to remove any credibility from > my > > > original post. I now stand by my original post > with > > > more conviction than ever. > > > > You had little to no credibility when you show up > acting like a troll > > from what most people would consider a throw away > account. > > > > > There were a lot of insightful replies. However, > none > > > of them were able to address the real problems of > the > > > asterisk community and come up with solutions. If > you > > > can't see your own faults, you are in for a bumpy > > > ride. > > > > This is due to the problem residing in the general > population, not the > > community. The problem resides in users who can't be > bothered to either > > expend energy, or patience for the software to > develop. Remember you > > came here, we didn't go recruiting you. So if you > are disappointed in > > your experience, blame yourself for your > expectations. As far as I can > > tell here, you haven't paid a single person for > anything, so any help > > you have received has been at a cost to the other > people of this > > community. > > > > So the solution is for you to grow up. You need to > learn that the > > comment you have made in this thread are worthless > as they don't advance > > anything here. If you want credibility in a > technical forum, you will > > have to show some technical skills. Otherwise you > will be cast aside and > > hopefully ignored. > > > > > Barbara Simpson > > > Qwest Voice Over Packet Services > > > > > > --- Brian West <[EMAIL PROTECTED]> wrote: > > > > You said it good Look what this person > posted to > > > > my blog... Now thats > > > > what I call grown up. > > > > > > > > Date: Thu, 1 Jan 2004 10:10:24 -0600 > > > > From: [EMAIL PROTECTED] > > > > To: [EMAIL PROTECTED] > > > > > > > > IP Address: 24.10.200.168 > > > > Name: Jeff Sowery > > > > Email Address: [EMAIL PROTECTED] > > > > URL: > > > > > > > > Comments: > > > > > > > > You're a complete idiot. Grow a brain or at > least > > > > some balls. > > > > > > > > -Jeff > > > > > > > > > > > > NEXT!!! > > > > > > > > bkw > > > > > > > > > > > > On Thu, 1 Jan 2004, JR Richardson wrote: > > > > > > > > > Piping in 2 cents, > > > > > > > > > > This is a great example of the Internet, Fast > Food > > > > generation, showing their > > > > > appreciation for all the magic that happens in > the > > > > labs, hearts and minds of > > > > > the courageous, hard working, dedicated and > > > > motivated group of people truly >
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Hmm Just my $.02 - no flames please. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Balaji NJL > Sent: Saturday, January 03, 2004 8:31 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... > don't walk. > > Hi All, > > Can we stop this thread pl. This lady has no > intentions to learn asterisk. > She is just a troll and wasting our time. With her > corporate attitude, what > she expects is support that available with paid > commercial products. Her > company has enough money to buy commercial products, > let she go there. Hey > lady, whoever u are, dont waste our time. this is not > for u. > > Lets move on to something useful pl. > -B > > - Original Message - > From: "Steven Critchfield" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Saturday, January 03, 2004 5:36 PM > Subject: RE: [Asterisk-Users] New to asterisk? RUN... > don't walk. > > > > On Sat, 2004-01-03 at 14:12, Me wrote: > > > Mr. West, > > > > > > Sorry to burst your bubble, but that is not me. > My > > > name is Barbara Simpson. Either you are lying or > > > someone is trying to remove any credibility from > my > > > original post. I now stand by my original post > with > > > more conviction than ever. > > > > You had little to no credibility when you show up > acting like a troll > > from what most people would consider a throw away > account. > > > > > There were a lot of insightful replies. However, > none > > > of them were able to address the real problems of > the > > > asterisk community and come up with solutions. If > you > > > can't see your own faults, you are in for a bumpy > > > ride. > > > > This is due to the problem residing in the general > population, not the > > community. The problem resides in users who can't be > bothered to either > > expend energy, or patience for the software to > develop. Remember you > > came here, we didn't go recruiting you. So if you > are disappointed in > > your experience, blame yourself for your > expectations. As far as I can > > tell here, you haven't paid a single person for > anything, so any help > > you have received has been at a cost to the other > people of this > > community. > > > > So the solution is for you to grow up. You