Re: [Asterisk-Users] why even use SIP
On Sat, 26 Mar 2005 04:14:54 +0200, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Mar 23, 2005 at 04:37:02PM -0500, Dana Olson wrote: My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. scan for a MAC address? ping all the addresses in the range and then /usr/sbin/arp -n |grep -i that_mac_addr The scanning part could be done using something like: nmap -sP 192.168.1-5.* Another simple trick (assuming a mostly windows network) is to simply ping to the broadcast address. Linux-es and macs tend to respond to those pings and so are most devices. Windows tend to ignore those pings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend The MAC addresses are not labeled on the units. I swear I said that already. ___ 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] why even use SIP
after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t On Thu, 24 Mar 2005 16:39:25 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 24 Mar 2005 13:03:46 -0700, JD Austin [EMAIL PROTECTED] wrote: xlite doesn't seem to have this problem. X-Lite doesn't support IAX. ___ 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] why even use SIP
On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] why even use SIP
each of the desks already has two rj45 network ports so it makes sense for me to put a sipura 2001 at each of the deks rather then getting a channel bank and then having to do new cabling, will i be ok with ordering 25 of the sipura 2001 since each one of them have 2 FXS ports. Or are there firmware/voice quality/asterisk integration issues to use both FXS ports on a sipura 2001 simultaneously. t On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] why even use SIP
On Fri, 25 Mar 2005 13:14:12 -0800, Sys Admin wrote: each of the desks already has two rj45 network ports so it makes sense for me to put a sipura 2001 at each of the deks rather then getting a channel bank and then having to do new cabling, will i be ok with ordering 25 of the sipura 2001 since each one of them have 2 FXS ports. Or are there firmware/voice quality/asterisk integration issues to use both FXS ports on a sipura 2001 simultaneously. t On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. If you already have all that Cat5 then why not consider low end IP phones? I shudder to think about having to configure and support all those SPAs. All those wall-wart PSUs. Seems less than elegant. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] why even use SIP
exactly u bring me to one more anamoly in this industry: (the first one was IAX being beter then SIP but not yet ready for prime time) with a sipura 2001 FXS port the company I am consulting with can plug in very high end quality analog phones each say for $50 offering speaker phone / cordless phone / answering machine integrated with the phone which they can hear before they decide to pick up or not etc etc... So each workstation ends up costing say $90 ($50 for the phone and $40 for the one FXS port on a sipura 2001) Now this same feature set if I was to look for in a IP phone, the cost would be more then $300, whats up with that ? t On Fri, 25 Mar 2005 15:21:16 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 13:14:12 -0800, Sys Admin wrote: each of the desks already has two rj45 network ports so it makes sense for me to put a sipura 2001 at each of the deks rather then getting a channel bank and then having to do new cabling, will i be ok with ordering 25 of the sipura 2001 since each one of them have 2 FXS ports. Or are there firmware/voice quality/asterisk integration issues to use both FXS ports on a sipura 2001 simultaneously. t On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. If you already have all that Cat5 then why not consider low end IP phones? I shudder to think about having to configure and support all those SPAs. All those wall-wart PSUs. Seems less than elegant. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] why even use SIP
I'd look at the Polycom IP300 or IP500. They do an awfull lot for under $200 each. Moreover, you'd have multi-line capability, transfers, conferencing on-phone, etc. They are provisioned centrally from an ftp server and easily configured. You can can also power them over ethernet, which would eliminate the gaggle of wall-warts that the Sipura's need. You might also consider the Sipura IP phone, although I have no experience with it. It's supposed to be a two line phone. I just think that an all-digital solution is way more promising in the long run. Worth the effort, even if the initial cost is slightly more. Michael On Fri, 25 Mar 2005 13:28:20 -0800, Sys Admin wrote: exactly u bring me to one more anamoly in this industry: (the first one was IAX being beter then SIP but not yet ready for prime time) with a sipura 2001 FXS port the company I am consulting with can plug in very high end quality analog phones each say for $50 offering speaker phone / cordless phone / answering machine integrated with the phone which they can hear before they decide to pick up or not etc etc... So each workstation ends up costing say $90 ($50 for the phone and $40 for the one FXS port on a sipura 2001) Now this same feature set if I was to look for in a IP phone, the cost would be more then $300, whats up with that ? t On Fri, 25 Mar 2005 15:21:16 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 13:14:12 -0800, Sys Admin wrote: each of the desks already has two rj45 network ports so it makes sense for me to put a sipura 2001 at each of the deks rather then getting a channel bank and then having to do new cabling, will i be ok with ordering 25 of the sipura 2001 since each one of them have 2 FXS ports. Or are there firmware/voice quality/asterisk integration issues to use both FXS ports on a sipura 2001 simultaneously. t On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. If you already have all that Cat5 then why not consider low end IP phones? I shudder to think about having to configure and support all those SPAs. All those wall-wart PSUs. Seems less than elegant. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] why even use SIP
On Wed, Mar 23, 2005 at 04:37:02PM -0500, Dana Olson wrote: My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. scan for a MAC address? ping all the addresses in the range and then /usr/sbin/arp -n |grep -i that_mac_addr The scanning part could be done using something like: nmap -sP 192.168.1-5.* Another simple trick (assuming a mostly windows network) is to simply ping to the broadcast address. Linux-es and macs tend to respond to those pings and so are most devices. Windows tend to ignore those pings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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] why even use SIP
