[asterisk-users] Manager events safety
Hello list, is it save to send manager events from a remote website (php) to an Asterisk-server ? Is there some tutorial on how to implement a tight safety policy ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager events safety
Use simple PHP's telnet classes for AMI. If you need special security - use Stunnel (SSL tunnel) and iptables on asterisk side for IP forwarding. This all is really straight-forward, I doubt you need a tutorial here.. Both stunnel and PHP Telnet have tutorials on how to accomplish this. You just need to merge 2 techniques :) Good luck On Sat, Apr 24, 2010 at 12:30 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, is it save to send manager events from a remote website (php) to an Asterisk-server ? Is there some tutorial on how to implement a tight safety policy ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Improving CLI Help - was [Re: 1.6.2 No soft hangup?]
Jose P. Espinal wrote:
Hi,
Maybe you could do something in shellscript too:
e.g.
asterisk -rx help | grep -ia something
That would behave just as describe in the suggestion (but it's easier to
do :P)
You could place that in a tiny shellscript, that takes the 'something'
as an argument:
#!/bin/bash
token=$1
asterisk -rx help | grep -ia ${token}
Save that with the name of your preference (somewhere inside your $PATH,
would be nice), and just execute it like a normal command:
[name_you_gave_it] sip
[name_you_gave_it] peer
[name_you_gave_it] whatever
Note:
You then could make a lot of fancy customizations to parameters of your
script, etc., and even use other tools for if needed (e.g. gawk, sed, etc.)
sean darcy wrote:
Olle E. Johansson wrote:
Further to Steve Edward's comment, I think things would be more
obvious if the help system was improved slightly, for instance:
If you were trying to figure out the commands dealing with peers, you
would be able to type:
*CLI help peer
No peer command found. Possible alternatives:
iax2 show peer Show details on specific IAX peer
iax2 show peers List defined IAX peers
sip show peers List defined SIP peers
sip show peer Show details on specific SIP peer
(and so on, maybe using the [More] option to help it be
readable)
In this case, if I could use the help system to search on all
occurrences of the word hangup in the available commands, I would
probably have found it myself instead of bothering the list.
THat's a very good idea. Thank you!
Now we need someone that codes it :-)
/O
Well I'm certainly not the one who could code it, but is there any way
to simply grep all the help. So, for instance, if you did help
hangup you got:
hangup request no description available
which you now get,
followed by all the commands that have hangup in them, including their
descriptions. For instance:
But see:
channel request hangup channel
Request that a channel be hung up. The hangup takes effect
the next time the driver reads or writes from the channel
etc
etc
sean
Very slick. Using your script as asterisk-help:
asterisk-help hangup
channel request hangup Request a hangup on a given channel
console hangup Hangup a call on the console
hangup request no description available
sean
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Re: [asterisk-users] Manager events safety
Hi, Please avoid HTML-only mail. On Sat, Apr 24, 2010 at 12:30 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello list, is it save to send manager events from a remote website (php) to an Asterisk-server ? Is there some tutorial on how to implement a tight safety policy ? What do you mean by safe? For instance, allowing a remote user to apss any event through a manager account that has full priviliges, provides that remote user practically full control over Asterisk. Is that a problem for you? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] RTP over TCP
On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote: Hi Guys, On 04-23-2010 21:40, Nathan Clemons wrote: SIP is just the control protocol, and can be negotiated over TCP or UDP. The actual payload is done over RTP, which is a UDP-based protocol. thanks, for both of you for pointing this out. i was obviously on the wrong track here. since i see the rtp traffic via tcpdump, i'm going to ask the other gw's sysadmin to see into this, maybe there is some logging on the other side. If you had to add firewall exceptions/PAT config for the TCP SIP traffic, you'll also need to add the same for RTP traffic as well. this is a private pilot network for testing purposes, no internet connection, no nat, no firewall. it's like the 90s :) Actually, it is more common for RTP to be over UDP, but RTP over TCP is possible. In fact, this is the default RTP mode for M$ Office Communications Server. I beleive that it may be possible to use RTP over TCP in Asterisk as there was someone work on this inorder to have interop with M$. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP over TCP
Doesn't surprise me if Microsoft tries sending RTP over TCP. Maybe their engineers didn't know basics of VoIP when they were programming their communication server. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-24 12:04 PM, Michael Graves mgra...@mstvp.com wrote: On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote: Hi Guys, On 04-23-2010 21:40, Nathan Cle... Actually, it is more common for RTP to be over UDP, but RTP over TCP is possible. In fact, this is the default RTP mode for M$ Office Communications Server. I beleive that it may be possible to use RTP over TCP in Asterisk as there was someone work on this inorder to have interop with M$. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com sip%3amgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Coloc... -- _ -- 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] RTP over TCP