need to > learn that the > > comment you have made in this thread are worthless > as they don't advance > > anything here. If you want credibility in a > technical forum, you will > > have to show some technical skills. Otherwise you > will be cast aside and > > hopefully ignored. > > > > > Barbara Simpson > > > Qwest Voice Over Packet Services > > > > > > --- Brian West <[EMAIL PROTECTED]> wrote: > > > > You said it good Look what this person > posted to > > > > my blog... Now thats > > > > what I call grown up. > > > > > > > > Date: Thu, 1 Jan 2004 10:10:24 -0600 > > > > From: [EMAIL PROTECTED] > > > > To: [EMAIL PROTECTED] > > > > > > > > IP Address: 24.10.200.168 > > > > Name: Jeff Sowery > > > > Email Address: [EMAIL PROTECTED] > > > > URL: > > > > > > > > Comments: > > > > > > > > You're a complete idiot. Grow a brain or at > least > > > > some balls. > > > > > > > > -Jeff > > > > > > > > > > > > NEXT!!! > > > > > > > > bkw > > > > > > > > > > > > On Thu, 1 Jan 2004, JR Richardson wrote: > > > > > > > > > Piping in 2 cents, > > > > > > > > > > This is a great example of the Internet, Fast > Food > > > > generation, showing their > > > > > appreciation for all the magic that happens in > the > > > > labs, hearts and minds of > > > > > the courageous, hard working, dedicated and > > > > motivated group of people truly >
RE: [Asterisk-Users] rtp sound quality?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson >pstn -> sip gw -> * -> C7960 > >When I dial into * via the pstn, I hear the ivr menu just fine (good >quality). I press 3000 (valid extn), and I begin to hear ringing however >the ring back is very very choppy. Where are you getting timing from? Zaptel device? Ztdummy? -Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf / misdn
This may be what you need: http://www.misdn.org/index.php/FAQ#Why_are_my_dtmf_tones_not_detected_everytime.3F Also, something here may be helpful: http://www.voip-info.org/wiki/view/Asterisk+DTMF#Troubleshooting -jr On Nov 6, 2007 2:12 PM, Hans Witvliet <[EMAIL PROTECTED]> wrote: > Hi all, > > Perhaps someone can give me a hint i the right direction... > > Sometimes dtmf is recognized, sometimes not. > I'm using 1.2.19 asterisk with misdn for my hfc card. > When i got in incoming sip-call, dtmf is recognized, > When i phone my self (isdn-phone or gsm-phone) no problem with dtmf > When SOME (not all) people phone me (isdn-incoming) DTMF is not > recognized. > How come? > > Either it works for a particular configuration, or it doesn't. > It doesn't make sense to me that it works sometimes... > > -- Josh Richards - Grover Beach, California US [EMAIL PROTECTED] (don't forget the middle 't' initial when writing) http://blog.joshrichards.org/ 805/471-6923 (cell) Geek Research (Technology Management Consulting) - http://www.geekresearch.com/ Support These Nifty Causes: http://Kiva.org http://RoomToRead.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] Board configuration - specification or recommendation
To determine what you are going to need, you'll want to start by discussing what you plan to do with the system, including capacity (e.g. number of users/extensions, types of calls, etc.). I'd suggest some research, starting with the following resources which have a lot of good background, theory, as well as real-world experiences from other users: http://www.voip-info.org/wiki/view/Asterisk+dimensioning http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations http://www.voip-info.org/wiki/view/Asterisk+hardware#PCServerMotherboards There are also off-the-shelf appliance style solutions or you can build your own: http://www.voip-info.org/wiki/view/Asterisk+embedded+systems http://www.rowetel.com/ucasterisk/store.html http://www.digium.com/en/products/appliance/ -jr On Nov 7, 2007 2:56 AM, Kim Joung-il <[EMAIL PROTECTED]> wrote: > Hello, > > We're about to deploy an Asterisk system. > So far we have the following (below) configuration > but before we start anything we would like to hear some > suggestions on it, specification or recommendation too... > > Thank you! > > miniITX board > dual core CPU > 2 LAN > 1x IDE (or more) > 2x SATA (or more) > 2x PCI slots (must work with Linux correctly in regards to IRQs) > > Optional: > On board hardware RAID > miniPCI > CF slot > IPMI interface > > > -- Josh Richards - Grover Beach, California US [EMAIL PROTECTED] (don't forget the middle 't' initial when writing) http://blog.joshrichards.org/ 805/471-6923 (cell) Geek Research (Technology Management Consulting) - http://www.geekresearch.com/ Support These Nifty Causes: http://Kiva.org http://RoomToRead.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] Channel variables, any difference with SIP vs. IAX?