A few ideas to help the Iaxy little device work better: at the very first startup (or with a switch), finding the LAN and a phone connected, it could call it and say it's IP (like 'one' 'nine' 'eight' 'dot' and so on). Maybe with a little improvment it could also be called from it's telephone and acquire a fixed IP address by DTMF codes... Alex is some one from digium reading this thread. !! Looks like they have a ready and a big market for this device. And all they need to do is invest say 6 man months of development effort :) come on digium do it !! How about making the firmware open source so we can hack on it ... t On Wed, 23 Mar 2005 16:37:02 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit [EMAIL PROTECTED] wrote: So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? Oups, sorry, didn't think about this one. Check winiaxyprov, the version 1.01 can scan your network to find IAXy. Now the only thing we need is for Digium to write the MAC address on the device before sending it in the open world. Because if you have more than one on your network, you can't really know which one you need to provision. hth Yeah, I found that app earlier in the thread and thanked whoever it was (maybe you, can't remember) for linking to it. It's handy, as I had no way to determine the MAC or IP address prior to this, my IAXys kinda sat on the shelf collecting dust. (I did bring one home and plugged it into my Linksys router, but that's hardly an option in a large IT organization with many IAXys.) My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. Luckily I actually have a Windows desktop here at work. I'd like a scanner like that for Linux though. Maybe it's possible with some other kind of application? Anyhow, I still think it wouldn't kill them to add an IP address feature or something (an alternative would be to allow the iaxyprov tool to provision by MAC or IP, and yes, start labeling the devices with their MACs). To me, it just doesn't seem like a product that was really ready for release yet. I think it could be really great after a bit of development though, and wouldn't discourage Digium from doing so, but for now, our company can't really use these for many applications. -- Dana ___ 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 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] why even use SIP
Dana Olson wrote: On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote: 6) Configuration requires Linux, as opposed to a web browser or something more standard. I compiled iaxprov on Cygwin, works nicely. There's somebody on this list that made a Windows version to provision it. Works nicely, GUI interface, can even scan the LAN to find IAXy. Here is the link to it : http://dacosta.dynip.com/asterisk Thanks for that link, I'm gonna try it. The main issue here is that this is a large company and I don't have access to the DHCP servers, and therefore can't just find out the IP address of this thing. There's another feature request. Let me dial ### or something to find my IP... How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe -- http://www.umich2.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
Re: [Asterisk-Users] why even use SIP
On Thu, 24 Mar 2005 08:09:19 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Dana Olson wrote: On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote: 6) Configuration requires Linux, as opposed to a web browser or something more standard. I compiled iaxprov on Cygwin, works nicely. There's somebody on this list that made a Windows version to provision it. Works nicely, GUI interface, can even scan the LAN to find IAXy. Here is the link to it : http://dacosta.dynip.com/asterisk Thanks for that link, I'm gonna try it. The main issue here is that this is a large company and I don't have access to the DHCP servers, and therefore can't just find out the IP address of this thing. There's another feature request. Let me dial ### or something to find my IP... How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe -- http://www.umich2.com Digium doesn't label the MAC address on the device, unless it's such a fine print that no one can read it. I believe this has been said a few times in the conversation. ___ 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] why even use SIP
How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe -- http://www.umich2.com Digium doesn't label the MAC address on the device, unless it's such a fine print that no one can read it. I believe this has been said a few times in the conversation. Connect it with a cross-over ethernet cable to a Linux box and run tcpdump on the Linux box, before long you'll see the IP address come up on the tcpdump logs. Don't power it off, you want it to have an existing DHCP lease. If you don't see any traffic, try making a call. Once you have the IP you can put it back on the normal network and configure it. ___ 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] why even use SIP
On Thu, 24 Mar 2005 15:34:26 -, Giles Coochey [EMAIL PROTECTED] wrote: How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe -- http://www.umich2.com Digium doesn't label the MAC address on the device, unless it's such a fine print that no one can read it. I believe this has been said a few times in the conversation. Connect it with a cross-over ethernet cable to a Linux box and run tcpdump on the Linux box, before long you'll see the IP address come up on the tcpdump logs. Don't power it off, you want it to have an existing DHCP lease. If you don't see any traffic, try making a call. Once you have the IP you can put it back on the normal network and configure it. I know how to work around these limitations already. My point is that this is not an enterprise-ready solution. If I order 1000 of these for our IT staff, I have to go through each and every one with a crossover cable just to find the IP? Why would we bother when there so many other devices that don't have any of the flaws of the IAXy? Of course they are SIP-only, so that's the answer to the question of why use SIP at all. Because there is no good solution for IAX yet. With a little work, the IAXy can become a product not only for hobbyists but for the corporate world as well. Until then, we will need to rely on Sipura, Grandstream, and the like for devices that can be much easier provisioned, either by keypad entry on the device itself, TFTP config files, or an HTTP interface, that support DNS name resolution, G729/iLBC/GSM codecs, have their MAC addresses labeled on them, etc. This is for my company only. Perhaps yours isn't so large and you have the time and desire to go through this process for every device in your organization, but we don't. Yes, for home users who run Asterisk, it's fine, except if they want to take the IAXy on the road with them and they don't have a static IP address. For internal use in a small company, yeah, the IAXy may be a fine solution. But when you're looking at purchasing hundreds of devices at a time, I don't think this is a good product at this time. All of that said, I like the IAXy, and I will gladly recommend buying it if you're not in my position, or if Digium develops it further to address these issues. ___ 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] why even use SIP