Hi! Doesn't surprise me if Microsoft tries sending RTP over TCP. Maybe their engineers didn't know basics of VoIP when they were programming their communication server. Not quite - doing SIP over TCP rather then UDP is the right thing to do (tm). It's just that everyone started out with UDP in the early ages of SIP - also think about the benefit of TLS. As for MS OCS: That uses TCP for SIP and UDP for RTP. Where MS failed: OCS _only_ does SIP over TCP, whereas Asterisk _only_ does/did SIP over UDP. Neither is/was RFC compliant. Philipp -- _ -- 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] RTP over TCP
On Fri, Apr 23, 2010 at 3:21 PM, ad...@3a.hu wrote: i have to put an * between two other SIP gateways and due to some circumstances, i have to use sip over tcp. With 1.6.2.6 this is working fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B (ocs) and that's about it. In the other direction however (ocs - me - deverto4) the call setup is complete but there is no audio. Don't do it. There have been any number of posts to asterisk-users begging asterisk to bend over backwards to accommodate Microsoft's foray into the world of VoIP. Nobody seems to be asking Microsoft to build a stack compatible with the rest of the world of VoIP. I disagree that sending SIP over TCP is superior to sending it over UDP. Think about it for a second. UDP is 'unreliable' in that lost packets are not rebroadcast. Now let's say you have an 'unreliable' connection where it's just barely good enough that the SIP call setup goes through, but the RTP stream immediately fails. Why would that be superior to having the SIP call setup getting dropped? The end result of no reliable voice is the same, but now you have a funkier debug condition that's going to be more complex to track down. I personally think it would be superior to see the bad connection as early in call setup as possible. And of course, SIP over UDP is the way the rest of the world works. If anybody wants to speak up about a framework that supports BOTH SIP over UDP AND SIP over TCP, maybe somebody already built a middleware layer that will let Microsoft join the world of voip. -- _ -- 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] RTP over TCP
RTP stands for Real-Time Transport Protocol. TCP is not designed to deal with real-time data transfer as it takes time to acknowledge packets and re-send them if missing. All audio video data transfer happens in real time, and it doesn't make any sense to retransmit missing packets. Real time packets mixed with old missing packets which are submitted would result in an non-understandable audio and video. So how come any system can use TCP for real time data transfer, while assuring the quality at the same time. Does their exist any such system? I would certainly like to try it, maybe they are doing it right using some trick which I don't know yet. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-24 1:48 PM, David Backeberg dbackeb...@gmail.com wrote: On Fri, Apr 23, 2010 at 3:21 PM, ad...@3a.hu wrote: i have to put an * between two other SIP ga... Don't do it. There have been any number of posts to asterisk-users begging asterisk to bend over backwards to accommodate Microsoft's foray into the world of VoIP. Nobody seems to be asking Microsoft to build a stack compatible with the rest of the world of VoIP. I disagree that sending SIP over TCP is superior to sending it over UDP. Think about it for a second. UDP is 'unreliable' in that lost packets are not rebroadcast. Now let's say you have an 'unreliable' connection where it's just barely good enough that the SIP call setup goes through, but the RTP stream immediately fails. Why would that be superior to having the SIP call setup getting dropped? The end result of no reliable voice is the same, but now you have a funkier debug condition that's going to be more complex to track down. I personally think it would be superior to see the bad connection as early in call setup as possible. And of course, SIP over UDP is the way the rest of the world works. If anybody wants to speak up about a framework that supports BOTH SIP over UDP AND SIP over TCP, maybe somebody already built a middleware layer that will let Microsoft join the world of voip. -- _ -- Bandwidth and Colocati... -- _ -- 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] Asterisk and Archlinux
Hi all, Is anyone here using Asterisk on Archlinux? If so, was it much to do in order for it to work? Do you also use Dahdi? many thanks, Christian -- _ -- 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] RTP over TCP