Off-hand, have you compared the output of "agi debug" (on the console) between the working and non-working calls? I believe the variables all get displayed. -jr On Nov 8, 2007 6:13 AM, Jason Wolfe <[EMAIL PROTECTED]> wrote: > I have some extensions that are using variables loaded by an AGI program. > Everything works fine and I am able to use NoOp to see the value of my > variables when using IAX, but the same variables don't work when using SIP. > I can provide further details, but right off of the bat does is there > something I need to know about the use of user defined variables in with SIP > channels vs. IAX channels? > -- Josh Richards - Grover Beach, California US [EMAIL PROTECTED] (don't forget the middle 't' initial when writing) http://blog.joshrichards.org/ 805/471-6923 (cell) Geek Research (Technology Management Consulting) - http://www.geekresearch.com/ Support These Nifty Causes: http://Kiva.org http://RoomToRead.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] Requiring a login to a phone
For such a simple application I'd use AstDB to avoid having to hassle with
an external database (and also means this sort of dialplan will work even on
embedded/slimmed Asterisk boxes that may not have db modules
loaded/available). In any case, what Tilghman said is what I'd suggest as
well.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db
http://www.the-asterisk-book.com/unstable/funktionen-db.html
Also consider allowing emergency number dialing to bypass authentication, if
applicable.
-jr
On Dec 1, 2007 5:32 PM, Tilghman Lesher <[EMAIL PROTECTED]>
wrote:
> On Saturday 01 December 2007 18:09:27 Steve Johnson wrote:
> > Hi List,
> >
> > We have a remote asterisk SIP phone at the cottage.
> >
> > I'd like it to have minimal privileges when it first registers with
> > Asterisk. Ideally it should be in a restricted context. Dialing any
> > number would intercept the call and tell the person to log on. This
> > way, if the phone was stolen or someone got into the cottage, we
> > wouldn't have a bunch of surprise charges on our phone bill... :-)
> >
> > Once the phone has been authenticated, it should go into a context
> > with normal privileges. After a couple of days of non-use, it should
> > auto-logout to the restricted context.
> >
> > How can I change the sip context of a phone on the fly, based on
> > authentication login?
>
> I wouldn't. I'd do authentication on the fly, using a database of some
> kind.
>
> extensions.conf:
> [sip-phones]
> exten => _X.,1,Set(lastlogin=${ODBC_LOGIN(${CUT(CHANNEL,-,1)})})
> ; Logins expire after 86400 sec = 24 hours
> exten => _X.,n,GosubIf($[0${lastlogin} + 86400 < ${EPOCH}]?restricted,s,1)
> exten => _X.,n,Dial(Zap/g1/${EXTEN})
>
> [restricted]
> ; VMAuthenticate terminates the call if authentication fails.
> exten => s,1,VMAuthenticate
> exten => s,n,Set(ODBC_LOGIN(${CUT(CHANNEL,-,1)})=${EPOCH})
> exten => s,n,Set(lastlogin=${EPOCH})
> exten => s,n,Return
>
> func_odbc.conf:
> [LOGIN]
> dsn=asterisk
> read=SELECT lastlogin FROM logins WHERE channel='${ARG1}'
> write=UPDATE logins SET lastlogin=${VAL1} WHERE channel='${ARG1}'
>
> logins.sql:
> CREATE TABLE logins (
>channel CHAR(50) PRIMARY KEY,
>lastlogin INTEGER,
> );
> INSERT INTO logins VALUES ('SIP/100',0);
> INSERT INTO logins VALUES ('SIP/101', 0);
> INSERT INTO logins VALUES ('SIP/102', 0);
>
>
--
Grover Beach, California, USA
http://blog.joshrichards.org[EMAIL PROTECTED]+1 (805)
471-6923
http://www.linkedin.com/in/joshrichards
Supporting these causes: Kiva.org & RoomToRead.org
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[asterisk-users] phoniceq e400p driver for DAHDI
Hello everyone, We have an E400P Card from phoniceq. There is a DAHDI Driver posted at: http://e400p.phoniceq.com/driver/dahdi-tor2-tormenta3-e1.tgz but, it doesn't work. The author (martin (or marcin) pycko) says that it isn't finished. I've e-mailed martin, and he stated that he would fix the driver AFTER we order 10 more cards, not before. I have a difficult time ordering thousands of dollars worth of product BEFORE i see it working correctly. During my life, I've been promised a lot of things If anyone can fix the driver, I'll gladly pay a reasonable fee for the service. The hardware seems to work well, and it definately fits my budget, but i'm more comfortable paying a few dollars to get this done BEFORE i buy the cards. Cheap insurance. To replicate my setup, you would need to download dahdi-linux-complete-2.0.0.tar.gz , untar it, and then download the file above and copy the contents over the top of .../dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi/tor2.c Below is the debug information I provided when I first contacted phoniceq. If anyone here thinks they can tackle this for me, Please get in touch via direct email and