just called digium using firefly softphone connected to a asterisk server using IAX2 they said that the IAXy device is not in stock and the earliest expected arrival is after a month. On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! t On Thu, 24 Mar 2005 12:07:24 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 24 Mar 2005 15:34:26 -, Giles Coochey [EMAIL PROTECTED] wrote: How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe -- http://www.umich2.com Digium doesn't label the MAC address on the device, unless it's such a fine print that no one can read it. I believe this has been said a few times in the conversation. Connect it with a cross-over ethernet cable to a Linux box and run tcpdump on the Linux box, before long you'll see the IP address come up on the tcpdump logs. Don't power it off, you want it to have an existing DHCP lease. If you don't see any traffic, try making a call. Once you have the IP you can put it back on the normal network and configure it. I know how to work around these limitations already. My point is that this is not an enterprise-ready solution. If I order 1000 of these for our IT staff, I have to go through each and every one with a crossover cable just to find the IP? Why would we bother when there so many other devices that don't have any of the flaws of the IAXy? Of course they are SIP-only, so that's the answer to the question of why use SIP at all. Because there is no good solution for IAX yet. With a little work, the IAXy can become a product not only for hobbyists but for the corporate world as well. Until then, we will need to rely on Sipura, Grandstream, and the like for devices that can be much easier provisioned, either by keypad entry on the device itself, TFTP config files, or an HTTP interface, that support DNS name resolution, G729/iLBC/GSM codecs, have their MAC addresses labeled on them, etc. This is for my company only. Perhaps yours isn't so large and you have the time and desire to go through this process for every device in your organization, but we don't. Yes, for home users who run Asterisk, it's fine, except if they want to take the IAXy on the road with them and they don't have a static IP address. For internal use in a small company, yeah, the IAXy may be a fine solution. But when you're looking at purchasing hundreds of devices at a time, I don't think this is a good product at this time. All of that said, I like the IAXy, and I will gladly recommend buying it if you're not in my position, or if Digium develops it further to address these issues. ___ 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 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] why even use SIP
Because the video driver is a kernel thread and not allowed to lag. That would cause framerate issues with games. :) oh winderz... quote who=Sys Admin On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! -- END OF LINE -MCP ___ 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] why even use SIP
well i even pressed ctrl+alt+del went into the process monitor and gave the firefly process high priority. Still it looses half a second of sound each time i maximize or minimize a app like putty, whats the word for this . sucks . why doesnt skype have this problem ? t On Thu, 24 Mar 2005 10:39:54 -0800 (PST), Robert Hajime Lanning [EMAIL PROTECTED] wrote: Because the video driver is a kernel thread and not allowed to lag. That would cause framerate issues with games. :) oh winderz... quote who=Sys Admin On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! -- END OF LINE -MCP ___ 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 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] why even use SIP
xlite doesn't seem to have this problem. Sys Admin wrote: well i even pressed ctrl+alt+del went into the process monitor and gave the firefly process high priority. Still it looses half a second of sound each time i maximize or minimize a app like putty, whats the word for this . sucks . why doesnt skype have this problem ? t On Thu, 24 Mar 2005 10:39:54 -0800 (PST), Robert Hajime Lanning [EMAIL PROTECTED] wrote: Because the video driver is a kernel thread and not allowed to lag. That would cause framerate issues with games. :) oh winderz... quote who="Sys Admin" On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! -- END OF LINE -MCP ___ 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 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 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] why even use SIP
On Thu, 24 Mar 2005 13:03:46 -0700, JD Austin [EMAIL PROTECTED] wrote: xlite doesn't seem to have this problem. X-Lite doesn't support IAX. ___ 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] why even use SIP
5) MWI, Call Waiting, 3-way calling missing If I remember correctly (only used an IAXy a couple of times), it uses shutter-tone to tell you when there's a message waiting It definitely support Call Waiting : just use Flash as with normal call waiting on the PSTN Never tried 3-way calling, but I think it supports it 6) Configuration requires Linux, as opposed to a web browser or something more standard. I compiled iaxprov on Cygwin, works nicely. There's somebody on this list that made a Windows version to provision it. Works nicely, GUI interface, can even scan the LAN to find IAXy. Here is the link to it : http://dacosta.dynip.com/asterisk Let's face facts there, the IAXy sucks by any definition. No it doesn't. Granted it has a couple shortcomings, but nothing that bad. If Digium can fix the most important ones and find a way to drop the price a bit, this would be a great little device. Just my $0.02 CDN ___ 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] why even use SIP
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote: 6) Configuration requires Linux, as opposed to a web browser or something more standard. I compiled iaxprov on Cygwin, works nicely. There's somebody on this list that made a Windows version to provision it. Works nicely, GUI interface, can even scan the LAN to find IAXy. Here is the link to it : http://dacosta.dynip.com/asterisk Thanks for that link, I'm gonna try it. The main issue here is that this is a large company and I don't have access to the DHCP servers, and therefore can't just find out the IP address of this thing. There's another feature request. Let me dial ### or something to find my IP... Let's face facts there, the IAXy sucks by any definition. No it doesn't. Granted it has a couple shortcomings, but nothing that bad. If Digium can fix the most important ones and find a way to drop the price a bit, this would be a great little device. You just admitted that without the features, it's not great... Seems like we're all on the same page here. No sense arguing about it. ___ 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] why even use SIP
There's another feature request. Let me dial ### or something to find my IP... That's not something to do with the IAXy, you can make an AGI script that will tell you your IP. I had this script somewhere but I can't find it at the moment. This would not only be valid for the IAXy, but for any phone connected to asterisk (well, except analog phones) If I find it, I'll let you know. But I'm confident that somebody on this list has something like this. You dial some extension that call this script and it tells you your IP using SayDigits. hth ___ 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] why even use SIP
On Wed, 23 Mar 2005 13:59:07 -0500, Time Bandit [EMAIL PROTECTED] wrote: There's another feature request. Let me dial ### or something to find my IP... That's not something to do with the IAXy, you can make an AGI script that will tell you your IP. I had this script somewhere but I can't find it at the moment. This would not only be valid for the IAXy, but for any phone connected to asterisk (well, except analog phones) If I find it, I'll let you know. But I'm confident that somebody on this list has something like this. You dial some extension that call this script and it tells you your IP using SayDigits. hth So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? ./iaxyprov Usage: provision ip file ___ 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] why even use SIP