On Sat, 2010-04-24 at 10:56 -0500, Michael Graves wrote: On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote: Hi Guys, On 04-23-2010 21:40, Nathan Clemons wrote: SIP is just the control protocol, and can be negotiated over TCP or UDP. The actual payload is done over RTP, which is a UDP-based protocol. thanks, for both of you for pointing this out. i was obviously on the wrong track here. since i see the rtp traffic via tcpdump, i'm going to ask the other gw's sysadmin to see into this, maybe there is some logging on the other side. If you had to add firewall exceptions/PAT config for the TCP SIP traffic, you'll also need to add the same for RTP traffic as well. this is a private pilot network for testing purposes, no internet connection, no nat, no firewall. it's like the 90s :) Actually, it is more common for RTP to be over UDP, but RTP over TCP is possible. In fact, this is the default RTP mode for M$ Office Communications Server. I beleive that it may be possible to use RTP over TCP in Asterisk as there was someone work on this inorder to have interop with M$. No, that was just sip over tcp (instead of udp) I friend of mine had an * talking to M$, so that was one of the reasons for early deployment of an 1.6.x asterisk... -- _ -- 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] RTP over TCP
Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over TCP. We are actually developing a flash phone which needs only TCP to transmit both signal and audio. -Bruce On Sat, Apr 24, 2010 at 2:01 PM, Zeeshan Zakaria zisha...@gmail.com wrote: RTP stands for Real-Time Transport Protocol. TCP is not designed to deal with real-time data transfer as it takes time to acknowledge packets and re-send them if missing. All audio video data transfer happens in real time, and it doesn't make any sense to retransmit missing packets. Real time packets mixed with old missing packets which are submitted would result in an non-understandable audio and video. So how come any system can use TCP for real time data transfer, while assuring the quality at the same time. Does their exist any such system? I would certainly like to try it, maybe they are doing it right using some trick which I don't know yet. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-24 1:48 PM, David Backeberg dbackeb...@gmail.com wrote: On Fri, Apr 23, 2010 at 3:21 PM, ad...@3a.hu wrote: i have to put an * between two other SIP ga... Don't do it. There have been any number of posts to asterisk-users begging asterisk to bend over backwards to accommodate Microsoft's foray into the world of VoIP. Nobody seems to be asking Microsoft to build a stack compatible with the rest of the world of VoIP. I disagree that sending SIP over TCP is superior to sending it over UDP. Think about it for a second. UDP is 'unreliable' in that lost packets are not rebroadcast. Now let's say you have an 'unreliable' connection where it's just barely good enough that the SIP call setup goes through, but the RTP stream immediately fails. Why would that be superior to having the SIP call setup getting dropped? The end result of no reliable voice is the same, but now you have a funkier debug condition that's going to be more complex to track down. I personally think it would be superior to see the bad connection as early in call setup as possible. And of course, SIP over UDP is the way the rest of the world works. If anybody wants to speak up about a framework that supports BOTH SIP over UDP AND SIP over TCP, maybe somebody already built a middleware layer that will let Microsoft join the world of voip. -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP over TCP
Adobe has not been successful in VoIP. I have worked for two companies who were trying to make flash phones, and it always came down to the issue of RTP over TCP. This is the primary reason there are no successfully working flash phones out there though some companies are trying to offer service over them. I once set it up myself using redphone and red5. But a java based RTP over UDP is what I ended up with, which simply works. I would suggest you to look into a java based solution. RTP is simply not meant for TCP, those who are doing it, they must not be using TCP as TCP. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-24 3:07 PM, Hans Witvliet h...@a-domani.nl wrote: On Sat, 2010-04-24 at 10:56 -0500, Michael Graves wrote: On Fri, 23 Apr 2010 23:11:06 +0200, adamk... No, that was just sip over tcp (instead of udp) I friend of mine had an * talking to M$, so that was one of the reasons for early deployment of an 1.6.x asterisk... -- _ -- Bandwidth and Colocat... -- _ -- 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] PrivacyManager
Hi thwe PrivacyManger app states thast you can use a context to match against for the input , but gives no real examples or explaination, does anyone have a an example context for this Thanks in advance Robb -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Archlinux
On Sat, Apr 24, 2010 at 21:19, Christian christia...@runbox.com wrote: Hi all, Is anyone here using Asterisk on Archlinux? Yes and no, I do use it on Archlinux for testing purpose but not as a server. Arch linux is not built to be a server distro, unlike Debian that have extra steps for process handling, like restarting a service that was just updated and more. If so, was it much to do in order for it to work? You need to either use the AUR builds or download the ABS information and build it for yourself. Personally I use the yaourt tool as a pacman front end. Do you also use Dahdi? Like any other Asterisk it must have a dahdi module, at least dahdi_dummy. many thanks, Christian -- _ -- 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 Ido -- _ -- 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] Jitter Buffer and MeetMe.