let me know how much you want to fix it up. Thanks Everyone, Josh --- When I insmod/modprobe tor2.c, however, I get a segmentation fault, and I can't use the driver, or even unload it. The only way to remove the driver is to reboot the machine. develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # uname -a Linux develop 2.6.25.5-1.1-pae #1 SMP 2008-06-07 01:55:22 +0200 i686 i686 i386 GNU/Linux develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # insmod tor2.ko Segmentation fault develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # rmmod tor2 ERROR: Module tor2 is in use develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # *NOTE*(tor2 is NOT in use)**NOTE* develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # dahdi_cfg -vvv DAHDI Tools Version - 2.0.0 DAHDI Version: 2.0.0 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: Clear channel (Default) (Slaves: 32) Channel 33: Clear channel (Default) (Slaves: 33) Channel 34: Clear channel (Default) (Slaves: 34) Channel 35: Clear channel (Default) (Slaves: 35) Channel 36: Clear channel (Default) (Slaves: 36) Channel 37: Clear channel (Default) (Slaves: 37) Channel 38: Clear channel (Default) (Slaves: 38) Channel 39: Clear channel (Default) (Slaves: 39) Channel 40: Clear channel (Default) (Slaves: 40) Channel 41: Clear channel (Default) (Slaves: 41) Channel 42: Clear channel (Default) (Slaves: 42) Channel 43: Clear channel (Default) (Slaves: 43) Channel 44: Clear channel (Default) (Slaves: 44) Channel 45: Clear channel (Default) (Slaves: 45) Channel 46: Clear channel (Default) (Slaves: 46) Channel 47: Clear channel (Default) (Slaves: 47) Channel 48: D-channel (Default) (Slaves: 48) Channel 49: Clear channel (Default) (Slaves: 49) Channel 50:
[asterisk-users] Override sip.conf settings in extensions.conf? Possible?
Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.ful...@telus.com The views expressed in this e-mail are mine alone and do not necessarily reflect the views of my employer. ___ -- 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] Override sip.conf settings in extensions.conf? Possible?
> > Does anyone know if it is possible to override sip.conf settings in > extensions.conf > > (for example: session-minse=90) without needing to create an overarching > peer in sip.conf > > and selecting it specifically in the dial plan? > > > > You can do this to some extent starting with Asterisk 1.6.1. With the > AST_CONFIG > function, you can change a configuration file from the dialplan. The > problem is > that you would also have to reload the configuration file so that the > change > would take effect. After the call was completed, you would then have to > reset > the value of the option and reload the config file again, since you only > want > the option set for one call. > > If this doesn't sound absolutely horrible to you and you want the same > functionality in Asterisk 1.4, you may be able to get away with simply > copying > func_config.c from Asterisk 1.6.1 into Asterisk 1.4's funcs/ directory. I > haven't tried this myself, so I don't know what tweaks, if any, would be > required to make the code compile. > > > I'm on the 1.4 stable code base and looking to implement session-timers > on certain call > > flows in a modular dial plan. > > > > (Sorry if I'm not making the correct logical leap here) > > Being able to set the session-timers variables via the dialplan will not > be > sufficient in 1.4 in order to enable session timers on certain calls. You > would > also have to modify chan_sip.c so that the Asterisk would understand the > concept > of session timers and how to properly behave. > > Mark Michelson Thanks for the fantastic answer, Mark. I'm hesitant to migrate to 1.6 because some of the servers I want to make these changes on are production units. I think-- based on what you've suggested-- that the best action for me would be to clone my sip peer definitions in sip.conf and add the specific session-timers I need for origination in the peer. Then, I can just call the alternate peer with timers invoked. I've used this successfully in the past to create unknown and anonymous calls so I think this is the easiest course. I'm not opposed to hacking code but it seems redundant when chan_sip technically already has the functionality I need. At any rate, great answer! You did make the correct logical leap, by the way. It's nice to see an answer right from Digium on this question as well. We use your cards in our labs and production and find your hardware very useful. Thanks, Josh Fuller josh.ful...@telus.com The views expressed in this e-mail are mine alone and do not necessarily reflect the views of my employer. ___ -- 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] VoIP over satellite internet