so seems like the verdict is go IAXy with a IAX only network ? Most of the problems of the IAXy device seems like will be fixed with firmware updates and wont require a hardware update.. this way we get the advantage of a Hardphone (human factor, just feel good to talk on a real phone) with all the goodies of the IAX protocol. t On Wed, 23 Mar 2005 14:23:17 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Wed, 23 Mar 2005 13:59:07 -0500, Time Bandit [EMAIL PROTECTED] wrote: There's another feature request. Let me dial ### or something to find my IP... That's not something to do with the IAXy, you can make an AGI script that will tell you your IP. I had this script somewhere but I can't find it at the moment. This would not only be valid for the IAXy, but for any phone connected to asterisk (well, except analog phones) If I find it, I'll let you know. But I'm confident that somebody on this list has something like this. You dial some extension that call this script and it tells you your IP using SayDigits. hth So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? ./iaxyprov Usage: provision ip file ___ 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 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] why even use SIP
So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? Oups, sorry, didn't think about this one. Check winiaxyprov, the version 1.01 can scan your network to find IAXy. Now the only thing we need is for Digium to write the MAC address on the device before sending it in the open world. Because if you have more than one on your network, you can't really know which one you need to provision. hth ___ 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] why even use SIP
so seems like the verdict is go IAXy with a IAX only network ? Most of the problems of the IAXy device seems like will be fixed with firmware updates and wont require a hardware update.. The best part is how you update the firmware : each time an IAXy connect to Asterisk, it check what firmware version it as. If an update is available (in Asterisk), it automagically update it. After seeing how much pain a Cisco phone needs to get to the latest firmware version, I thank god (or Digium) for this device :) Now, if only Digium can fix those small issues ___ 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] why even use SIP
On Wed, 23 Mar 2005 12:55:45 -0800, Sys Admin [EMAIL PROTECTED] wrote: so seems like the verdict is go IAXy with a IAX only network ? Most of the problems of the IAXy device seems like will be fixed with firmware updates and wont require a hardware update.. this way we get the advantage of a Hardphone (human factor, just feel good to talk on a real phone) with all the goodies of the IAX protocol. t Has Digium said that they will fix the issues most of us have with the IAXy? I haven't seen it, but maybe I missed the message? If I were you, I'd get one IAXy device in and test it first, see if it is what you want... And for comparison, grab a low-end Sipura ATA in as well, see what you decide. ___ 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] why even use SIP
On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit [EMAIL PROTECTED] wrote: So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? Oups, sorry, didn't think about this one. Check winiaxyprov, the version 1.01 can scan your network to find IAXy. Now the only thing we need is for Digium to write the MAC address on the device before sending it in the open world. Because if you have more than one on your network, you can't really know which one you need to provision. hth Yeah, I found that app earlier in the thread and thanked whoever it was (maybe you, can't remember) for linking to it. It's handy, as I had no way to determine the MAC or IP address prior to this, my IAXys kinda sat on the shelf collecting dust. (I did bring one home and plugged it into my Linksys router, but that's hardly an option in a large IT organization with many IAXys.) My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. Luckily I actually have a Windows desktop here at work. I'd like a scanner like that for Linux though. Maybe it's possible with some other kind of application? Anyhow, I still think it wouldn't kill them to add an IP address feature or something (an alternative would be to allow the iaxyprov tool to provision by MAC or IP, and yes, start labeling the devices with their MACs). To me, it just doesn't seem like a product that was really ready for release yet. I think it could be really great after a bit of development though, and wouldn't discourage Digium from doing so, but for now, our company can't really use these for many applications. -- Dana ___ 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] why even use SIP
is some one from digium reading this thread. !! Looks like they have a ready and a big market for this device. And all they need to do is invest say 6 man months of development effort :) come on digium do it !! How about making the firmware open source so we can hack on it ... t On Wed, 23 Mar 2005 16:37:02 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit [EMAIL PROTECTED] wrote: So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? Oups, sorry, didn't think about this one. Check winiaxyprov, the version 1.01 can scan your network to find IAXy. Now the only thing we need is for Digium to write the MAC address on the device before sending it in the open world. Because if you have more than one on your network, you can't really know which one you need to provision. hth Yeah, I found that app earlier in the thread and thanked whoever it was (maybe you, can't remember) for linking to it. It's handy, as I had no way to determine the MAC or IP address prior to this, my IAXys kinda sat on the shelf collecting dust. (I did bring one home and plugged it into my Linksys router, but that's hardly an option in a large IT organization with many IAXys.) My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. Luckily I actually have a Windows desktop here at work. I'd like a scanner like that for Linux though. Maybe it's possible with some other kind of application? Anyhow, I still think it wouldn't kill them to add an IP address feature or something (an alternative would be to allow the iaxyprov tool to provision by MAC or IP, and yes, start labeling the devices with their MACs). To me, it just doesn't seem like a product that was really ready for release yet. I think it could be really great after a bit of development though, and wouldn't discourage Digium from doing so, but for now, our company can't really use these for many applications. -- Dana ___ 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 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] why even use SIP
Sys Admin wrote: After 20 posts, in 2005 the ideal setup for a new installtion of a 50 user asterisk is: Option1: IAX2 with softphone firefly Option2: SIP with softphone Option3: IAX2 with hardphones (which brand?) Option4: SIP with hardphones. Seems like we cannot come to a definite conclusion, poll ? On a LAN where NAT is not an issue I would go for SIP + decent hardphones with good echo cancellation. On the internet with all sort of NATs + Firewalls, IAX is a must but unfortunately I don't know of any good, readily available hardphones. ___ 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] why even use SIP
On Tue, 22 Mar 2005 12:10:17 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: On a LAN where NAT is not an issue I would go for SIP + decent hardphones with good echo cancellation. On the internet with all sort of NATs + Firewalls, IAX is a must but unfortunately I don't know of any good, readily available hardphones. If the IAXy had a bit more work done on it, it could be a good option, but it's not at the current time. -- Dana ___ 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] why even use SIP