On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty russian.qwe...@gmail.com wrote: Hello. As I see, there is a lot of threads about jitter buffer... Maybe anybody knows something about my case? Any help will be appreciate. So, the problem with voice quality was completely solved, BUT some customers have informed me about big latency. It's really hard to make dialogue with current latency. You're on the right track here, but I don't think your problem is jitter. I think your problem is VoIP and voice activity detection, and depending on your version of asterisk, MeetMe conference 'talker optimization'. I've posted all of this before. Here goes again... * 'talker optimization' should be disabled on MeetMe() conferences. * /etc/asterisk/dsp.conf set silencethreshold=1024 * /etc/asterisk/codecs.conf set vad=false Give those a try, restart or reload asterisk to apply changes, and tell us if it fixes it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions
Hi, I look after this but have been very busy for months. Maybe you canhelp me test? Thanks, Gavin. On 23/04/2010, Sean Brady sbr...@gtfservices.com wrote: Not sure if this is the right place to ask, but what do we need to do to get this patch merged? How can I help? I'm no dev, but I use LDAP with Asterisk and I might be of some help. Thanks guys. -- _ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Monitoring tools........
Hey all What VoIP networking monitoring, asterisk monitoring tools would you recommend? I started working with an IT company that insists on using DSL with a Sonicwall router. The problem is that the clients are having sound problems. The owner is convinced that it's the Asterisk box. In the 4 yrs I have been doing this I have not had this bad a sound problem and it always came down to a bad setup in the Cisco router. Asterisk just doesn't have sound problems so I am going to have to convince him that its either the router or DSL. Has anyone used DSL for SIP traffic? How about Sonicwall routers? Michael D Mosier -- _ -- 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] VoIP monitoring tools
Howdy all 1. does anyone know a good voip / sip / qos monitoring tool? 2. Has anyone had luck running asterisk phone systems over DSL? 3, Has anyone used sonic wall routers for qos over dsl. The company I am consulting for would like to install asterisk boxes over dsl with sonicwall routers. For the last 4 years I have installed all my boxes over t1 lines with cisco routers and have had no sound problems, The couple of clients we tried this out on are having sound issues so you see why I would like to find a good monitoring tool. We are using Broadsoft SIP trunks. Thanks for any suggestions. Michael D Mosier -- _ -- 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] VoIP monitoring tools
I think DSL is 1/2 duplex and in most cases way to slow on the way Up for VOIP. I use a sonic wall on a T1 and it works great. It even has some features for tweaking VOIP. Any time I have tried VOIP on a 1/2 duplex connection all the way up to 7down and 1.5 up I have call quality issues. At 10:06 PM 4/24/2010, you wrote: Howdy all 1. does anyone know a good voip / sip / qos monitoring tool? 2. Has anyone had luck running asterisk phone systems over DSL? 3, Has anyone used sonic wall routers for qos over dsl. The company I am consulting for would like to install asterisk boxes over dsl with sonicwall routers. For the last 4 years I have installed all my boxes over t1 lines with cisco routers and have had no sound problems, The couple of clients we tried this out on are having sound issues so you see why I would like to find a good monitoring tool. We are using Broadsoft SIP trunks. Thanks for any suggestions. Michael D Mosier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP monitoring tools
hello, mike mosier schrieb: Howdy all 1. does anyone know a good voip / sip / qos monitoring tool? you could try smokeping or iperf but real monitoring of the dsl quality isnt easy. 2. Has anyone had luck running asterisk phone systems over DSL? we dont run asterisk itself over dsl, but our clients connect over dsl to the asterisk server in our housing center. 3, Has anyone used sonic wall routers for qos over dsl. i dont know sonic wall routers, but sonic wall firewalls are the death of voip. every time a customer has used one of these, he got problems like call drops, packet loss and so on. The company I am consulting for would like to install asterisk boxes over dsl with sonicwall routers. For the last 4 years I have installed all my boxes over t1 lines with cisco routers and have had no sound problems, The couple of clients we tried this out on are having sound issues so you see why I would like to find a good monitoring tool. the only way of doing this right, is that the ISP of your dsl lines install some kind of QOS for you. If you make this by your own, you will only have a one way solution which only helps for outgoing call legs, not incoming. DSL isnt Half duplex as Michael Wilson said, but the frequency used for upstream is lower range. Or is it an sdsl connection and not adsl? If its sdsl or gshdsl or something like this, the frequencys used are the same range for up and download. the best way to do this is to ask for an QoS for your DSL from your ISP. We are using Broadsoft SIP trunks. Thanks for any suggestions. Michael D Mosier best regards steve smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