> Message: 10 > Date: Fri, 8 May 2009 20:30:11 -0700 > From: Eric Fort > Subject: [asterisk-users] VoIP over satellite internet > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > <2ad2af430905082030w389822aduc877f8b0a1afe...@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Could those on the list who have used or tried to use VoIP over a > satellite internet connection comment on how well it works or if it > even works at all in a reliable way. What is the effect of latency on > the VoIP path and how much is generally tolerable? routing via > satellite adds about a quarter second of latency to the path. Is that > too much? It is possible-- barely-- but you have to be able to put up with two to six second lags between replies and lots of stepping on each other in conversations. The feasibility will also depend on the traffic shaping/filtering of the provider and whether they black hole VoIP ports/packets. There will be a lot of delay and echo which can be compounded by an imbalance in upstream and downstream bandwidth. If you're using dialup for upstream the bandwidth _will_ be an issue. If you're looking for point-to-point communication a client-to-client push-to-talk solution like Speak Freely [1] might be a better choice. You may also want to consider-- if you're trying to use Asterisk-- a narrowband codec such as Speex. [2] I used Speak Freely over 28.8 dialup links to have conversations between Florida and Ontario almost fifteen years ago. It's more like a two-way radio than a telephone but it works very well and is win/lin cross-platform. [1] http://speak-freely.sourceforge.net/ [2] http://speex.org/ Thanks, Josh Fuller josh.ful...@telus.com The views expressed in this e-mail are mine alone and do not necessarily reflect the views of my employer. ___ -- 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] Step-by-Step Asterisk and MeetMe Help
> Message: 19 > Date: Tue, 19 May 2009 22:20:59 +0300 > From: Tzafrir Cohen > Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help > To: asterisk-users@lists.digium.com > Message-ID: <20090519192059.gb3...@xorcom.com> > Content-Type: text/plain; charset=us-ascii > > On Tue, May 19, 2009 at 11:11:40AM -0700, Jimmy Ezell wrote: > > > > > > >-Original Message- > > >From: asterisk-users-boun...@lists.digium.com > > >[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Tzafrir > > >Cohen > > >Sent: Tuesday, May 19, 2009 11:03 AM > > >To: Asterisk Users Mailing List - Non-Commercial Discussion > > >Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help > > > > > > > > >On Mon, May 18, 2009 at 05:26:21PM -0700, Jimmy Ezell wrote: > > > > > >> make[2]: Entering directory > > >`/usr/src/kernels/2.6.9-78.0.13.EL-smp-i686' > > > > > >You seem to be using RHEL4.7 or Centos 4.7 . > > > > > >Is this a new installation? > > > > > >-- > > > Tzafrir Cohen > > > > Centos 4.7 and yes I installed this new for this Asterisk project. > > > > Jimmy > > http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html > > This is the latest version of Centos4. However, why would you use > Centos4 on a new installation? > > If you want Centos, go with Centos 5 (now at 5.3). > > You install bison, which is no longer required to build Asterisk. > > You install Zaptel, but DAHDI is now the "latest version of Zaptel". > Zaptel is now frozen and has not been updated for quite some time. > > You set the /usr/src/linux-2.4 symlink , which is a pointless old piece > of woodoo. Can you point me to the place in the documentation that > pointed you to it? > > (Not to mention that the whole messing with kernels is so bad in Centos. > It is not better in SUSE, where you have to install the full kernel > source to build Zaptel / DAHDI). > > Next: no need to run 'make clean' before running ./configure. It is > mearly pointless. > > 'make menuselelct' is something you normally don't need to run, unless > you know you want to remove things. In other words: don't let it > needlessly complicate your installation instructions. * snip* You seem to be missing the headers for your distro. You may wish to... a) remove all those symlinks you created b) yum install kernel-devel (or kernel-smp-devel as the case may be) - or, the less painful road - a) move to a deb-based distro such as debian (or ubuntu if you're not well versed in *nix) and get out of rpm hell. b) install headers: apt-get install build-essential linux-headers-`uname -r` c) still having problems? Install all the dev packages for zaptel: apt-get build-dep zaptel Thanks, Josh Fuller josh.ful...@telus.com The views expressed in this e-mail are mine alone and do not necessarily reflect the views of my employer. ___ -- 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 & Adit 600 Configuration
Has anyone ever gotten an Adit 600 to work with Asterisk1.4 via MGCP.