After 20 posts, in 2005 the ideal setup for a new installtion of a 50 user asterisk is: Option1: IAX2 with softphone firefly Option2: SIP with softphone Option3: IAX2 with hardphones (which brand?) Option4: SIP with hardphones. As the other poster said, I doubt you'll find a consensus as there are too many variables and no single perfect solution. With 50 users is Option 1 even viable? Firefly doesn't support more than one line appearance and with that large an office I'd think people would insist on more than one line. But Firefly did work the best of the IAX softphones I tried in terms of sound quality. Options 2 and 4 are both going to depend on whether there are users outside the local LAN because if there are you'll have to deal with the SIP NAT/firewall issues and that's an ugly project. That leaves Option 3 and none of the reviews I've read of the IAX hardphones made me want to go out and buy 50 of them. At this point I don't think there's any solution that will work without tradeoffs specific to your situation. Be seeing you. ___ 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] why even use SIP
After a few years working with various business PBX's, I've found that users see, feel, and hear the phones. Assuming that voice quality is not crap and you have enough trunks available, users will evaluate the PBX based on their experience with the phone. I'm always amazed that even small business that are basically very cheap will pick a higher quality, higher priced phone almost every time. This leads to option4 as there is getting to be a nice selection of SIP phones available. Polycom and Cisco phones look and feel good. I haven't tried all the other options so there may be a great new option. Brian On Tue, 2005-03-22 at 01:32, Sys Admin wrote: After 20 posts, in 2005 the ideal setup for a new installtion of a 50 user asterisk is: Option1: IAX2 with softphone firefly Option2: SIP with softphone Option3: IAX2 with hardphones (which brand?) Option4: SIP with hardphones. Seems like we cannot come to a definite conclusion, poll ? so the verdict is ? On Tue, 22 Mar 2005 11:48:00 +0800, Shaun Dwyer [EMAIL PROTECTED] wrote: Hi Scott, Intresting to know, cheers :) Only down side though, is that most people using softphones will be using Windows... If only ALSA was available for windows ;) -Shaun Scott Williamson wrote: That should be the program alsamixer, not amixer. Make sure to press F5 to get all of the playback/capture devices shown. On Mon, 2005-21-03 at 22:02 -0500, Scott Williamson wrote: Ah, Console sound card echo. I found that with my cheep YMFPCI sound card that there is a channel called wave capture that is enabled for recording by default. And this is only visible when one uses ALSA sound drivers. One needs to use an ALSA mixer control program (I use amixer, the text mode one) to disable, or reduce the volume on these sound cards. Once this is done there is no more echo at all! Why this channel does not appear under OSS, and why it appears to be enabled by default in the hardware is beyond me. The GUI mixers all list this channel simply as WAVE, and I have two WAVE channels. So remember amixer. The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734) by ALSA On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote: Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Not every problem someone has with his girlfriend is necessarily due to the capitalist mode of production. -- Herbert Marcuse ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Humor in the Court: Q: The truth of the matter is that you were not an unbiased, objective witness, isn't it. You too were shot in the fracas? A: No, sir. I was shot midway between the fracas and the naval. ___ 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 mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] why even use SIP
If the IAXy had a bit more work done on it, it could be a good option, but it's not at the current time. Yep! Things like: - more codecs (just ulaw? come on...) - proper DHCP and possibility of static IP - a 'reset' button To start with would be nice to have. And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( ___ 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] why even use SIP
Yeah, if we're making a list, add DNS name resolution to that list. :) -- Dana On Tue, 22 Mar 2005 23:48:46 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: If the IAXy had a bit more work done on it, it could be a good option, but it's not at the current time. Yep! Things like: - more codecs (just ulaw? come on...) - proper DHCP and possibility of static IP - a 'reset' button To start with would be nice to have. And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( ___ 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] why even use SIP
Hello Scott, Thanks a lot for the info. Ill need to do a bit more testing with 3 to 5 simultaneous calls to see if there is any problem. I dont really like the hardphones, because you dont have all the flexibility offered by the sofphone solution (click-to-call, paste-n-call, hands-free talking with a headset, etc). Moreover, headsets are much cheaper than the hardphones. Actually Im using IAX to hook to VoipJet.com and there seems to be no NAT between my server and that of sipsnip.de. Maybe your Asterisk version was pretty old? Perhaps the problems are solved in the new releases (Im now using a month-old CVS version). Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Bussinger Sent: Montag, 21. März 2005 22:32 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] why even use SIP I'm testing a softphone-only setup (SJPhone with Plantronics 80 Headsets plugged into Soundcard) with around 40 users for that are linked over LAN in an organization of around 300 people and never had any of the problems you described (the test is going for over a month now). I'm glad it worked for you. In my case it generally worked pretty well for me as an individual, but as soon as we ramped it up for real use by more than 1 or 2 people it pretty much fell apart. X-Lite had the feature set we needed, but voice quality was terrible and was full of clicks. All of the IAX softphones based on the iaxclient library had the same problem with too long a delay in them. Most of the softphones were not useful to use because they were frankly too hard to use for average users. And on it went. I'm using iLBC codec and the Asterisk is running on a PIII PC with 256 MB RAM :-) I tried ulaw, gsm, and speex codecs and the results were pretty similar for each. I ended up using ulaw internally and gsm to connect to the external provider (to cut down bandwidth requirements). Outgoing calls to landline via VoipJet and sipsnip.de, incoming from FWD (landline number provided by ipkall.com for free). I use FWD for my phone at home to connect to the office. I haven't had the energy to tackle the SIP over NAT issues yet. :) We are pretty happy with this setup so far. Do you think there might be problems later? If it's working for you, great! Our problems were noticable as soon as we had 4 or 5 people on the phones at the same time so as long as you test it carefully under realistic use you should be fine. I never could figure out why my systems would never work well, but it just wasn't worth spending any more time on (cheaper to just buy the hardphones which didn't have the problem). Be seeing you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] why even use SIP