Asterisk keeps giving me the following error in the LOGs:
[Jun 30 08:32:59] NOTICE[26785]: chan_mgcp.c:1726
find_subchannel_and_lock: Gateway 10.0.0.245' (and thus its endpoint
'*') does not exist
MGCP Config:
[AFSWestAdit600]
host = dynamic
context = default
canreinvite = no
threewaycalling = yes
cancallforward = yes
transfer = yes
callwaiting = yes
slowsequence = yes
line => aaln/3
line => aaln/2
line => aaln/1
Extensions.conf
[default]
exten = 3412,1,Dial(MGCP/aaln/1...@afswestadit600)
exten = 3413,1,Dial(MGCP/aaln/1...@afswestadit600)
exten = 3414,1,Dial(MGCP/aaln/1...@afswestadit600)
Adit Config
set verification off
set 6 autologout 0
-set 6 password view {password} is manual
-set 6 password config {password} is manual
-set 6 password admin {password} is manual
-set 6 enhanced security enable is manual
-set 6 password security {password} is manual
set 6 priority tos 0xFC 0xB8
delete 6 remote "RemoteUnit"
set 6:1 framing ipx ieee8023 disable
set 6:1 framing ipx ieee8022 disable
set 6:1 framing ipx snap disable
set 6:1 framing ipx ethii disable
set 6:1 ip address 10.0.0.245 255.255.255.0
set 6:1 gateway 10.0.0.1
set 6 dns domain "local.local"
set 6 dns name "afswestadit600"
set 6 dns server 1 10.0.0.135
set 6 dns resolver enable
set 6:1 up
add 6:1 static ip network 10.0.0.0 255.255.255.0 10.0.0.1 1
add 6 remote "RemoteUnit"
set 6 snmp name "unknown"
set 6 snmp contact "unknown"
set 6 snmp location "unknown"
set 6 "RemoteUnit" up
set 6 log last detail
set 6 mgcp callagent address 10.0.0.167
set 6 mgcp gatewayid 10.0.0.245
set 6 mgcp quarantine step discard
set 6 mgcp port 2727
set 6 mgcp up
set 6 mgcp rsipwildcard enable
set 6 mgcp tos 0x68
set 6 voip osi 500
set 6:1:1:1 log start both
set 6:1:1:1-48 echo tail 64
set 6:1:1:1-48 tos 0xB8
set 6:1:1:1-48 algorithm preference g711mu g729a
set 6:1:1:1-48 dtmfrelay enable
set 6:1:1:1-48 cpd osi
reset 6
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[asterisk-users] Phone system "ping" checker
Does anyone know of an online tool or program that would call our phone numbers to confirm that they are up? What I'm imagining is something like pingdom.com (a ping checker for web services), except for the phone system. It would call us, confirm an answer, then hang up. If no answer, it would send an e-mail to our cell phones so we knew about it right away. It seems like there would be a market for this type of tool, but I can't seem to find any service or software that will do what I'm wanting. Any suggestions? -- Josh Hunholz ___ -- 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] TellMe Voice Recognition in Asterisk working..
A couple of weekends ago I decided to see if I could get Asterisk to play nice with TellMe's VoiceXML studio. They provide the VoiceXML studio for free, and you can access it through SIP, so I thought this would be a fun and cheap way to integrate voice recognition into my IVR. I have posted a brief tutorial with code and examples on the voip-info.org wiki ( http://www.voip-info.org/wiki/index.php?page=Add%20Voice%20Recognition%20to%20Asterisk ) as well as at my blog ( http://www.spinepunch.com ). The code is a little rough, and I'm sure there is a better way of doing what I did, but this was easy and it worked for me. What's next on my to-do list is trying to cover up the TellMe jingle before it starts the VoiceXML app. If anyone would like to help clean up the code, or has a better way of interacting with the Asterisk manager, please let me know. Thanks, -Josh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom and "hint" priority
(boy mail in this list piles up fast when I can't check it) On Apr 8, 2005, at 10:03 AM, Michael George wrote: - It appears that the extension used with the "hint" must be the same as the extension used to dial that channel. So if extension 22 will ring Zap/2, then "exten => 22,hint,Zap/2" will work, but "exten => 222,hint,Zap/2" will not. Why is that? The extension is how asterisk maps SIP URLs to chunks of your dialplan -- if you program a button on a snom to "dest ", the phone will use that same URL for both dialing and subscribing to extension state. Unless you have a phone that lets you specify different URLs for dialing and subscribing to state, they have to match in asterisk. - If I am correct in the above, then there is no way for me to monitor a channel that is not an extension. As an example, I have a TDM400 with 3 FXS (Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel for dialing out. I can monitor the states of the extensions with extension entries like "exten => 21,hint,Zap/1" but I cannot monitor the state of the FXO with "exten => 0,hint,Zap/4" because 0 is not the extension of Zap/4. Indeed, Zap/4 has no extension. Is it not possible to monitor that line, then? There has to be a SIP URL for the phone to subscribe to -- if you put: exten => zap4,hint,Zap/4 in your extensions.conf (with no zap4,1,... entry) it wouldn't be dialable (although the phone would still try if you pushed it) but would have a valid SIP URL. -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom and "hint" priority
On Apr 17, 2005, at 12:23 AM, Lance Grover wrote: I have rebooted the phone and restarted asterisk after each change. Did you do it in that order? If so, that is probably a source of trouble (you should restart or reload asterisk before the phone boots, not after). -- Joshua P. Dady http://www.indecisive.com/ smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wIPPhone with Asterisk
Has anyone sucessfully set up wIP Phone for asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] clarent hardware
Just wondering if anyone has had any luck with the Clarent CPGs ( uses MGCP ). I have a couple CPG 201s laying around that I am trying to get working but am having difficulties. They successfully register with the asterisk box, but when I lift the handset of the phone plugged into any of its ports, there is no dial tone, I hear no DTMF tones when I press keys, etc. But I can make the phone ring by using a soft phone, although when I lift the handset it just keeps ringing. Any help would be great, I did see a post I found with google about someone getting CPG 101s going well, but they never posted any config files or how they did it. I also can't find any good information about using MGCP with asterisk, any links, tips, help would be appreciated. Thanks in advance! ---- Josh Krueger <[EMAIL PROTECTED]> Urban Communications http://www.urbancom.net/
[Asterisk-Users] toll access - account code
Our telco has setup toll access account codes for outgoing calls. I would like to include these account codes in the dialplan for certain extensions (fax lines, modems) so that they are not prompted for the 4 digit code when making a toll call. I have played around with the 'w' command with ZAP channels, commas, and DISA, all with no success. I see a lot of examples of Asterisk with account codes, but they are all using internal codes, not those setup by a telco. Has anyone set this up successfully or can point me in the right direction? Thanks, Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: toll access - account code
Here is an example of my dialplan, where 1234 is the account code required
by my telco for a toll access call.
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1}1234}
exten => _91NXXNXX,2,Congestion
When I include the 1234, with any variation of www before the code, I
get a "Reorder" error on my Cisco 7940 talking SIP. Asterisk CLI spits this
out:
Jun 24 12:07:34 WARNING[1234455344]: pbx.c:1215 pbx_extension_helper: No
applic
ation 'Dial(Zap/g1/${EXTEN:1}1234}' for extension (test, 919074749606,
1)
== Spawn extension (test, 919074749606, 1) exited non-zero on
'SIP/4299-cee1'
If I take out the www1234, I can complete the call, but have to enter in the
account code by hand.
Any ideas?
Thanks,
Josh
-Original Message-
From: Josh Reineke
Sent: Thursday, June 24, 2004 9:40 AM
To: '[EMAIL PROTECTED]'
Subject: toll access - account code
Our telco has setup toll access account codes for outgoing calls. I would
like to include these account codes in the dialplan for certain extensions
(fax lines, modems) so that they are not prompted for the 4 digit code when
making a toll call. I have played around with the 'w' command with ZAP
channels, commas, and DISA, all with no success. I see a lot of examples of
Asterisk with account codes, but they are all using internal codes, not
those setup by a telco. Has anyone set this up successfully or can point me
in the right direction?