- proper DHCP and possibility of static IP Never had a problem with mine. I set my DHCP server to hand out a specific IP to the IAXy, too. - a 'reset' button What's the advantage over unplugging the unit and plugging it back in? And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( My IAXy works perfectly fine with a cheapo Panasonic cordless I bought here in Germany. jens ___ 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] why even use SIP
Did you check SJPhone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Bussinger Sent: Montag, 21. März 2005 22:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] why even use SIP Did you consider vonage? I played with Packet8 as a reality check before investing the time money into the Asterisk solution. I certainly wouldn't use Packet8 or Vonage or any other service instead of an Asterisk solution. It's easy to get termination delivered by IAX now and is both cheaper and more flexible. I wonder if I am making a rod for my own back trying to set up voip for such a small configuration. I've got about 10 users and the VoIP is wonderful. It's _MUCH_ cheaper than the old system I replaced and has lots of flexibility going forward. I'm actually very happy about the switch to Asterisk and VoIP. I just wish I could have found a good softphone to use. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] why even use SIP
And some kind of password to protect the config, so not any intelligent person with a prov. Utility can change your iaxy! Back on topic, I much prefer iax personally, however the lack of hardphone options is a real pain. Obviously depending on the situation, I use SIP hardphones on internal LAN, IAX softphone (firefly) on laptop for roaming, and all incoming and outgoing calls from * are sent via IAX. I use firefly with a Bluetooth headset for audio device and quality is stunning. However I agree firefly is not really upto it in a business environment. Regards C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson Sent: 22 March 2005 20:20 To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] why even use SIP Yeah, if we're making a list, add DNS name resolution to that list. :) -- Dana On Tue, 22 Mar 2005 23:48:46 +0400, Jean-Michel Hiver [EMAIL PROTECTED] wrote: If the IAXy had a bit more work done on it, it could be a good option, but it's not at the current time. Yep! Things like: - more codecs (just ulaw? come on...) - proper DHCP and possibility of static IP - a 'reset' button To start with would be nice to have. And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( ___ 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 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] why even use SIP
How about sipXphone? They say STUN will be added within a week or two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Zhovtulya Sent: Tuesday, March 22, 2005 4:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] why even use SIP Did you check SJPhone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Bussinger Sent: Montag, 21. März 2005 22:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] why even use SIP Did you consider vonage? I played with Packet8 as a reality check before investing the time money into the Asterisk solution. I certainly wouldn't use Packet8 or Vonage or any other service instead of an Asterisk solution. It's easy to get termination delivered by IAX now and is both cheaper and more flexible. I wonder if I am making a rod for my own back trying to set up voip for such a small configuration. I've got about 10 users and the VoIP is wonderful. It's _MUCH_ cheaper than the old system I replaced and has lots of flexibility going forward. I'm actually very happy about the switch to Asterisk and VoIP. I just wish I could have found a good softphone to use. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 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] why even use SIP
Never had a problem with mine. I set my DHCP server to hand out a specific IP to the IAXy, too. Works on some, but not all. It fails on Microsoft and Cisco DHCP Servers. Just because it works for you doesn't mean it's implemented correctly. What's the advantage over unplugging the unit and plugging it back in? That's not resetting, that's rebooting. There is a difference. If I forget the password to my linksys, I can hit the reset button to reset to defaults. Besides there are many things wrong with the IAXy. 1) Price, twice as expensive as the SIPURA equivalent 2) Codecs, only supports ulaw/pcm 3) Doesn't properly support DHCP 4) Security, there is none. If it can be communicated with, then it can be zapped without a password 5) MWI, Call Waiting, 3-way calling missing 6) Configuration requires Linux, as opposed to a web browser or something more standard. Let's face facts there, the IAXy sucks by any definition. Digium support is pretty much go on IRC or e-mail this list and ask here. After my experience with the IAXy, I never thought I'd be happy to give Linksys my money (PAP2-NA's) On Tue, 22 Mar 2005 22:02:43 +0100, Jens Vagelpohl [EMAIL PROTECTED] wrote: - proper DHCP and possibility of static IP Never had a problem with mine. I set my DHCP server to hand out a specific IP to the IAXy, too. - a 'reset' button What's the advantage over unplugging the unit and plugging it back in? And my IAXy doesn't work with my european phone (no tone) it's kind of a drag :( My IAXy works perfectly fine with a cheapo Panasonic cordless I bought here in Germany. jens ___ 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 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] why even use SIP
On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. That is rather subjective. Many people consider SIP better than IAX since: * There is a published standard, as opposed to some source code. * It leverages a lot of existing standards already. * It is media and addressing agnostic. SIP can support any media, and nearly any kind of addressing. IAX is heavily mired in legacy telphone standards. Routing 3XX to PBX A, and 4XX to PBX B, is 80's PBX style telephony over IP. * It separates media and signalling. This is the biggest IAX problem: Why should a call switch have to get in the middle of the media to make a call routing decision? I find that since Asterisk has an overly complex and still incomplete SIP implementation, if the only exposure you have to SIP, is via Asterisk, you should have a poor opinion of SIP. Asterisk makes SIP hard. Asterisk doesn't even support a lot of the SIP capabilities. For instance, why can't the media type change during a call-transfer? Why does Asterisk have to be in the media path to support a t or T type transfer if SIP INFO DTMF signalling is used? Why on earth does every two-bit RTP implementation support VAD, but not Asterisk? Tom ___ 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] why even use SIP
I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! t ___ 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] why even use SIP
Sys Admin wrote: After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? ___ 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] why even use SIP
On Mon, 21 Mar 2005 10:01:04 -0800, Sys Admin [EMAIL PROTECTED] wrote: I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! t Do you have your voip hardphones picked out yet? ___ 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] why even use SIP