Thanks,
Josh
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[Asterisk-Users] RE: toll access - account code
Sorry for thinking out loud here. The bracket at the end of the first line
should have been a parantheses.
-- Executing Dial("SIP/4299-cf70", "Zap/g1/190747496061234") in
new stack
-- Called g1/190747496061234
-- Hungup 'Zap/1-1'
Now I get the prompt for the access code, but it doesn't appear to dial the
1234 to complete the dial string.
?
-----Original Message-
From: Josh Reineke
Sent: Thursday, June 24, 2004 12:15 PM
To: '[EMAIL PROTECTED]'
Subject: RE: toll access - account code
Here is an example of my dialplan, where 1234 is the account code required
by my telco for a toll access call.
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1}1234}
exten => _91NXXNXX,2,Congestion
When I include the 1234, with any variation of www before the code, I
get a "Reorder" error on my Cisco 7940 talking SIP. Asterisk CLI spits this
out:
Jun 24 12:07:34 WARNING[1234455344]: pbx.c:1215 pbx_extension_helper: No
applic ation 'Dial(Zap/g1/${EXTEN:1}1234}' for extension (test,
919074749606, 1)
== Spawn extension (test, 919074749606, 1) exited non-zero on
'SIP/4299-cee1'
If I take out the www1234, I can complete the call, but have to enter in the
account code by hand.
Any ideas?
Thanks,
Josh
-Original Message-
From: Josh Reineke
Sent: Thursday, June 24, 2004 9:40 AM
To: '[EMAIL PROTECTED]'
Subject: toll access - account code
Our telco has setup toll access account codes for outgoing calls. I would
like to include these account codes in the dialplan for certain extensions
(fax lines, modems) so that they are not prompted for the 4 digit code when
making a toll call. I have played around with the 'w' command with ZAP
channels, commas, and DISA, all with no success. I see a lot of examples of
Asterisk with account codes, but they are all using internal codes, not
those setup by a telco. Has anyone set this up successfully or can point me
in the right direction?
Thanks,
Josh
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[Asterisk-Users] RE: toll access - account code
>The last char on the line needs to be a ) not a }
>
>"w" only works on ANALOG PSTN ports.
>
>Recent CVS -head has an option to send digits after the remote side
answers. See "show application dial"
I'm running Asterisk CVS-HEAD-06/15/04-17:27:40
I now have this in my dialplan:
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1},,D(1234))
exten => _91NXXNXX,2,Congestion
With the results:
-- Executing Dial("SIP/4299-5393", "Zap/g1/19074749606||D(1234)") in new
stack
-- Called g1/19074749606
Still get the prompt for the access code. I have relaxdtmf set to yes in
zapata.conf? I can't do a complete restart of asterisk until off hours, but
will try setting it to no to see if it makes a difference.
Thanks guys,
Josh
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Re: [Asterisk-Users] Which Linux ?
Gentoos great but everytime I see people talk about it and ask if theres any special USE flags or crap like that to make somthing compile right I just cant help but laugh. and heres why : http://funroll-loops.org/ everyone needs a good laugh.. I'm not insulting Gentoo or anything, I like it. - Original Message - From: "Remco Barende" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 02, 2004 10:22 AM Subject: RE: [Asterisk-Users] Which Linux ? > Cool, did you just use the standard ebuilds in portage (although the > 'unstable' versions) ror did you build from cvs? > > I have just received my hardware and want to build asterisk on a gentoo > box too :) > > > On Fri, 25 Jun 2004, Kevin Walsh wrote: > > > Ed Brady [EMAIL PROTECTED] wrote: > > > I am about to build my first asterisk box, I want to make it Gentoo based > > > with a 2.4 kernel. > > > > > I'm on 2.6.7-gentoo-r6, which I installed today (upgraded from r5). > > I have found the 2.6 kernel to be a lot better, in my unscientific > > opinion, than the 2.4 kernel I've used in the past. > > > > > > > > When performing the initial system emerge on the Gentoo bos, are there > > > any special USE flags you would recommend setting to make the asterisk > > > build go smoothly? > > > > > I don't set the USE variable at all when I built my Asterisk box. > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