Sys Admin wrote: I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! Because most equipment doesn't support IAX -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] why even use SIP
On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ 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] why even use SIP
If you answer the question why 95% of the desktop market is owned by Microsoft or why gasoline is used as the fuel for internal combustion engines, you will know the answer as to why sip. The best technology dosn't always win the market. -JB Andrew Kohlsmith wrote: On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ 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 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] why even use SIP
2 reasons for not using IAX: A. CDR as part of the media B. Not good hardphone / softphone Problem A (CDR as part of the media) I am not worried about too much, as long as the data is there some parsing will allow me to extract it and then do what i want to do with it. Problem B (Not good hardphone / softphone) what would be the best hardphone/softphone to make it work ? t On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ 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 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] why even use SIP
btw if there is no good softphone for IAX which does G729, i could possibly get my company to buy consulting time from say DIAX and make dante develop it further, t On Mon, 21 Mar 2005 11:03:48 -0800, Sys Admin [EMAIL PROTECTED] wrote: 2 reasons for not using IAX: A. CDR as part of the media B. Not good hardphone / softphone Problem A (CDR as part of the media) I am not worried about too much, as long as the data is there some parsing will allow me to extract it and then do what i want to do with it. Problem B (Not good hardphone / softphone) what would be the best hardphone/softphone to make it work ? t On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ 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 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] why even use SIP
Forget older years but in 2005 do hard phones really add any value over softphones. The call center agents already have p4 2.4ghz with 512 MB ram Win2K why not just get them a nice USB headset with a softphone IAX client, We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues from picking up radio stations on the audio inputs, voice quality being quite variable (sometimes almost impossible to listen to), some echo issues, and severe delays (as much as 1/2 second at times). I tried SJPhone, Xlite, DIAX, IAXPhone, Firefly, and a bunch more I can't think of at the moment and all showed one problem or another. Also, very few of the clients are really business-ready (i.e. multiple call appearances, multiple lines, hold, transfer, DND, etc.). We finally gave up and bought everyone Sipura SPA841 hard phones and we're _much_ happier! I wish I hadn't had to go SIP, but none of the IAX phones looked like good choices to me. Be seeing you. ___ 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] why even use SIP
Thats really strange. Im testing a softphone-only setup (SJPhone with Plantronics 80 Headsets plugged into Soundcard) with around 40 users for that are linked over LAN in an organization of around 300 people and never had any of the problems you described (the test is going for over a month now). Im using iLBC codec and the Asterisk is running on a PIII PC with 256 MB RAM :-) Outgoing calls to landline via VoipJet and sipsnip.de, incoming from FWD (landline number provided by ipkall.com for free). We are pretty happy with this setup so far. Do you think there might be problems later? Regards, Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Bussinger Sent: Montag, 21. März 2005 20:19 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] why even use SIP Forget older years but in 2005 do hard phones really add any value over softphones. The call center agents already have p4 2.4ghz with 512 MB ram Win2K why not just get them a nice USB headset with a softphone IAX client, We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues from picking up radio stations on the audio inputs, voice quality being quite variable (sometimes almost impossible to listen to), some echo issues, and severe delays (as much as 1/2 second at times). I tried SJPhone, Xlite, DIAX, IAXPhone, Firefly, and a bunch more I can't think of at the moment and all showed one problem or another. Also, very few of the clients are really business-ready (i.e. multiple call appearances, multiple lines, hold, transfer, DND, etc.). We finally gave up and bought everyone Sipura SPA841 hard phones and we're _much_ happier! I wish I hadn't had to go SIP, but none of the IAX phones looked like good choices to me. Be seeing you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] why even use SIP
Did you consider vonage? I played with Packet8 as a reality check before investing the time money into the Asterisk solution. I certainly wouldn't use Packet8 or Vonage or any other service instead of an Asterisk solution. It's easy to get termination delivered by IAX now and is both cheaper and more flexible. I wonder if I am making a rod for my own back trying to set up voip for such a small configuration. I've got about 10 users and the VoIP is wonderful. It's _MUCH_ cheaper than the old system I replaced and has lots of flexibility going forward. I'm actually very happy about the switch to Asterisk and VoIP. I just wish I could have found a good softphone to use. ___ 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] why even use SIP
I'm testing a softphone-only setup (SJPhone with Plantronics 80 Headsets plugged into Soundcard) with around 40 users for that are linked over LAN in an organization of around 300 people and never had any of the problems you described (the test is going for over a month now). I'm glad it worked for you. In my case it generally worked pretty well for me as an individual, but as soon as we ramped it up for real use by more than 1 or 2 people it pretty much fell apart. X-Lite had the feature set we needed, but voice quality was terrible and was full of clicks. All of the IAX softphones based on the iaxclient library had the same problem with too long a delay in them. Most of the softphones were not useful to use because they were frankly too hard to use for average users. And on it went. I'm using iLBC codec and the Asterisk is running on a PIII PC with 256 MB RAM :-) I tried ulaw, gsm, and speex codecs and the results were pretty similar for each. I ended up using ulaw internally and gsm to connect to the external provider (to cut down bandwidth requirements). Outgoing calls to landline via VoipJet and sipsnip.de, incoming from FWD (landline number provided by ipkall.com for free). I use FWD for my phone at home to connect to the office. I haven't had the energy to tackle the SIP over NAT issues yet. :) We are pretty happy with this setup so far. Do you think there might be problems later? If it's working for you, great! Our problems were noticable as soon as we had 4 or 5 people on the phones at the same time so as long as you test it carefully under realistic use you should be fine. I never could figure out why my systems would never work well, but it just wasn't worth spending any more time on (cheaper to just buy the hardphones which didn't have the problem). Be seeing you. ___ 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] why even use SIP
For a easier comprehension, nowadays, H323 is like english. SIP is like spanish and IAX is esperanto. You can IAX. It's wonderful, modern, lot of advantages, pass through any firewall, blah...blah..blah... but you can find only some strange guys using that. :-) Isamar ___ 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] why even use SIP
Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ 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] why even use SIP
Ah, Console sound card echo. I found that with my cheep YMFPCI sound card that there is a channel called wave capture that is enabled for recording by default. And this is only visible when one uses ALSA sound drivers. One needs to use an ALSA mixer control program (I use amixer, the text mode one) to disable, or reduce the volume on these sound cards. Once this is done there is no more echo at all! Why this channel does not appear under OSS, and why it appears to be enabled by default in the hardware is beyond me. The GUI mixers all list this channel simply as WAVE, and I have two WAVE channels. So remember amixer. The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734) by ALSA On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote: Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Not every problem someone has with his girlfriend is necessarily due to the capitalist mode of production. -- Herbert Marcuse ___ 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] why even use SIP
That should be the program alsamixer, not amixer. Make sure to press F5 to get all of the playback/capture devices shown. On Mon, 2005-21-03 at 22:02 -0500, Scott Williamson wrote: Ah, Console sound card echo. I found that with my cheep YMFPCI sound card that there is a channel called wave capture that is enabled for recording by default. And this is only visible when one uses ALSA sound drivers. One needs to use an ALSA mixer control program (I use amixer, the text mode one) to disable, or reduce the volume on these sound cards. Once this is done there is no more echo at all! Why this channel does not appear under OSS, and why it appears to be enabled by default in the hardware is beyond me. The GUI mixers all list this channel simply as WAVE, and I have two WAVE channels. So remember amixer. The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734) by ALSA On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote: Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Not every problem someone has with his girlfriend is necessarily due to the capitalist mode of production. -- Herbert Marcuse ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Humor in the Court: Q: The truth of the matter is that you were not an unbiased, objective witness, isn't it. You too were shot in the fracas? A: No, sir. I was shot midway between the fracas and the naval. ___ 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] why even use SIP
Hi Scott, Intresting to know, cheers :) Only down side though, is that most people using softphones will be using Windows... If only ALSA was available for windows ;) -Shaun Scott Williamson wrote: That should be the program alsamixer, not amixer. Make sure to press F5 to get all of the playback/capture devices shown. On Mon, 2005-21-03 at 22:02 -0500, Scott Williamson wrote: Ah, Console sound card echo. I found that with my cheep YMFPCI sound card that there is a channel called wave capture that is enabled for recording by default. And this is only visible when one uses ALSA sound drivers. One needs to use an ALSA mixer control program (I use amixer, the text mode one) to disable, or reduce the volume on these sound cards. Once this is done there is no more echo at all! Why this channel does not appear under OSS, and why it appears to be enabled by default in the hardware is beyond me. The GUI mixers all list this channel simply as WAVE, and I have two WAVE channels. So remember amixer. The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734) by ALSA On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote: Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Not every problem someone has with his girlfriend is necessarily due to the capitalist mode of production. -- Herbert Marcuse ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Humor in the Court: Q: The truth of the matter is that you were not an unbiased, objective witness, isn't it. You too were shot in the fracas? A: No, sir. I was shot midway between the fracas and the naval. ___ 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 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] why even use SIP
After 20 posts, in 2005 the ideal setup for a new installtion of a 50 user asterisk is: Option1: IAX2 with softphone firefly Option2: SIP with softphone Option3: IAX2 with hardphones (which brand?) Option4: SIP with hardphones. Seems like we cannot come to a definite conclusion, poll ? so the verdict is ? On Tue, 22 Mar 2005 11:48:00 +0800, Shaun Dwyer [EMAIL PROTECTED] wrote: Hi Scott, Intresting to know, cheers :) Only down side though, is that most people using softphones will be using Windows... If only ALSA was available for windows ;) -Shaun Scott Williamson wrote: That should be the program alsamixer, not amixer. Make sure to press F5 to get all of the playback/capture devices shown. On Mon, 2005-21-03 at 22:02 -0500, Scott Williamson wrote: Ah, Console sound card echo. I found that with my cheep YMFPCI sound card that there is a channel called wave capture that is enabled for recording by default. And this is only visible when one uses ALSA sound drivers. One needs to use an ALSA mixer control program (I use amixer, the text mode one) to disable, or reduce the volume on these sound cards. Once this is done there is no more echo at all! Why this channel does not appear under OSS, and why it appears to be enabled by default in the hardware is beyond me. The GUI mixers all list this channel simply as WAVE, and I have two WAVE channels. So remember amixer. The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734) by ALSA On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote: Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Not every problem someone has with his girlfriend is necessarily due to the capitalist mode of production. -- Herbert Marcuse ___ 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 -- Scott J. Williamson [EMAIL PROTECTED] Humor in the Court: Q: The truth of the matter is that you were not an unbiased, objective witness, isn't it. You too were shot in the fracas? A: No, sir. I was shot midway between the fracas and the naval. ___ 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 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 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] why even use SIP
Sys Admin wrote: After 20 posts, in 2005 the ideal setup for a new installtion of a 50 user asterisk is: Option1: IAX2 with softphone firefly Option2: SIP with softphone Option3: IAX2 with hardphones (which brand?) Option4: SIP with hardphones. Seems like we cannot come to a definite conclusion, poll ? so the verdict is ? The verdict is that Asterisk is a very flexible and configurable tool, and there are many ways to get it to do useful work. There is no magic bullet, and I fear you're not going to find any easy one-stop answers to this question, either. B. ___ 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
